Re: [OSL | CCIE_Voice] Admin XML
Hello,
I attached an example here on using the 'executeSQLQuery' method exposed by
the AXL API. I put some comments in but let me know if you need further
explanation. Just rename it to a php extension. The Cisco developer
network is a great place to find more info on the available methods and how
to use them. If you're interested in learning about the DB tables then you
should check out the data dictionary.
https://developer.cisco.com/site/tech/communication-collaboration/management/axl/axl/
BR,
Marty
On Mon, Dec 16, 2013 at 1:24 PM, Olusegun Oguntuga
segunogunt...@gmail.comwrote:
Hi there,
Does Anyone out there have an Idea or sample code in PHP to connect to
CUCM Informix Database to read content of fields in DB.
Regards
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?php
/*
* You can download the AXL WSDL and XSD from the SQL Toolkit included in
* CUCM. Place those files in the same directory as this script, or modify
* the path accordingly.
*/
//Create your PHP SoapClient Object and pass the CUCM arguments
$client = new SoapClient(../AXLAPI.wsdl,
array('trace'=true,
'exceptions'=true,
'location'=https://192.168.158.10:8443/axl/;,
'login'='AXL-User',
'password'='cisco',
));
try {
//Query the CUCM
$results = $client-executeSQLQuery(array('sql' = 'select name from
processnode'));
//Do something with $results.
var_dump($results);
//Handle errors
} catch (SoapFault $E) {print_r($E-faultstring); }
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Re: [OSL | CCIE_Voice] Volume 1 Task 11.3
Also make sure to assign the 'Greetings Administrator' role to the subscriber/end user account. On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher wchatc...@gmail.com wrote: First you will need to configure some users in unity, and assign them as administrators on Call Handlers. When you hit the greeting administrator you will be prompted to enter your user ID and password, example 5002 and a vm password of 12345. Once you have been authenticated it will ask you to enter the number of the call handler you wish to change followed by #. After that just follow the prompts. On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Dear All, In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what would be testing result if I press 3 as the caller input. For caller input 3, the question says, option 3 should allow callers to modify and enable any greeting for the call handler (including Alternate Greetings) providing that the caller is the subscriber HQ Phone2 or BR1 Phone2. I tried to call 2123945000 from PSTN and pressed 3 as the caller input. It reaches the system callhander I created with extension as 5000. Since I have chosen for input 3, the conversation as Greetings Administrator, greeting administrator prompt is asking me to dial the call handler extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it is saying wrong call handler extension. It accepts only when I dial 5000, so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner? Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic understanding? Could you please someone explain, how the Greetings Administrator works. I could not find the testing or the verification in the solution guide. Thanks, Viki ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Volume 1 Task 11.3
Viki, Under the user menu you can assign different roles like system administrator or greetings administrator. Make sure to assign this greetings administrator role to the subscriber accounts, as well as making them the call handler owner. BR, Marty On Oct 21, 2013, at 5:53 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: I meant the owner of HQph2 and BR1ph2 as the call handler owner. On Monday, October 21, 2013, Vignesh Sethuraman wrote: Hello Martin and Bill, I have already assigned HQph2 and BR1ph2 as call handler owners, is this you mean as assigning the role or something else? Thanks, Viki On Monday, October 21, 2013, Martin Sloan wrote: Also make sure to assign the 'Greetings Administrator' role to the subscriber/end user account. On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher wchatc...@gmail.com wrote: First you will need to configure some users in unity, and assign them as administrators on Call Handlers. When you hit the greeting administrator you will be prompted to enter your user ID and password, example 5002 and a vm password of 12345. Once you have been authenticated it will ask you to enter the number of the call handler you wish to change followed by #. After that just follow the prompts. On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Dear All, In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what would be testing result if I press 3 as the caller input. For caller input 3, the question says, option 3 should allow callers to modify and enable any greeting for the call handler (including Alternate Greetings) providing that the caller is the subscriber HQ Phone2 or BR1 Phone2. I tried to call 2123945000 from PSTN and pressed 3 as the caller input. It reaches the system callhander I created with extension as 5000. Since I have chosen for input 3, the conversation as Greetings Administrator, greeting administrator prompt is asking me to dial the call handler extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it is saying wrong call handler extension. It accepts only when I dial 5000, so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner? Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic understanding? Could you please someone explain, how the Greetings Administrator works. I could not find the testing or the verification in the solution guide. Thanks, Viki ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Sent from IPhone -- Sent from IPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Volume 1 Task 11.3
Hi Bill, I thought it was required to manage the call handler greetings but I'd have to test or look up the docs to be 100% sure. Ill lab it up as well. Later, Marty On Oct 21, 2013, at 6:44 PM, Bill Hatcher wchatc...@gmail.com wrote: Marty, I've never add the specific role to the user before. I'll test it out tomorrow when I lab. Bill On Mon, Oct 21, 2013 at 5:29 PM, Martin Sloan martinsloa...@gmail.com wrote: Viki, Under the user menu you can assign different roles like system administrator or greetings administrator. Make sure to assign this greetings administrator role to the subscriber accounts, as well as making them the call handler owner. BR, Marty On Oct 21, 2013, at 5:53 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: I meant the owner of HQph2 and BR1ph2 as the call handler owner. On Monday, October 21, 2013, Vignesh Sethuraman wrote: Hello Martin and Bill, I have already assigned HQph2 and BR1ph2 as call handler owners, is this you mean as assigning the role or something else? Thanks, Viki On Monday, October 21, 2013, Martin Sloan wrote: Also make sure to assign the 'Greetings Administrator' role to the subscriber/end user account. On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher wchatc...@gmail.com wrote: First you will need to configure some users in unity, and assign them as administrators on Call Handlers. When you hit the greeting administrator you will be prompted to enter your user ID and password, example 5002 and a vm password of 12345. Once you have been authenticated it will ask you to enter the number of the call handler you wish to change followed by #. After that just follow the prompts. On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Dear All, In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what would be testing result if I press 3 as the caller input. For caller input 3, the question says, option 3 should allow callers to modify and enable any greeting for the call handler (including Alternate Greetings) providing that the caller is the subscriber HQ Phone2 or BR1 Phone2. I tried to call 2123945000 from PSTN and pressed 3 as the caller input. It reaches the system callhander I created with extension as 5000. Since I have chosen for input 3, the conversation as Greetings Administrator, greeting administrator prompt is asking me to dial the call handler extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it is saying wrong call handler extension. It accepts only when I dial 5000, so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner? Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic understanding? Could you please someone explain, how the Greetings Administrator works. I could not find the testing or the verification in the solution guide. Thanks, Viki ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Sent from IPhone -- Sent from IPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Volume 1 Task 11.3
Hello, I just checked this out and the greetings administrator role is required to manage the greetings but (!) if you assign the end user as a call handler owner, it automatically assigns the required role to the subscriber account. I had been assigning the role, then the ownership. Looks like I can save a step by just assigning ownership. Thanks for the time saver! Marty On Mon, Oct 21, 2013 at 7:06 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi Bill, I thought it was required to manage the call handler greetings but I'd have to test or look up the docs to be 100% sure. Ill lab it up as well. Later, Marty On Oct 21, 2013, at 6:44 PM, Bill Hatcher wchatc...@gmail.com wrote: Marty, I've never add the specific role to the user before. I'll test it out tomorrow when I lab. Bill On Mon, Oct 21, 2013 at 5:29 PM, Martin Sloan martinsloa...@gmail.comwrote: Viki, Under the user menu you can assign different roles like system administrator or greetings administrator. Make sure to assign this greetings administrator role to the subscriber accounts, as well as making them the call handler owner. BR, Marty On Oct 21, 2013, at 5:53 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: I meant the owner of HQph2 and BR1ph2 as the call handler owner. On Monday, October 21, 2013, Vignesh Sethuraman wrote: Hello Martin and Bill, I have already assigned HQph2 and BR1ph2 as call handler owners, is this you mean as assigning the role or something else? Thanks, Viki On Monday, October 21, 2013, Martin Sloan wrote: Also make sure to assign the 'Greetings Administrator' role to the subscriber/end user account. On Mon, Oct 21, 2013 at 3:15 PM, Bill Hatcher wchatc...@gmail.comwrote: First you will need to configure some users in unity, and assign them as administrators on Call Handlers. When you hit the greeting administrator you will be prompted to enter your user ID and password, example 5002 and a vm password of 12345. Once you have been authenticated it will ask you to enter the number of the call handler you wish to change followed by #. After that just follow the prompts. On Mon, Oct 21, 2013 at 12:21 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Dear All, In the CCIE Voice, IPX volume 1 task 11.3, I am unable to understand what would be testing result if I press 3 as the caller input. For caller input 3, the question says, option 3 should allow callers to modify and enable any greeting for the call handler (including Alternate Greetings) providing that the caller is the subscriber HQ Phone2 or BR1 Phone2. I tried to call 2123945000 from PSTN and pressed 3 as the caller input. It reaches the system callhander I created with extension as 5000. Since I have chosen for input 3, the conversation as Greetings Administrator, greeting administrator prompt is asking me to dial the call handler extension number. when I dial the HQ Phone2 or BR1 phone 2 extn number it is saying wrong call handler extension. It accepts only when I dial 5000, so how it relates to HQ phone 2 or BR1 Phone 2 as the call handler owner? Do i need to dial 5000 from HQ phone 2 or BR1 Phone 2 or am I missing basic understanding? Could you please someone explain, how the Greetings Administrator works. I could not find the testing or the verification in the solution guide. Thanks, Viki ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Sent from IPhone -- Sent from IPhone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME as SRST help.
I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: CME as SRST help.
Bill, Here's a great reference for CME MWI: http://ciscovoiceguru.com/518/cue-mwi-notification-methods/ I've used this a lot through my studies. @Ramcharan - good call. I think the symptom with that bug is that the phones will register but display no DN's and if you issue 'show ephone reg' it will show the DN's as 'invalid' or something like that. I have also hit the other bug with SC MGCP after coming out of SRST the dial-peers are still chosen on inbound calls from the PSTN. I believe the fix for that is to globally set 'voice hunt 2' and under the ephone-dn's assign a higher preference (I use 9). On Thu, Oct 17, 2013 at 11:53 AM, Bill Hatcher wchatc...@gmail.com wrote: Ok, I figured out my issue with the MWI not coming on in SRST more. Need to ass the key word unsolicited to the mwi-server command. Now to get the CME-SRST working. On Thu, Oct 17, 2013 at 10:36 AM, Bill Hatcher wchatc...@gmail.comwrote: Seifeddine, I've run that debug, but there is absolutly no output when I'm using CME-SRST. Bill On Thu, Oct 17, 2013 at 10:30 AM, Seifeddine Tlili seifeddine.tl...@lvs1.com wrote: Can you send the output of debug ephone register? ** ** Thx ** ** *Kindly*** * * *Seifeddine Tlili* [image: Description: Description: Long View Systems] M.Eng CCIE # 26440 Systems Consultant .. * Direct:* 403.387.3069 | *Mobile:* 403.973.4840 | *Main:* 403.515.6900 [image: Description: Description: Linkedin]http://www.linkedin.com/company/17908 [image: Description: Description: Twitter]http://twitter.com/LongViewSystems [image: Description: Description: Facebook]http://www.facebook.com/longviewsystems [image: Description: Description: Facebook]http://www.youtube.com/longviewsystems www.longviewsystems.com This message and any attached documents are only for the use of the intended recipient(s), are confidential and may contain privileged information. Any unauthorized review, use, retransmission, or other disclosure is strictly prohibited. If you have received this message in error, notify the sender immediately, and delete the original message. * *** ** ** ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Bill Hatcher *Sent:* Thursday, October 17, 2013 9:29 AM *To:* ccievoice *Subject:* [OSL | CCIE_Voice] Fwd: CME as SRST help. ** ** Marty, The weird thing is they work when I use call-manager-fallback. Looking at the cnf file, all seems well. ** ** On Thu, Oct 17, 2013 at 10:05 AM, Martin Sloan martinsloa...@gmail.com wrote: I had a similar issue recently which ended up being a DB replication problem. You could check the phones config file: 10.10.210.11:6970/SEP123456789123.cnf.xml Check subscribers copy of the file Right-click, select view source and search for 'srst' and see what it has there. I could be missing something but it looks like you have enough to get them registered. Marty ** ** On Thu, Oct 17, 2013 at 10:34 AM, Bill Hatcher wchatc...@gmail.com wrote: It’s not working!! Can anyone see something I may be doing wrong? My PRI and CUE register, I can even see SIP MWI being sent, but my phones will not register. They worked when I was using call-manager-fallback though so I know my SRST configuration is correct on the CallManager. telephony-service srst mode auto-provision all srst ephone description auto provisioned ephone : Oct 17 2013 14:47:07 srst dn line-mode octo max-ephones 4 max-dn 4 ip source-address 10.10.202.1 port 2000 max-conferences 12 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Oct 17 2013 16:07:18 ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] have the navigation links to the docs on cisco.com changed?
Hi Vir, Thanks for the heads up. It looks like they're shuffling things around which is a great added challenge for my lab in 2 weeks :-D You can find the docs here: Products - Voice and Unified Communications - Call Control - Mid-Market Call Control - Cisco Unified Communications Manager Express It's the same from there on. I'll re-post if I hit any others. Marty On Mon, Oct 14, 2013 at 10:59 AM, virajith vir...@rediffmail.com wrote: hi Guys, I was practicing the lab exercises today . When I was trying to configure BACD I went to the cisco product link for the doc on this location: http://www.cisco.com/cisco/web/psa/default.html?mode=prod And found that CME doc unders Voice unified communication IP Telephony Unified Communication Platform has been removed. 1) Is anyone else seeing this problem ? 2) Is the CME srnd provided to the candidates in the exam ? -Vir http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Get your own *FREE* website, *FREE* domain *FREE* mobile app with Company email. *Know More *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-1-10-13___cmp=hostlnk=sign-1-10-13nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] End User to Device Association
You could do a quick SQL query from the pub cli. I can't recall the table off hand but I will check it out when I get back to my computer. On Oct 12, 2013, at 10:47 AM, Ryan Maxam ryan.ma...@gmail.com wrote: Is there a quick and easy way to see which device an End User is associated with? Without having to run a report or going into the individual End User configuration. It is not an offered search under Find and List End User's. Thanks. Ryan Maxam Sent from my iPad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] End User to Device Association
Hey Ryan, So the query isn't super simple but it's definitely something you could memorize for a quick look at user/device associations. The table that holds the relationship between the user and device is the enduserdevicemap table but all the records for user and device are references to the pkid's of the primary table so you have to join those in, the enduser and device table, to get the friendly names. Here's the query: run sql select enduser.userid,device.name from enduserdevicemap inner join enduser on enduser.pkid = enduserdevicemap.fkenduser inner join device on device.pkid = enduserdevicemap.fkdevice The results would give you something like this: userid name == === SBPH2 SEP1234567891236 HQPH2 SEP123456789125 SBPH1 SEP123456789124 HQPH1 SEP123456789123 HTH Marty On Sat, Oct 12, 2013 at 11:27 AM, Martin Sloan martinsloa...@gmail.comwrote: You could do a quick SQL query from the pub cli. I can't recall the table off hand but I will check it out when I get back to my computer. On Oct 12, 2013, at 10:47 AM, Ryan Maxam ryan.ma...@gmail.com wrote: Is there a quick and easy way to see which device an End User is associated with? Without having to run a report or going into the individual End User configuration. It is not an offered search under Find and List End User's. Thanks. Ryan Maxam Sent from my iPad ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
Hi Ramesh, I also meant to ask for Ramcharan for some clarification on his advice but I saw another similar discussion where I think someone suggested the same work around (for a different issue). Correct me if I'm wrong but I think the solution is to use the translation pattern to prefix some identifying digits to the calling party number (something like ***) so that you can pick out that calling party number using the calling party transformation on the gateway port with a pattern like ***.! with DDI predot and settings to format the TON as needed. As for the VM port ENM, I think the recommendation is to use the pilot number for ports like VM and UCCX so that when AAR is invoked, the pilot number will be dialed and not the port number. Marty On Mon, Oct 7, 2013 at 9:27 PM, ramesh rameshdol...@rediffmail.com wrote: Hi Guys , Thanks for your inputs here. Hi Ramcharan, I am not fully understanding your suggestion here of using translation pattern. Would you be able to illustrate this with an example? Also my VM ports are set with the external mask of +1408202 . Do you guys recommend doing so? -Ramesh From: Ramcharan Arya ramcharan.a...@gmail.com Sent: Thu, 03 Oct 2013 06:41:54 To: Martin Sloan martinsloa...@gmail.com Cc: Justin Carney justin.s.car...@gmail.com, ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com, ramesh rameshdol...@rediffmail.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. Hi Marty, In order to preserve Original calling party TON you have to consider existing route pattern should not override so two possible ways to achieve this you can try to use a translation patter which evaluation prior to route pattern. Let us assume you prefix some additional character and create clng party x-formation pattern and DDI -predot what was prefix in TP and set appropriate plan and type. Use separate pt/css for clng party x-formation pattern. Another option is using application dial-rule can also use for this. Regards, Ramcharan Arya CCIE # 28926 (Voice/Routing Switching) On Wed, Oct 2, 2013 at 12:53 PM, Martin Sloan martinsloa...@gmail.comwrote: Yeah, I wasn't sure on that one either and had to test it out. I can't recall what the exact requirement, if any, for calling party TON was on the 'practice test' that I had with a similar task but I'm thinking the only way to properly set the calling TON would be with Xforms on the port level since it could be any number on the PSTN phone, even the number you're trying to dial out to from VM. They'd have to be very specific Xforms though since it could potentially override the current dial-plan manipulations in RL and RP if general masks are used like 10 X's. On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney justin.s.car...@gmail.comwrote: I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.com wrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515
Re: [OSL | CCIE_Voice] MVA confusion and quesiton
Hi Ramesh, Here's some answers based on my approach to configuring MVA. 1) I would use the 4 digit number for my dial-peer and CUCM MVA number (3300). Since you probably already have a translation-profile in place on the voice port or inbound dial-peer to chop the called number down to 4 digits, it makes sense to use that. 2) I don't change the calling party number and I use 'complete match' on the service parameter. I set my remote destination to that full number (either 7 or 10 digits). 3) No manipulation required, just set the remote destination to the full number. Marty On Sun, Oct 6, 2013 at 9:52 AM, ramesh rameshdol...@rediffmail.com wrote: Hi San, Thanks for your reply. 1) So you're suggestion is to use 3300 or 3033300 ? 2)At the dial-peer level are you using 3300 or 3033300? I way I use it is as given below : - = (a) If I use 3300 at the dial-peer level and on the callmanger as MVA number with 525 as the calling party number then I am able to have MVA functionality . (b) I normally call from the pstn using 3033300 from line 525( pstn phone) then on my h323 gateway I strip the called number to last 4 digits and send to the callmanger . (c) On the callmanger my MVA number is 3300. Are the above steps ( a to b ) correct? Regards, Ramesh From: san r luv...@gmail.com Sent: Sun, 06 Oct 2013 13:37:52 To: ramesh rameshdol...@rediffmail.com Subject: Re: [OSL | CCIE_Voice] MVA confusion and quesiton if you're stripping number for MVA , then mostly it wont work. Should use the exactly same number in Dial peer and CCM MVA configurations. I had the same issue in lab On Sat, Oct 5, 2013 at 8:12 PM, ramesh rameshdol...@rediffmail.comwrote: Hello Guys, I have the following questions for MVA. 1) I am following a 4 digit internal dial-plan for my site B phones and there is a requirement that I use 3033300 ( 7 digit number ) as my MVA number then can strip this 7 digit number to the last 4 digit number ( 3300 ) as my MVA number ? 2) Also my calling number is a 7 digit number coming from pstn as 525 then do I change it to 9525? 3) If incase calling number is a 10 digit number then It would come into site B as 972525 ( which is 10 digits) is manipulation required for this or can I just use the complete match with 10 digits on the service parameter level? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Get your own *FREE* website, *FREE* domain *FREE* mobile app with Company email. *Know More *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-1-10-13___cmp=hostlnk=sign-1-10-13nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? Get your own *FREE* website, *FREE* domain *FREE* mobile app with Company email. *Know More *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-1-10-13___cmp=hostlnk=sign-1-10-13nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] translation-rule
Hi Anthony, I'm not sure how to deep to go on the explanation but basically you have 2 capture groups in the 'match' string which are denoted by the parentheses, which have to be escaped by the backslash. These translations are based on the Unix Stream EDitor (SED) program and certain metacharaters need to be escaped to work properly, like the parentheses. They're called capture groups because whatever is included between the parentheses will be 'captured' to a buffer. You can then refer to it in the 'replace' string by referencing it's capture group number, which also has to be escaped with a backslash, like '\1'. In the *nix OS, you can create named capture groups so you can better identify the capture group and also insert new groups without having to update all others, but I don't believe this is possible in IOS. The '6' in your replace string is a literal 6. HTH Marty On Fri, Oct 4, 2013 at 1:30 PM, Anthony Nwachukwu anwachu...@apafrica.comwrote: I need with Translation -rule can someone help me explain the translation rule below. voice translation-rule 1 rule 1 /^\(12\)3\(45\)$/ /6\1\2/ · Set 1: 12 · Set 2: 45 · Ignore: 3 router#test voice translation-rule 1 12345 Matched with rule 1 Original number: 12345 Translated number: 61245 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] translation-rule
Wow. Great explanation! That was above and beyond. On Fri, Oct 4, 2013 at 3:23 PM, Justin Carney justin.s.car...@gmail.comwrote: I agree with Marty's response. I happen to be a visual learner, so if you are too then below is a your example marked up with colors to highlight the different parts of the rule. (Also, read this: http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml ) voice translation-rule 1 rule 1 /^\(*12*\)3\(*45*\)$/ /6\1\2/ Set 1: *12* Default set 0: 3 (note, if you have \0 in the replace string I'm not sure if that would carry over the 3 or the full match set 12345 - it would be worth testing) Set 2: *45* router#test voice translation-rule 1 12345 Matched with rule 1 Original number: 12345 Translated number: 6*12**45* Walking through this rule left to right... 1. rule 1 /[match string]/ /[replace string]/ 2. your match string is 12345, with no digits before 1 or after 5, broken up into 2 named sets as listed above in green (set 1) and blue (set 2). 3. your replace string is 6\1\2. 4. the 6 is a literal 6 and is the first digit of the translated number. 5. next is \1 - the \ means the next character is special, so don't use it literally (ie, it's not a 1 it is instead set 1). The match string already defined set 1 as *12* by using the \( to to start the set and \) to close the set. You don't specify a number for the set - working left to right the first set is \1 second is \2 and so on. (If you don't specify any sets using \( and \) then you still have a default set 0 called as \0 in the replace string which would be used to insert the entire match string.) 6. at this point your translated number is 6 *12* (plus the remaining string). 7. next and final part of the replace string is \2 which means set 2 8. in the replace string that means put in the contents of set 2 or * 45*. 9. your translated number is 6*12**45* *Further notes, if needed:* · The use of ^ means starts with so you only match a string * starting* with 12345. o Input 12345 = MATCH, output is 61245 o Input 012345 = NO match, output is unchanged 012345 · The use of $ means ends with so you won't match any additional digits, and your string cannot contain any more digits. o Input 12345 = MATCH, output is 61245 o Input 123456 = NO match, output is 123456 The combination of using ^ and $ in this case means only match literal 12345 with nothing before or after. if you remove both ^ and $ you could match 99912345000 and get the output 99961245000. Hope this helps. If it doesn't, read the link at the top :-) On Fri, Oct 4, 2013 at 2:03 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi Anthony, I'm not sure how to deep to go on the explanation but basically you have 2 capture groups in the 'match' string which are denoted by the parentheses, which have to be escaped by the backslash. These translations are based on the Unix Stream EDitor (SED) program and certain metacharaters need to be escaped to work properly, like the parentheses. They're called capture groups because whatever is included between the parentheses will be 'captured' to a buffer. You can then refer to it in the 'replace' string by referencing it's capture group number, which also has to be escaped with a backslash, like '\1'. In the *nix OS, you can create named capture groups so you can better identify the capture group and also insert new groups without having to update all others, but I don't believe this is possible in IOS. The '6' in your replace string is a literal 6. HTH Marty On Fri, Oct 4, 2013 at 1:30 PM, Anthony Nwachukwu anwachu...@apafrica.com wrote: I need with Translation -rule can someone help me explain the translation rule below. voice translation-rule 1 rule 1 /^\(12\)3\(45\)$/ /6\1\2/ · Set 1: 12 · Set 2: 45 · Ignore: 3 router#test voice translation-rule 1 12345 Matched with rule 1 Original number: 12345 Translated number: 61245 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar Ganesha offers Company email website (FREE) at your own domain (FREE) - KNOW MORE ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
Hi Ramesh, Just to make sure we're on the same page, are you setting the CUCM service parameter: Display Original Calling Number on Transfer from Cisco Unity = True If you are setting this, can you explain in a little more detail the call flow and outcome? Thanks, Marty On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.com wrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.comwrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers the option to press 9 and ring his cell phone number at 9515 ( on the pstn) when the call gets forwarded to voicemail I have made the on hook transfer on the service parameter level on the callmanger to true and have update the users caller input option 9 transfer to extn to 9515111. However would like to know if this is enough ? also how do I preserve the orginal calling party type and plan? Is there a service parameter or an easy route to use in the unity connection server? -Ramesh Dollar http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/signatureline.htm@Middle? *Ganesha offers* Company email website (*FREE*) at your own domain (* FREE*) - *KNOW MORE *http://track.rediff.com/click?url=___http://businessemail.rediff.com/company-email-hosting-services?sc_cid=sign-060913___cmp=hostlnk=sign-060913nsrv1=host
Re: [OSL | CCIE_Voice] unity connection - transfer option for users.
Yeah, I wasn't sure on that one either and had to test it out. I can't recall what the exact requirement, if any, for calling party TON was on the 'practice test' that I had with a similar task but I'm thinking the only way to properly set the calling TON would be with Xforms on the port level since it could be any number on the PSTN phone, even the number you're trying to dial out to from VM. They'd have to be very specific Xforms though since it could potentially override the current dial-plan manipulations in RL and RP if general masks are used like 10 X's. On Wed, Oct 2, 2013 at 1:21 PM, Justin Carney justin.s.car...@gmail.comwrote: I wasn't sure RDNIS would matter here but figured I would throw it out there anyway (as it applies when redirecting TO CUC). It seems the unity service parameter mentioned earlier obviates the need to use RDNIS. With the option you proposed on creating a new RP/RL just for this requirement I would just set the digit manipulation/TON on the RL to whatever you see inbound from that specific PSTN ANI to HQ - unless the question told you what the expected outbound ANI/TON should be. Another option would be to compare the original PSTN number with the destination PSTN and set to local if same NPA, LD if different NPA, or international different country codes. If it comes in unknown/unknown then send it back out that way. On Wed, Oct 2, 2013 at 12:48 PM, Martin Sloan martinsloa...@gmail.comwrote: The RDNIS shouldn't be a factor here. I just labbed this up and there is no Redirecting Number IE in the ISDN messages for this scenario. It's more of a straight dial from Unity. I think the places to be checked are: CUCM service parameter Call Routing Path Whatever Route Pattern - Route List is being used needs to have the Use Calling Party's External Phone Number Mask checked and no masking being done below, like truncating the calling party number to 7 digits if that was part of the requirement for the sites local PSTN dialing. I recommend partitioning out a new pattern that matches the number you're trying to dial and handling the digit manipulation separately from the rest of the dial plan to keep it conceptually simple, but not necessarily 'cleaner'. Kind of along the lines of keeping AAR, CFUR, SNR separate. As for the calling party TON on this, your guess is as good as mine. If the task doesn't specifically ask to set the calling party TON and it says to use any line from the PSTN phone, what do you do? Marty On Wed, Oct 2, 2013 at 12:14 PM, Justin Carney justin.s.car...@gmail.com wrote: What do you see on the voice gateway for ANI/DNIS of the two separate calls - inbound (PSTN to Phone (which is then CFNA to CUC)) and outbound (CUC to PSTN alt dest 9515)? Take a look your gateways settings for Redirecting Number IE Delivery (RDNIS) for both inbound/outbound. These checkboxes are adjacent to the Display IE Delivery (which is usually turned on). To test and understand the behavior of these settings I would recommend ticking these boxes on/off and retrying your inbound/outbound calls in this (and other) scenario. As a test try setting up a call such as PSTN SA phone CFA to a different PSTN number and look at the q931 debugs for ANI/DNIS/RDNIS. I haven't tested this recently and not sure if it applies in your stated scenario but try checking the box on SA gateway for the outbound RDNIS. This should allow CUC to send 3 IE out to the PSTN - the original ANI (PSTN caller), the redirecting number/RDNIS (would expect this to be either the DN of original called IP phone or possibly the vm port/pilot's DN/EPNM) and lastly the DNIS should be 9515. For a different scenario with SB in SRST - when a call to a SB phone does CFNA to vm you will send it out the PSTN to HQ vm pilot and on SA gw you need to allow (check) the inbound RDNIS. In this case the IE at SA router is ANI-original caller, DNIS-DID of vm pilot, RDNIS-the SB phone's DN. -Justin On Wed, Oct 2, 2013 at 10:56 AM, ramesh rameshdol...@rediffmail.comwrote: hi Martin, I have done that ( preserve original calling number ) in unity however it does not preserve the correct calling number for pstn callers . Also the type and plan of the called and calling numbers are messed up. Any other steps we can take? From: Martin Sloan martinsloa...@gmail.com Sent: Wed, 02 Oct 2013 18:39:21 To: ramesh rameshdol...@rediffmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] unity connection - transfer option for users. I can't remember the name but if you go to ccm service parameters and search for 'unity' you'll hit the parameter to preserve the calling number. On Oct 2, 2013, at 9:02 AM, ramesh rameshdol...@rediffmail.com wrote: hello Guys, How do I preserve the original calling number for a call made to a user on unity connection. The idea is to give callers
Re: [OSL | CCIE_Voice] Presence - on hook and off hook status
Hi MJ, Is the end user assigned on the line level of the hard phone? That assignment is unique per line appearance so if you make the association on the CUPC device it does not automatically populate to the hard phone/any other line appearance. When the phone goes off-hook CUCM checks the end user assignment for that appearance and if there is an end user assigned it check whether that end user is assigned CUP licensing to decide if the publish message is sent over the CUPS SIP trunk. BR, Marty On Tue, Oct 1, 2013 at 10:29 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello all, I have configured presence and both softphone and deskphone modes , IM and voicemail is working fine on the clients However I have a question when I lift the handset of the phone ( hard phone ) that is assoicated with the CUPC clients . I see that the presence status does not show On the phone and does not turn yellow. I have tried reseting my sip trunk pointing to the presence server yet I see the same issue. Please let me know what can be done to fix this ? Also is this a major issue ? -MJ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP
I'm really disappointed as well. I just failed my second attempt on Wed and was worried about getting a 4th try in when I logged on to see no seats left for a 3rd! I figured it would get tight but this is nuts. I made a big improvement on my score from the first try and feel like the third time could have been the charm. Oh well. On Fri, Sep 27, 2013 at 11:35 AM, Dane Warner dwar...@epochuniversal.comwrote: There are no open dates in either San Jose or RTP anymore, period. ** ** Looks like if we want to take the Voice exam, which I’m sure Cisco doesn’t want us to do anymore, then it’s either Tokyo or we’re SOL. ** ** Very disappointing. ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *[image: Epoch_Logo_Smaller_Transparent]* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *OSL StudyList *Sent:* Friday, September 27, 2013 3:19 AM *To:* Josh Petro *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP ** ** Do you know what times the lab dates are released for those who have not paid? I thought it was at midnight SJC time but, I am not sure. ** ** On Fri, Sep 27, 2013 at 5:17 AM, Josh Petro josh.pe...@gmail.com wrote:* *** If you mean Voice availability, then you are correct in that RTP is filled. San Jose had a few open spots in Jan Feb last week. I don't believe Collaboration dates are open yet for scheduling. Josh On Sep 27, 2013 5:58 AM, OSL StudyList collaboration.c...@gmail.com wrote: Is anyone having any luck scheduling exams at RTP or SJC? When I try to find an available date, I am seeing NOTHING available. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com image001.png___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP
Hey Alex, I hear ya. I went through all locations and checked availability and Tokyo is the closest for me, which is about an 18 hour flight. I've priced it all out. I'm on the fence a bit about traveling there but at this point I'm leaning toward not. On top of the additional expense for travel and time away from family, I'm paid by the hour so it would be at least 4 days unpaid for me. It starts to add up. Like everyone else I've invested a lot of money into this and I'm starting to get a little gun shy on putting up another couple thousand dollars for something that's not guaranteed. It could be money straight down the toilet. At a certain point, enough is enough. Good luck on your lab in RTP in Feb! Marty On Fri, Sep 27, 2013 at 12:17 PM, Alex Mendoza aa.mend...@icloud.comwrote: As Dave says, you can book at Tokyo or other location. I'm from Mexico and can book at RTP in february just one week ago. More pressure because will be my 2nd and last attempt. If you are so close to get your CCIE, look for a seat at other location even if you must pay for travel expenses. All my best for the last candidates. best regards Alex On Sep 27, 2013, at 10:57 AM, Martin Sloan martinsloa...@gmail.com wrote: I'm really disappointed as well. I just failed my second attempt on Wed and was worried about getting a 4th try in when I logged on to see no seats left for a 3rd! I figured it would get tight but this is nuts. I made a big improvement on my score from the first try and feel like the third time could have been the charm. Oh well. On Fri, Sep 27, 2013 at 11:35 AM, Dane Warner dwar...@epochuniversal.comwrote: There are no open dates in either San Jose or RTP anymore, period. ** ** Looks like if we want to take the Voice exam, which I’m sure Cisco doesn’t want us to do anymore, then it’s either Tokyo or we’re SOL. ** ** Very disappointing. ** ** *Dane Warner, CCVP* *Sr. Network Engineer* *Epoch Universal, Inc.* *(909)226-0755* *dwar...@epochuniversal.com * *image001.png* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *OSL StudyList *Sent:* Friday, September 27, 2013 3:19 AM *To:* Josh Petro *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab exam date availability at SJC and RTP ** ** Do you know what times the lab dates are released for those who have not paid? I thought it was at midnight SJC time but, I am not sure. ** ** On Fri, Sep 27, 2013 at 5:17 AM, Josh Petro josh.pe...@gmail.com wrote: If you mean Voice availability, then you are correct in that RTP is filled. San Jose had a few open spots in Jan Feb last week. I don't believe Collaboration dates are open yet for scheduling. Josh On Sep 27, 2013 5:58 AM, OSL StudyList collaboration.c...@gmail.com wrote: Is anyone having any luck scheduling exams at RTP or SJC? When I try to find an available date, I am seeing NOTHING available. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] MVA the right way to configure it
Hi MJ, 1) If you set the partial match to 7 digits and then configure your remote destination as a 10 digit number, you'll get a match if the ANI is either 7 or 10 digits since the match rule takes 'X' partial-match digits from the RD starting with the last number (2 in this case) and compares it to the ANI of the calling number, *but* the calling party number must be equal to or shorter in length than the configured remote destination, which is why it's good to just set your RD at 10+ digits if you're using partial match. Here are some scenarios and the outcome for partial match: Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 972525 Calling Party Number = 525 Result = *Match* Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 972525 Calling Party Number = 972525 Result = *Match* Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 525 Calling Party Number = 525 Result = *Match* * * Partial Match = True Number of Digits For Match = 7 digits Remote Destination = 525 Calling Party Number = 972525 Result = *No* *Match (ANI is longer than RD)* When using Complete match, the ANI and RD have to be exactly the same. I like to make a call into SB from the PSTN phone prior to configuring SNR and I can quickly see what the ANI is, which is what I then make my RD. I had mentioned some buggy behavior with SNR though I never spent time working with partial match since when I heard about that issue I just stuck with complete match but I wanted to test my info above to make sure I wasn't sending incorrect info. It wasn't too hard to run into this buggy behavior. I found a workaround as well so I thought I'd share. When changing the Complete Match service parameter to Partial Match you get a screen pop that says to remember and set the Number of Digits for Caller ID Partial Match service parameter. The default for that parameter is 10 and the bug that I found is that on the initial change from default 10 to 7, the new setting does not take effect. After changing from 10-7 I started to make test calls and my CLID to SB PH1 was showing as the 7 digit ANI of the PSTN phone and not SB PHONE 2 3002 like it should. I dug around for a bit and tweaked a couple parameters and re-tested. The deal is that you have change Complete Match to Partial Match - Save then change Partial Match digits from 10 to 7 and Save again. 2) For this one if your service parameter is set to Complete Match and your ANI is 7 digits, just set your RD to the 7 digit number then use route patterns/xlations to manipulate as needed. 3) Not sure about that one. I've definitely seen conflicting information on certain things but I've realized that some of the training material is years in the making and when things are discovered or updated, maybe the old information is not or it's just floating out there. I can confirm that based on some recent experience with trusted trainers it was reiterated not to use partial match, maybe in part because of the issue that I hit today. Marty On Tue, Sep 24, 2013 at 8:19 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hi Guys , Thanks a lot for taking time out to reply to my question. It was really helpful. I was trying to understand the difference between full match with 10 digits and partial match with 7 digits. Here are my scenarios... 1) If I use partial match with 7 digits then this will satisfy the condition where my calling number is 7 digits ( in this instance it is 525) but what happens if my calling number is in the form 972525 in this case it is 10 digits whereas my service parameter indicates just 7 digits ? 2) If I use complete match with 10 digits then will satisfy the condition where my calling number is 10 digits but not when 7 digits . I am not sure where complete match means it includes the condition of the calling number with 7 digits as well. Would you be able to throw some light on this? 3)In some of the IPexpert walk through videos I see the instructor seems to prefer partial match with 7 digits . However this may be for a specific condition. I am I correct on this ? MJ On Wed, Sep 18, 2013 at 8:55 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi MJ, I did some research on this since I've been configuring MVA for a while but have had some questions about underlying architecture. Here's some responses to your info plus some of my findings. 1) If the MVA DID is in line with your standard DID range for the site, why not just piggy back on the existing CUCM dial-peers instead of creating a new one just for MVA. Say Site B for example with a 3XXX extension range, you could use the CUCM dial-peer: dial-peer voice 3000 voip destination pattern 3...$ session target ipv4:10.10.210.11 no vad voice-class codec 1 voice-class h323 1 dtmf-relay h245-alpha incoming called
Re: [OSL | CCIE_Voice] BACD Timer
I'm digging up an old one here but I just ran into an issue with this B-ACD parameter and I wasn't able to find the answer online so I thought I'd share in case it pops up in a search. From the info I gave above from the Cisco B-ACD documentation, the *param **max-time-call-retry *parameter *could* have a minimum value of 30 but when I was setting that param to 30 I still wasn't getting sent to my final destination (*param voice-mail*) at 30 seconds, but at 66. I did a 'debug voip application script' and got these messages as soon as I was dropped in queue: Sep 24 01:08:38.733: //8//TCL :/tcl_PutsObjCmd: TCL AA: ++ max-time-call-retry is set to less than minimum allowed value of 60 ++ Sep 24 01:08:38.733: //8//TCL :/tcl_PutsObjCmd: TCL AA: ++ Setting max-time-call-retry to minimum value of 60 ++ Sep 24 01:08:38.737: //8//TCL :/tcl_PutsObjCmd: TCL AA: -- Valid mandatory parameter max-time-call-retry = 60 -- Looks like the minimum value is actually 60 based on the embedded B-ACD script in my IOS c2800nm-adventerprisek9_ivs-mz.124-24.T8.bin. Marty On Mon, Jun 17, 2013 at 7:13 PM, Martin Sloan martinsloa...@gmail.comwrote: I got the info below from this guide - http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 It has good examples you can copy/past/edit. I believe 'Call-Queue and AA Tcl Scripts in Flash Memory: Example' is the best one to use. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Step 29 *param* *max-time-call-retry* *seconds * Example: Router(config-app-param)# param max-time-call-retry 700 (Optional) Sets the maximum amount of time for the call-retry timer. This is the maximum period of time for which a call can stay in a call queue and retry to connect with a hunt group before the call is sent to an alternate destination number. •*seconds*—Maximum period of time, in seconds. The range is from 30 to 3600. The default is 600. On Mon, Jun 17, 2013 at 5:11 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: hi experts, I am trying to understand timer for *param* *max-time-call-retry can anyone share there knowledge about how does it effect the bacd script and the purpose of this field* * * *Thnks* * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] OWLE lab 4 - Voicemail issue
Thanks for the info Brian. If anyone experiences the ICD extension not
showing up on lab day, you can get the SQL statement from a doc that's on
the same page as where you'd get the IPPA service URL doc:
Voice and Unified Communications-Customer Collaboration-Cisco Unified
Contact Center Products-Cisco Unified Contact Center
Express-Configuration Examples and TechNotes
Just search for 'ICD E' instead of 'one button'. The SQL statement is at
the end of the doc but you do have to change the 'paramvalue=F' to
'paramvalue=T
Marty
On Sat, Sep 21, 2013 at 7:28 PM, VanBenschoten, Brian
brian.vanbenscho...@corebts.com wrote:
To fix a bug with Unity Conn:
SSH to CUC
run cuc dbquery unitydirdb EXECUTE PROCEDURE csp_SmtpAddressMigrate('
proctorlabs.com','cuc7-pub')
To fix the IPCC Extension not showing/ Not installed:
SSH into CUCM PUB
run sql update processconfig set paramvalue=T where paramname like
IAQInstalledFlag
___
Brian Van Benschoten - CCIE # 5421
Managing Consultant - Unified Communications
Core BTS - North Central Region
3001 West Beltline Highway
Madison, WI 53713 USA
(P) +1 (608) 661-7780
(F) +1 (608) 661-7701
brian.vanbenscho...@corebts.com
www.corebts.com
[image: BV-CustomCoreSmall1]
*From:* ccie_voice-boun...@onlinestudylist.com [mailto:
ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Karen Johnson
*Sent:* Sunday, September 15, 2013 12:56 PM
*To:* ccie_voice@onlinestudylist.com
*Subject:* [OSL | CCIE_Voice] OWLE lab 4 - Voicemail issue
hi folks.
in OWLE lab 4, i can't get the VM when I left the messasge from PSTN
phones to HQ and SB phones.
any parameter to check ?
K
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For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
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Re: [OSL | CCIE_Voice] CUE License Installation Issue
Hi Hesham, Any chance this is a QoS issue like FRTS applied on the HQ WAN interface but no map-class applied to the SB sub-interface so traffic is at default 56k? Maybe try to do a copy tftp flash of the file from the SB router itself eliminate a step in between. Later, Marty On Fri, Sep 20, 2013 at 3:04 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, I have been trying to install the CUE License and till last week CUE License for CME was working perfectly now when I try to install any license whether CCME or CCM I get the following error Error: Download error Can not download cue-vm-license_25mbx_cme_7.0.3.pkg error code 150 : error type 'Operation too slow. Less than 50 bytes/sec transfered the last 30 seconds software install clean url ftp://142.100.64.14/cue-vm-license_25mbx_cme_7.0.3.pkg username heathrow password heathrow I have tried 2 different machines the UCCX VM as well as my candidate machines some time I get this error operation too slow and another error I have tried to reload the CUE many times. I am using FreeFTPd and I created a totally new accoutn still didn't work I reset the CUE still the problem exists. Reloaded the router itself many times still no chance. Tried another files same version to check if the file is corrupted still no chance. Please share your thought. Many Thanks, Hesham ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] dhcp one scope two manual assignments
Here is an outstanding walk through on this topic. I've adopted the tcl script approach which can be very quick once you practice a couple times. http://www.ucguerrilla.com/2013/05/ccie-voice-tactical-dealing-with-ios.html HTH Marty On Mon, Sep 16, 2013 at 12:21 PM, @ Mitchell andre...@gmail.com wrote: Excuse me if this has been asked before as I did not find it in my search. Is there a way to manually assign addresses to two phones but within one dhcp pool from the iOS standpoint? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] UCCX fresh install
Hi Hugo, Administrator/ciscocisco Marty On Wed, Sep 11, 2013 at 4:41 PM, Barrera, Hugo hugo.barr...@nexusis.comwrote: What is the default username and passwd if you have to do the “fresh install” workaround? ** ** *Hugo * ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cue + callmanger srst problem
Hello, You can get that message if the SIP trigger is enabled for SRST but for some reason the voicemail application isn't. Login to the CUE via CLI and check that your SIP trigger is pointing to the voicemail application and also do a 'show ccn application' to check the status of the voicemail application. Guessing from your error, it might not be enabled. Marty On Tue, Sep 10, 2013 at 10:27 AM, probert...@gmail.com probert...@gmail.com wrote: Hi, *I'm sorry*, *we* are currently experiencing system problems and are *unable to process your call * Is usually played by UCCX I have never heard it from CUE. Try factory reset on CUE just to make sure there is nothing wrong with it. On Tue, Sep 10, 2013 at 7:54 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello Guys, Still waiting any update on this ? On Mon, Sep 9, 2013 at 4:22 PM, sanity insanity networksanitytoinsan...@gmail.com wrote: hi Guys, In the normal mode when wan is up I can call into the cue ( on site c ) through jtapi . However when the wan link breaks and the when my site c router and phones fall into srst and then try placing calls to the cue using sip dial peer I hear the following prompt - *I'm sorry*, *we* are currently experiencing system problems and are *unable to process your call * *I have checked everything in the setup and unable to figure out what the problem is . Has anyone seen this ? * *-MJ * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cue + callmanger srst problem
FWIW - I know I'm also used to getting that message from UCCX but you will get it from CUE as well for reasons like the one I mentioned above. I tested this in my lab to confirm and if you'd like to try, just disable the voicemail application in CUE while using the SIP trigger that points to it. It will ring out several times and then play the message which the OP reported. The fact that it's getting to CUE and the error prompt is being played eliminates much of the troubleshooting outside of CUE. Also, if you'd like to confirm for sanity sake that the call is in fact hitting CUE and not UCCX, just debug ccsip messages on the SC router and watch the call route to CUE. On Tue, Sep 10, 2013 at 12:32 PM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: First of all, How is your SiteC Router CUE? Is it originally MGCP Gateway integrated with CUCM and the CUE is integrated with CUCM or CME? If that happens , The only way I could think of that your CTI Route Point 85224044220 has mistakenly configured with Call Forward Unregister to the UCCX Pilot 4000 You have to check carefully the CTI RP for UCCX Trigger as well as the CUE there could be somesort of typo error caused that. Also make sure on SiteC Gateway you have that config application global service alternate default ccm-manager fallback-mgcp voice translation-rule 8 rule 1 /^\*/ // voice translation-profile vmredirect translate redirect-called 8 dial-peer voice 4220 voip destination-pattern 42..$ session protocol sipv2 session target ipv4:142.1.66.253 dtmf-relay sip-notify codec g711ulaw vad translation-profile out vmredirect Make sure you have that config on the CUE ccn subsystem sip gateway address 142.1.66.254 mwi sip unsolicited end subsystem ccn trigger sip phonenumber 4220 application voicemail enabled maxsessions 6 end trigger Also make sure the LO0 is routed properly and pingable from any router to CUE and from CUE to all your network Int lo0 ip ospf network point-to-point On 10 September 2013 08:37, Martin Sloan martinsloa...@gmail.com wrote: Hello, You can get that message if the SIP trigger is enabled for SRST but for some reason the voicemail application isn't. Login to the CUE via CLI and check that your SIP trigger is pointing to the voicemail application and also do a 'show ccn application' to check the status of the voicemail application. Guessing from your error, it might not be enabled. Marty On Tue, Sep 10, 2013 at 10:27 AM, probert...@gmail.com probert...@gmail.com wrote: Hi, *I'm sorry*, *we* are currently experiencing system problems and are *unable to process your call * Is usually played by UCCX I have never heard it from CUE. Try factory reset on CUE just to make sure there is nothing wrong with it. On Tue, Sep 10, 2013 at 7:54 AM, sanity insanity networksanitytoinsan...@gmail.com wrote: Hello Guys, Still waiting any update on this ? On Mon, Sep 9, 2013 at 4:22 PM, sanity insanity networksanitytoinsan...@gmail.com wrote: hi Guys, In the normal mode when wan is up I can call into the cue ( on site c ) through jtapi . However when the wan link breaks and the when my site c router and phones fall into srst and then try placing calls to the cue using sip dial peer I hear the following prompt - *I'm sorry*, *we* are currently experiencing system problems and are *unable to process your call * *I have checked everything in the setup and unable to figure out what the problem is . Has anyone seen this ? * *-MJ * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?
Let me just say that I love this thread! @Laksh about Asterisk, from my experience you'll be hard pressed to find anything (non-proprietary) that Cisco UC can do that Asterisk cannot. Complex dial plans, feature rich VM, native call recording, mobility, etc - Asterisk can do it all straight out of the box. That being said I only use Asterisk to fill in gaps when there is something that Cisco UC can't do easily or without costing a small fortune, since Asterisk can do it for free. Being an open source platform, if the feature doesn't exist you can code it yourself. I've never deployed it as an overall solution but just as a tool to fix a problem. I know there are some large(ish) SP's using Asterisk like SIP-UA, so I believe it has the ability to scale although I can't attest to that myself. In comparing reliability, there have been some kludge versions of CUCM out there as well so depending on who you talk to about Asterisk, you might get mixed results. I have never had a problem with it's reliability, outside of problems I've caused myself :-) If you're interested in a nice introduction to Asterisk without having to use the somewhat cryptic config files, download Elastix and deploy as a VM. It runs on CentOS with a GUI and it's really straight forward to setup. Use 'Elastix without tears' as a guide, although it's a little dated 95% of the info is accurate. You can get a free SIP trunk to the cloud using SIP-UA. I think Asterisk and it's soft-switch cousin FreeSwitch are going to become more and more popular. I've personally spoken with 3 tech start-up companies that are providing web-based telephony services using FreeSwitch ( https://www.speek.com http://anymeeting.com http://www.voysee.com) and I'm sure there are many more out there on the rise. Just like moving from a CO where an operator was physically plugging in cables to connect calls all the way up to our current IP infrastructure, the industry continues to change and advance so it's up to us to stay relevant. That's the thing I like most about Telephony/VoIP/UC/Collaboration is that even though it continues to evolve and update, until humans start using ESP to communicate it's going to remain absolutely necessary, which means (hopefully) a job for us! Marty On Thu, Aug 29, 2013 at 7:30 AM, Drake J jdrake...@gmail.com wrote: hi Laksh, Thanks for your inputs here.This was a good discussion. It is always good for us to all know about things that happen outside . Talking about Telco OTTs we can already see few of the Telcos have come out with Webrtc solutions for enterprise and service providers . Check this video out too depicting their solution... http://www.youtube.com/watch?v=Nz-BQZMp3sk Most of these applications written on software are supposed to open source and left for the users to customize . No real networking staff expertise required just download the SDK/API and customize and no more complex network topologies in future. Also no licensing fee too . Hence a real killer of techology in the future most likely we will see a wide spread of this starting 2014 if all predictions are to be believed. Hope someone from any of the TELCOs on this alias can add a few comments as well. Thanks once again for your inputs everyone. On Wed, Aug 28, 2013 at 11:05 PM, Lakshmish NS lakshmish...@gmail.comwrote: Hi Drake, I totally understand your concern, I'd be worried too. Having said that, we should always update ourselves with the latest technology. However, in future I believe Asterisk might be able to give tough run to Cisco UC. Not sure though, I hear stories that it is unstable and featureless compared to CUCM. I hope if someone aware of Asterisk would help us out here. Regard, Laksh On Wed, Aug 28, 2013 at 9:56 PM, Drake J jdrake...@gmail.com wrote: Hi Guys, Thanks for your responses I see u guys have empathized on call routing and and UC hardware for backend deployments. However Telco OTTs are coming up with directly provide these services over the cloud . Here is a disruptive analysis : http://www.slideshare.net/deanb/disruptive-analysis-web-rtc-overview-april-2013 Anyways, this might be not be so serious afterall . Just thought of brainstorming . Thanks guys for your responses again. On Tue, Aug 27, 2013 at 6:20 PM, Lakshmish NS lakshmish...@gmail.comwrote: Didn't have time to go through the video, I believe WebRTC is nothing but a Protocol, similar to SIP, H.323. Moreover, this protocol would only appeal to the Web audience, just like Skype, or Google talk. You still need to use UC hardware and their design for enterprise deployments. I mean we don't use Google talk and Skype in companies right? SIP is open source, but still Cisco uses it. As FAQ's suggest WebRTC is an open framework for the web that enables Real Time Communications in the browser. If only UC was that easy that could be implemented through browser, we
Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?
Hi Drake, That's an interesting point. I've definitely heard of Microsoft creating such road blocks and I know first hand that Internet Explorer loves to be different when it comes to web programming. Here's a funny take on MS (with a measure of truth): A thirty-two bit extension and graphical shell to a sixteen-bit patch to an eight-bit operating system originally coded for a four-bit microprocessor which was written by a two-bit company that can't stand one bit of competition. I've spent some time in Cisco's Jabber CAXL library which is about 40k lines of Javascript and while it's not truly WebRTC, they're really pushing for developers to integrate voice/video/im into the browser. From what I know of Cisco, they seem to typically be in support of standardizations and often incubate technology and then release for standardization. I could be way off on that one so feel free to disagree. Browser support is a big issue too since only the modern browsers are supporting it, it's going to be a while before the desktop/laptop world is fully ready but since mobile devices are refreshed so often they're typically pretty up to date and should support it now and if not soon. The bottom line for me is that competition breeds innovation so any company that blocks legitimate advances to protect their own profits will ultimately fail. Karma has no menu, you get served what you deserve! Marty On Thu, Aug 29, 2013 at 12:21 PM, Drake J jdrake...@gmail.com wrote: Hello Martin, Thanks for your inputs. Food for thought - the UC vendors otherwise rivals when it comes to competition seem to team up against Open source projects in the World Wide Web Consortium ( W3C) and keep causing roadblocks in the standardization of Webrtc. Why? it seems like it threatens their own products . However open source communities such as Mozilla are fighting hard to push this through. The Future definitely has a lot in store for IP Telephony. On Thu, Aug 29, 2013 at 7:20 PM, Martin Sloan martinsloa...@gmail.comwrote: Let me just say that I love this thread! @Laksh about Asterisk, from my experience you'll be hard pressed to find anything (non-proprietary) that Cisco UC can do that Asterisk cannot. Complex dial plans, feature rich VM, native call recording, mobility, etc - Asterisk can do it all straight out of the box. That being said I only use Asterisk to fill in gaps when there is something that Cisco UC can't do easily or without costing a small fortune, since Asterisk can do it for free. Being an open source platform, if the feature doesn't exist you can code it yourself. I've never deployed it as an overall solution but just as a tool to fix a problem. I know there are some large(ish) SP's using Asterisk like SIP-UA, so I believe it has the ability to scale although I can't attest to that myself. In comparing reliability, there have been some kludge versions of CUCM out there as well so depending on who you talk to about Asterisk, you might get mixed results. I have never had a problem with it's reliability, outside of problems I've caused myself :-) If you're interested in a nice introduction to Asterisk without having to use the somewhat cryptic config files, download Elastix and deploy as a VM. It runs on CentOS with a GUI and it's really straight forward to setup. Use 'Elastix without tears' as a guide, although it's a little dated 95% of the info is accurate. You can get a free SIP trunk to the cloud using SIP-UA. I think Asterisk and it's soft-switch cousin FreeSwitch are going to become more and more popular. I've personally spoken with 3 tech start-up companies that are providing web-based telephony services using FreeSwitch ( https://www.speek.com http://anymeeting.com http://www.voysee.com) and I'm sure there are many more out there on the rise. Just like moving from a CO where an operator was physically plugging in cables to connect calls all the way up to our current IP infrastructure, the industry continues to change and advance so it's up to us to stay relevant. That's the thing I like most about Telephony/VoIP/UC/Collaboration is that even though it continues to evolve and update, until humans start using ESP to communicate it's going to remain absolutely necessary, which means (hopefully) a job for us! Marty On Thu, Aug 29, 2013 at 7:30 AM, Drake J jdrake...@gmail.com wrote: hi Laksh, Thanks for your inputs here.This was a good discussion. It is always good for us to all know about things that happen outside . Talking about Telco OTTs we can already see few of the Telcos have come out with Webrtc solutions for enterprise and service providers . Check this video out too depicting their solution... http://www.youtube.com/watch?v=Nz-BQZMp3sk Most of these applications written on software are supposed to open source and left for the users to customize . No real networking staff expertise required
Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?
I definitely prefer a physical phone to a soft phone! Kind of a bit off topic, have you guys seen this: http://www.shoretel.com/about/newsroom/press_releases/New_ShoreTel_Dock_Transforms_iPad_and_iPhone_Into_Desk_Phone_.html I was just telling my buddy how Cisco had such a great idea with the Cius but missed out by trying to create their own tablet, and then I see an advertisement for this. If Cisco had only provided the dock for and already super competitive tablet/smartphone market, it would have been brilliant! I'm surprised Shoretel seems to be the only company that sees the opportunity here. Vendors can keep making money on hardware but provide a unified client experience across all platforms (Jabber). It's the best of both worlds! On Thu, Aug 29, 2013 at 4:17 PM, Michael Davis michaeldavis1...@yahoo.comwrote: No matter what, there will ALWAYS been a need for large scale Enterprise voice systems. I am one of those people, and I am sure I am not alone, I will always want a physical phone. I am also one of these engineers who will always recommned a system that is directly under your own site's controll. Clouds are great, but they have their place. I don't think telecom will ever be a total cloud based solution. *From:* Bill Lake whl...@gmail.com *To:* Drake J jdrake...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Thursday, August 29, 2013 8:12 AM *Subject:* Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore? As a former big Telco employee, they want three things: Stability Scalability Profitability At this time these applications are not there. On Thu, Aug 29, 2013 at 6:45 AM, Bill Lake whl...@gmail.com wrote: As a former big Telco employee, they want three things: Stability Scalability On Thu, Aug 29, 2013 at 6:30 AM, Drake J jdrake...@gmail.com wrote: hi Laksh, Thanks for your inputs here.This was a good discussion. It is always good for us to all know about things that happen outside . Talking about Telco OTTs we can already see few of the Telcos have come out with Webrtc solutions for enterprise and service providers . Check this video out too depicting their solution... http://www.youtube.com/watch?v=Nz-BQZMp3sk Most of these applications written on software are supposed to open source and left for the users to customize . No real networking staff expertise required just download the SDK/API and customize and no more complex network topologies in future. Also no licensing fee too . Hence a real killer of techology in the future most likely we will see a wide spread of this starting 2014 if all predictions are to be believed. Hope someone from any of the TELCOs on this alias can add a few comments as well. Thanks once again for your inputs everyone. On Wed, Aug 28, 2013 at 11:05 PM, Lakshmish NS lakshmish...@gmail.comwrote: Hi Drake, I totally understand your concern, I'd be worried too. Having said that, we should always update ourselves with the latest technology. However, in future I believe Asterisk might be able to give tough run to Cisco UC. Not sure though, I hear stories that it is unstable and featureless compared to CUCM. I hope if someone aware of Asterisk would help us out here. Regard, Laksh On Wed, Aug 28, 2013 at 9:56 PM, Drake J jdrake...@gmail.com wrote: Hi Guys, Thanks for your responses I see u guys have empathized on call routing and and UC hardware for backend deployments. However Telco OTTs are coming up with directly provide these services over the cloud . Here is a disruptive analysis : http://www.slideshare.net/deanb/disruptive-analysis-web-rtc-overview-april-2013 Anyways, this might be not be so serious afterall . Just thought of brainstorming . Thanks guys for your responses again. On Tue, Aug 27, 2013 at 6:20 PM, Lakshmish NS lakshmish...@gmail.comwrote: Didn't have time to go through the video, I believe WebRTC is nothing but a Protocol, similar to SIP, H.323. Moreover, this protocol would only appeal to the Web audience, just like Skype, or Google talk. You still need to use UC hardware and their design for enterprise deployments. I mean we don't use Google talk and Skype in companies right? SIP is open source, but still Cisco uses it. As FAQ's suggest WebRTC is an open framework for the web that enables Real Time Communications in the browser. If only UC was that easy that could be implemented through browser, we didn't have to work this hard for CCIE numbers. You might want to go through this... http://www.webrtc.org/faq You've clearly misinterpreted WebRTC here.. On Tue, Aug 27, 2013 at 5:17 PM, Drake J jdrake...@gmail.com wrote: hi All, Had a troubling question hence thought of putting it out .Looking at the UC and networking trends worldwide it appears that the future of UC and collaboration is web based. Webrtc is the protocol
Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?
Hi Drake, WebRTC is a media stack that enables voice and video in the web browser with HTML5 and Javascript but AFAIK it is not used for dial plan resolution, supplementary services, path selection, etc. This is something that CUCM and it's like are providing as a call agent. WebRTC as the media stack when coupled with something like SIPML5 really turns your browser into a SIP endpoint and there are some really cool projects going on with integration to Askterisk and Freeswitch, but the server side intelligence is still needed. I don't think the move to browser-based endpoints threatens our discipline at all, it does provide a new set of skills to learn that compliment the skills you've already acquired. I think the UC engineer of the future will need to have some programming chops to compete, but IMO there will still be a line between the front end developers and the back end engineers that maintain the system. Just my 2 cents! Marty On Tue, Aug 27, 2013 at 7:47 AM, Drake J jdrake...@gmail.com wrote: hi All, Had a troubling question hence thought of putting it out .Looking at the UC and networking trends worldwide it appears that the future of UC and collaboration is web based. Webrtc is the protocol that the world will use and individuals and organizations just need to code their requirement based on the WEBRTC. Here is the presentation that Google recently made http://www.youtube.com/watch?v=E8C8ouiXHHk Clearly many of the UC vendors are already losing out and will be losing out in year 2014. Most of the customers are already looking at reducing the cost involved in maintaining costly UC vendor networks and their networking staff . Therefore that brings me to my question is the CCIE voice worth anymore? -Drake ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] AAR Not Attempting to ReRoute
Hi Alex, Have you enabled AAR in CCM service parameters? Not Enough Bandwidth indicates that it's not enabled. You should receive something like 'Network Congestion - Rerouting' when AAR is invoked. Marty On Thu, Aug 22, 2013 at 5:00 PM, Alex Pishko alexpis...@gmail.com wrote: All, Hopefully someone can give me an idea on this one. Working on my own equipment but following the labs and am running into an issue with AAR. I'm sending a call from HQ to BR 1 using 4 digit dial 5002 --- 1002. When I attempt to make the call I receive the message Not enough bandwidth, however I never see AAR actually get invoked. In my setup as a quick way to simulate congestion I set the bandwidth to 23 Kbps between Hub non (hq) and BR1. When I place the call from HQ as I said I get the banner of not enough bandwidth but I never see the call actually hit the HQ gateway. I've run debug voip dialpeer, q931 as well voice ccapi inout and neither shows any sort of traffic hitting the gateway. In testing I can successfully dial into the HQ GW, I can dial emergency services from the HQ phone, just doesn't seem like it's ever invoking AAR. I also checked that the external number mask is correctly defined on the 1002 extension. AAR CSS is assigned to the HQ phone, AAR group is assigned to the HQ line. AAR group is prefixing 91 and there is a RP assigned to a partition that falls within the AAR CSS that is for 91617XXX that has a RL pointed to the HQ GW. I did see earlier in the lab that they recommend using 7962 phones, however I don't have any avaialble at the moment, so just wanted to make sure that this might not be it. Any help would be much appreciated. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Guide Me
https://supportforums.cisco.com On Fri, Aug 9, 2013 at 8:31 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi All, I have one branch site At UK and on HQ site at Mumbai. when i call from India to UK , two way audio is perfect. but whe the call comes from UK to India, Audio is intermittent,Uk user can not hear but india user is hearing. There is One firewall at UK and one firewall at India through IPSEC. Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voicemail question
Hello, Use the voice mail box mask under the VM pilot instead of the calling party mask on the hunt pilot. Marty On Wed, Aug 7, 2013 at 5:20 AM, Olusegun Oguntuga segunogunt...@gmail.comwrote: Hi Experts, Please could anyone advise where is best to chop ANI to 4 digits when a site is in SRST and pressing the message button to listen to their voicemail. And they must hear 'enter your pin'. There is a requirement not to use alternate extensions in unity connection. There is another requirement further down to play sender's ANI before each message is played when a subscriber attempts to listen to their messages in their voice mailbox. What I have done so far: I have masked calling party using at the calling party mask of the hunt pilot, so that the subscriber hears 'enter your pin' when they dial into unity connection. But that will break a previous requirement to play sender's ANI before each message is played. Any Ideas plesse. Regards, Olu. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] clock on HQ and SB Phone
Well as long as the correct time zone is set under the DTG and the phones are registered correctly I'd check DB replication? What if you register the phones to the PUB? On Fri, Jul 12, 2013 at 9:59 PM, Karen Johnson karen.johnson...@yahoo.cawrote: Yes date group and dp is set. But hq and sb show same time, which Hq is pst and sb is cst -- * From: * Martin Sloan martinsloa...@gmail.com; * To: * Karen Johnson karen.johnson...@yahoo.ca; * Cc: * ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; * Subject: * Re: [OSL | CCIE_Voice] clock on HQ and SB Phone * Sent: * Fri, Jul 12, 2013 11:34:51 PM Okay. Do you have the right timezone set under the date/time group and is that DTG applied to the DP? AFAIK, the phones don't get the time from the router, they get it from the CUCM date/time group applied to the DP. On Fri, Jul 12, 2013 at 3:48 PM, Karen Johnson karen.johnson...@yahoo.cawrote: it is SCCP phones *From:* Martin Sloan martinsloa...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Friday, July 12, 2013 6:49:54 AM *Subject:* Re: [OSL | CCIE_Voice] clock on HQ and SB Phone Just a guess but if they are SIP phones and you don't have an NTP reference set for the DP then they'll use the time stamp on the 200 OK from the CUCM when they register. On Thu, Jul 11, 2013 at 10:51 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: folks, my HQ and SB phone keep showing same clock. HQ:PST SB:CST both assign to DP and in routers, i can see HQ as PST time and SB as CST from show clock. what i missed here? K ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] single button login
For your first issue, I think that error occurs if the end user doesn't have the IPCC Extension set. On Fri, Jul 12, 2013 at 10:18 AM, Amit Sharma aryan231...@gmail.com wrote: guys, i work on single button login.,.but showing id or password wrong.. i re check and try again same issue? can someone tell me what is issue? for uccx script pint, i have done script point..but when call from pstn not working..getting busy tone... when callfrom internal phones of any site working.. what could be issue in this point? -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] file tail activelog
use file list activelog cm/* to get a list of subdirectories. No slash before the cm and if you want the ccm traces just remember file tail activelog cm/trace/ccm/sdi recent or file tail activelog cm/trace/ccm/sdl recent On Fri, Jul 12, 2013 at 5:20 PM, Edgar Feliz ejzi...@gmail.com wrote: put a / and hit enter On Fri, Jul 12, 2013 at 4:33 PM, Karen Johnson karen.johnson...@yahoo.cawrote: when we do : file tail activelog /cm - what command to do HELP, if we forgot what is the next directory ? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] css and partitions required
Singh, It seems like you take a different approach than I do, but in case it helps here's what I currently use. For what it's worth, I have not yet taken the lab so this is all based on my studies and does get appended/modified as I discover things that work better for me: For each lab I create a base config with these PT's: INT SA SB SC SHARED I make a note of whether or not globalization, AAR or CFUR is required during my initial read through and if so I also include SA-ANI SA-DNIS SB-ANI SB-DNIS SC-ANI SB-DNIS AAR CFUR For the CSS: INT -INT SA -INT -SA -SHARED SB -INT -SB -SHARED SC -INT -SC If globablization/AAR/CFUR is required: SA-ANI -SA-ANI SB-ANI -SB-ANI SC-ANI -SC-ANI SA-DNIS -SA-DNIS SB-DNIS -SB-DNIS SC-DNIS -SC-DNIS AAR -AAR CFUR -CFUR I then use the setup below and modify as necessary as I read more of the lab details. I find that these configs work for 95% of the time: - I assign all DN's to the INT PT - I give all GWs, VM ports, CTI ports etc the CSS INT - GW configs get the appropriate Called party xformation DNIS CSS when I create them. - I assign the primary phone CSS, calling party xform ANI CSS, AAR group and AAR CSS's to the respective phones when I create them. As well as adding the AAR group to the line with the AAR number. - Shared PSTN numbers like 911 go into the SHARED PT for SA SB - Site specific patterns for teho or route redundancy go into their respective site PT. - I keep AAR and CFUR completely separate just for the sake of keeping things less complicated (in my head, at least). There might be a more concise way of assigning patterns and PT's when it comes to this but from what I understand there are no points given for conciseness, just that it works so for me this is a better approach. - I avoid using the none PT as well as hub_none location to keep things in order in my head. I like knowing exactly where things are and what they're doing. It might cost a couple extra keystrokes but it could save 20 minutes of troubleshooting down the line. HTH Marty On Tue, Jul 9, 2013 at 11:55 AM, singh singh8...@in.com wrote: Any update on this Guys? Please help!! -- Original message -- From:singh singh8...@in.com Date: 6 Jul 13 22:22:39 Subject: Re: css and partitions required To: ccie_voice-requ...@onlinestudylist.com;ccie_voice@onlinestudylist.com modified... -- Original message -- From:singh singh8...@in.com Date: 6 Jul 13 22:20:56 Subject: css and partitions required To: ccie_voice-requ...@onlinestudylist.com;ccie_voice@onlinestudylist.com hello all, I am wondering what would be appropriate number of partitions and CSS to create ... 1) I normally create partitions and css in the following manner ( these are for 3 sites HQ, site B , site c)... CSS-HQ-all -- Having access to partitions - pt-HQ-all, pt-HQ-siteB-all , pt-meetme, pt-aar, pt-plus CSS-siteB-al l-- Having access to partitions - pt-siteB-all, pt-HQ-siteB-all , pt-meetme, pt-aar, pt-plus CSS-sitec-all-- Having access to partitions - pt-siteC-all, pt-HQ-siteB-siteC-all Is the above correct? 2) To my phones HQ, siteB , site c - I do not assign any internal partitions - it is all in none. Is this correct? 3) I assign the CSS stated above to the appropriate - phones , gateways, cti ports , CTI route points and VM pilot . The partitions I assign only to route patterns , translation rules , meetme , Remote Destination Profile . Is this correct? Please correct me if I am wrong. -singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Five-Lab, Self-Study Challenge -- Lab #2 Gatekeeper
It looks like your HQ RAS IP isn't the same IP which the remote zone is pointing to. Could that be the issue? A-RTR-HQ-CCIE-V-IP-Expert-Lab-1#sh run | begin gatekeeper gatekeeper zone local GK ipexpert.com *10.10.100.1* zone local ViaGK ipexpert.com zone remote backbone ipexpert.com 10.10.100.2 1719 outvia ViaGK zone prefix backbone 011* no shutdown Pod1-TS-FRS-VPN-NAT-PSTN-CCIE-V-Lab-2#sh run | begin gatekeeper gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com *10.10.110.1* 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown On Wed, Jun 26, 2013 at 8:18 AM, Todd Carswell tcar0...@gmail.com wrote: No, I tried that last night. I changed remote zone on backbone to ViaGK and got same results. I put it back to the default for continued t-shooting. --Todd On Jun 26, 2013, at 6:45 AM, Bill whl...@gmail.com wrote: A-RTR-HQ-CCIE-V-IP-Expert-Lab-1#sh run | begin gatekeeper gatekeeper zone local GK ipexpert.com 10.10.100.1 zone local ViaGK ipexpert.com zone remote backbone ipexpert.com 10.10.100.2 1719 outvia ViaGK zone prefix backbone 011* no shutdown Pod1-TS-FRS-VPN-NAT-PSTN-CCIE-V-Lab-2#sh run | begin gatekeeper gatekeeper zone local backbone ipexpert.com 10.10.100.2 zone remote US ipexpert.com 10.10.110.1 1719 zone prefix backbone 44* gw-type-prefix 1#* default-technology no shutdown Not sure these match, try fixing and see if that helps Sent from my iPad On Jun 25, 2013, at 9:40 PM, Todd Carswell tcar0...@gmail.com wrote: A-RTR-HQ-CCIE-V-IP-Expert-Lab-1#sh run | begin gatekeeper gatekeeper zone local GK ipexpert.com 10.10.100.1 zone local ViaGK ipexpert.com zone remote backbone ipexpert.com 10.10.100.2 1719 outvia ViaGK zone prefix backbone 011* no shutdown ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BAT and cucm
This might sound silly but are you logged into the SUB? BAT is only available from the PUB. On Thu, Jun 20, 2013 at 6:25 AM, Drake J jdrake...@gmail.com wrote: hi Guys, Does callmanger 7.0.1.11000-2 support BAT? I checked the CM admin page and the BULK ADMINISTRATION tab is not available on the CM admin page -Drake ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM
Hi Karen, I just spent a couple hours banging my head against the desk because I ran into this same issue! It certainly wrecked my timing on the practice lab but I learned something about the symptom and maybe some troubleshooting tips, so I wanted to share. My 2 BR1 phones were like yours, they were showing as registered in CUCM but they had no DN on the phone. I tried just about everything to get them back to normal and I won't go into details but I was checking DHCP, trunk ports, vlans, yada yada and nothing worked so I started capturing packets. I started at the phone and could see that it was sending TFTP read requests to the CUCM but it wasn't getting anything back. I moved a step closer each way and used 'monitor capture buffers' on the ios to create .pcap files to view in Wireshark. At the BR1 WAN interface and HQ WAN interface I was seeing the same one way requests until I got to the HQ RTR/SW trunk port and then there was nothing! It told me that the TFTP requests weren't making it to the CUCM. It also gave me the idea to try a TFTP from the HQ router, requesting the phone config file for a BR1 phone and it worked. I moved to the BR1 router and tried the same thing and it failed. Then I checked out my WAN configs and sure enough, I had botched up the QoS settings. Once I adjusted that, everything worked fine. I know in your case you said there was no WAN QoS so the fix might be different but I thought I'd share the troubleshooting technique of just attempting the TFTP from the IOS to see if it's even connecting up. Had I done that from the beginning it would have saved me a ton of time. The other interesting thing about this is that the phones were registered during this investigation. It makes sense now but I didn't get it at first and I think it's a good clue to remember. I'm assuming your phones were SCCP phones, and since they don't need a DN to register they will send a register message to the CUCM using the last config file it downloaded. So in my case, the CUCM IP's were exactly the same from the last lab and when the TFTP of the new config file failed, it gave a last ditch effort and sent a SCCP register message to the CUCM from it's old file. I could see the message and the response in the Wireshark traces. The difference between the SCCP register and the TFTP read is a matter of TCP/UDP so I guess the reliable transport was able to get the messages delivered. Not sure if this is able to help you in your practice now but I know the next time I see this issue I'll have it narrowed down pretty quick! HTH Marty On Sat, Jun 15, 2013 at 9:18 PM, Karen Johnson karen.johnson...@yahoo.cawrote: I checked it did not -- * From: * Martin Sloan martinsloa...@gmail.com; * To: * Randall Crumm rrcr...@yahoo.com; * Cc: * Bill Lake whl...@gmail.com; Karen Johnson karen.johnson...@yahoo.ca; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; * Subject: * Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM * Sent: * Fri, Jun 14, 2013 12:22:39 AM To add to the other great ideas provided, I'd say if the phone is getting an IP via DHCP try to visit the web interface for the phone and see if it knows about it's DN. It will be on the first page under 'Phone DN'. On Thu, Jun 13, 2013 at 2:06 PM, Randall Crumm rrcr...@yahoo.com wrote: I dont think she tried to register on CUCM Have a great day! Thanks, Randall -- *From:* Bill Lake whl...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* Randall Crumm rrcr...@yahoo.com; Pavan K pav.c...@gmail.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Thursday, June 13, 2013 10:46 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM Not sure it is firmware issue if you can register via UCM dhcp but not IOS dhcp On Thu, Jun 13, 2013 at 12:33 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: database is good. i think as per Micahel mention , firmware issue, i will test and confirm -- *From:* Randall Crumm rrcr...@yahoo.com *To:* Karen Johnson karen.johnson...@yahoo.ca; Pavan K pav.c...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Bill Lake whl...@gmail.com *Sent:* Thursday, June 13, 2013 10:51:52 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM is your database replication status good? Change the MAC address of one of your HQ phones in CUCM. Then put that phone in SC and see if you can register it? Did you delete any unassiged DN's Have a great day! Thanks, Randall -- *From:* Karen Johnson karen.johnson...@yahoo.ca *To:* Pavan K pav.c...@gmail.com; rrcr...@yahoo.com rrcr...@yahoo.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Bill Lake whl...@gmail.com *Sent:* Thursday, June
Re: [OSL | CCIE_Voice] BACD Timer
Ok, I'll try to explain in my own words as I understand it. When a call is in queue the service will attempt a transfer to the hunt group after 'param call-retry-timer' seconds. If the call is not picked up by the hunt group, it goes back into queue. The hunt group will continue to be tried (after 'param call-retry-timer' seconds) *until **the* *param **max-time-call-retry* *expires*. It then sends the call to the destination defined in 'param voice-mail' HTH Marty On Tue, Jun 18, 2013 at 2:24 AM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: Thanks Martin, i went through that doc but still its not clear to me the purpose of using it and how does it effect my B-ACD script On Tue, Jun 18, 2013 at 2:13 AM, Martin Sloan martinsloa...@gmail.comwrote: I got the info below from this guide - http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 It has good examples you can copy/past/edit. I believe 'Call-Queue and AA Tcl Scripts in Flash Memory: Example' is the best one to use. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Step 29 *param* *max-time-call-retry* *seconds * Example: Router(config-app-param)# param max-time-call-retry 700 (Optional) Sets the maximum amount of time for the call-retry timer. This is the maximum period of time for which a call can stay in a call queue and retry to connect with a hunt group before the call is sent to an alternate destination number. •*seconds*—Maximum period of time, in seconds. The range is from 30 to 3600. The default is 600. On Mon, Jun 17, 2013 at 5:11 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: hi experts, I am trying to understand timer for *param* *max-time-call-retry can anyone share there knowledge about how does it effect the bacd script and the purpose of this field* * * *Thnks* * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cti route point isnot registering
Amit, Not all route points register. What is it for? Marty On Mon, Jun 17, 2013 at 11:17 AM, Amit Sharma aryan231...@gmail.com wrote: dear guys i create the cti route point,..but not registering... how can register it? -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] BACD Timer
I got the info below from this guide - http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 It has good examples you can copy/past/edit. I believe 'Call-Queue and AA Tcl Scripts in Flash Memory: Example' is the best one to use. http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/bacd/configuration/guide/40bacd.html#wp1026305 Step 29 *param* *max-time-call-retry* *seconds * Example: Router(config-app-param)# param max-time-call-retry 700 (Optional) Sets the maximum amount of time for the call-retry timer. This is the maximum period of time for which a call can stay in a call queue and retry to connect with a hunt group before the call is sent to an alternate destination number. •*seconds*—Maximum period of time, in seconds. The range is from 30 to 3600. The default is 600. On Mon, Jun 17, 2013 at 5:11 PM, CISCO CCIE VOICE ccievoic...@gmail.comwrote: hi experts, I am trying to understand timer for *param* *max-time-call-retry can anyone share there knowledge about how does it effect the bacd script and the purpose of this field* * * *Thnks* * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Dialing *Extension to reach the person voicemail is not working on v9.1
Hi Hesham, Do you have the VMbox mask set on the VM profile to strip the * ? ie to send 1130 intead of *1130. Marty On Sun, Jun 16, 2013 at 11:28 AM, Hesham Abdelkereem heshamcentr...@gmail.com wrote: Dear Experts, I'd like to configure on CUCM when I dial *Extension then I can reach the voicemail of the person directly and says Sorry Extension 1130 is not available please record ur message In V7 I just make a CTI Route Point with extension * on the internal partition then forward all to voicemail then its absoultely working. I have CUCM and Unity connection v9.1 and when I did that it just telling enter your pin followed by pound like I am calling the normal voicemail pilot number I tried to tweak the forwarding routing rule and direct routing rule but no chance unfortunately. Any Ideas Thank you very much in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUCM GK Port Number
Hey Josh, No worries, it's great to get feedback on this. I agree with you as well. It has to be either a mis-wording of the task requirement or an error in presenting the gatekeeper endpoint data. I also think the IPExpert material has been stellar, which made me really cautious to call it an error in the material without getting some expert feedback. I didn't think there was a way to hard code the ports outside of the service parameters but I wanted to check first. Marty On Fri, Jun 14, 2013 at 6:22 AM, Josh Petro josh.pe...@gmail.com wrote: Marty Sorry to beat a dead horse but I actually think its easier that that. The gatekeeper signaling port from the CUCM should be in that ephemeral range on the left and right side of the show command by default (as shown). Really, I think all the question is asking is to configure the gatekeeper such that the CUCMs both show with the listed name as well as the other gateway. You are correct in that the question is listed incorrectly. The ephemeral ports should be x or something. IPExpert is actually pretty good with the proof reading of their questions compared to others I've seen. It is aggravating when you have to take the time to figure out an incorrect question though! :) Josh On Jun 13, 2013 11:56 PM, Martin Sloan martinsloa...@gmail.com wrote: Thanks, Josh. I was getting caught up because the signaling ports in the output provided in the task were in the ephemeral range but the task said that port matching was a requirement. Based on the feedback here, I think the screen shot in the task was not right and was supposed to show the ports as 1720. If that was the case, I'd update the service parameter for the gk trunk, but It seems I can't force a match on the ports that were provided in the task. Thanks all for the help. Marty On Thursday, June 13, 2013, Josh Petro wrote: Hi Marty Bill and Justin are correct in that you can change the port number in the CUCM and the Gateway to reflect the signaling port (first port) The RAS port (second port listed on the line) will be in the ephemeral range. The ephemeral port range can't be changed on the CUCM as far as I know. I also remember Vic talking about that in a VOD in volume 1 (its in the 4.6 gatekeeper section). Josh On Jun 13, 2013 7:25 PM, Bill Lake whl...@gmail.com wrote: Marty, You got to the first column and select System parameters then scroll Dow the the H323 section and there will be your 1720 and there you will put the name of the gk trunk setup in your trunk config Reset the trunk and it should start using 1720 Sent from my iPhone On Jun 13, 2013, at 4:11 PM, Martin Sloan martinsloa...@gmail.com wrote: Justin, Thanks for the assist! I'm still lost on the requirement for this task b/c based on the output supplied, the CUCM's register with GK using ports 40446 and 35246 (gk-trunk_1 and gk-trunk_2). I'm not sure if: 1) I'm misunderstanding the question and/or output 2) The output in the task was supposed to show the CUCM's registered with port 1720 3) There is really a way to register the CUCM's with the ports they have in the output and I don't know how to do it. I've attached a SS of the section. Can you make any suggestions? I'm leaning heavily towards option 1 :-\ Thanks, Marty On Thu, Jun 13, 2013 at 4:04 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: With regards to your service parameter. Just make sure that the name of your trunk is listed in this service parameter. Then re-register your UCMs to the Gatekeeper. Couldn't also hurt to shut / no shut the gatekeeper. The service parameter is the key to what you are looking for here. Depending on what time frames you have entered for your registration timeouts you should see it repair after some time as long as you have the service parameter configured correctly. Thanks, Justin This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com Vol2_Lab6_Task4.2.png ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUCM GK Port Number
Thanks, Bill. My problem with this one is that the trunks aren't registered under port 1720. They're ports 40446 and 35246. I read 'Port number matching is required' to mean my trunks must match those in the output in the task. With those port numbers I don't believe it's possible. Do you agree it's probably an error in the output on this one? I think the trunks are supposed to be registered under port 1720, but I hate to assume. I know (okay, I've heard) the exam tasks leave some room for interpretation but I think on this one, there's only one way to interpret. Thanks, Marty On Thu, Jun 13, 2013 at 6:52 PM, Bill Lake whl...@gmail.com wrote: Marty, You got to the first column and select System parameters then scroll Dow the the H323 section and there will be your 1720 and there you will put the name of the gk trunk setup in your trunk config Reset the trunk and it should start using 1720 Sent from my iPhone On Jun 13, 2013, at 4:11 PM, Martin Sloan martinsloa...@gmail.com wrote: Justin, Thanks for the assist! I'm still lost on the requirement for this task b/c based on the output supplied, the CUCM's register with GK using ports 40446 and 35246 (gk-trunk_1 and gk-trunk_2). I'm not sure if: 1) I'm misunderstanding the question and/or output 2) The output in the task was supposed to show the CUCM's registered with port 1720 3) There is really a way to register the CUCM's with the ports they have in the output and I don't know how to do it. I've attached a SS of the section. Can you make any suggestions? I'm leaning heavily towards option 1 :-\ Thanks, Marty On Thu, Jun 13, 2013 at 4:04 PM, Justin McIntyre justin.mcint...@blackbox.com wrote: With regards to your service parameter. Just make sure that the name of your trunk is listed in this service parameter. Then re-register your UCMs to the Gatekeeper. Couldn't also hurt to shut / no shut the gatekeeper. The service parameter is the key to what you are looking for here. Depending on what time frames you have entered for your registration timeouts you should see it repair after some time as long as you have the service parameter configured correctly. Thanks, Justin This email and any files transmitted with it are confidential and are intended for the sole use of the individual to whom they are addressed. Black Box Corporation reserves the right to scan all e-mail traffic for restricted content and to monitor all e-mail in general. If you are not the intended recipient or you have received this email in error, any use, dissemination or forwarding of this email is strictly prohibited. If you have received this email in error, please notify the sender by replying to this email. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com Vol2_Lab6_Task4.2.png ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM
To add to the other great ideas provided, I'd say if the phone is getting an IP via DHCP try to visit the web interface for the phone and see if it knows about it's DN. It will be on the first page under 'Phone DN'. On Thu, Jun 13, 2013 at 2:06 PM, Randall Crumm rrcr...@yahoo.com wrote: I dont think she tried to register on CUCM Have a great day! Thanks, Randall -- *From:* Bill Lake whl...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* Randall Crumm rrcr...@yahoo.com; Pavan K pav.c...@gmail.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Thursday, June 13, 2013 10:46 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM Not sure it is firmware issue if you can register via UCM dhcp but not IOS dhcp On Thu, Jun 13, 2013 at 12:33 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: database is good. i think as per Micahel mention , firmware issue, i will test and confirm -- *From:* Randall Crumm rrcr...@yahoo.com *To:* Karen Johnson karen.johnson...@yahoo.ca; Pavan K pav.c...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Bill Lake whl...@gmail.com *Sent:* Thursday, June 13, 2013 10:51:52 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM is your database replication status good? Change the MAC address of one of your HQ phones in CUCM. Then put that phone in SC and see if you can register it? Did you delete any unassiged DN's Have a great day! Thanks, Randall -- *From:* Karen Johnson karen.johnson...@yahoo.ca *To:* Pavan K pav.c...@gmail.com; rrcr...@yahoo.com rrcr...@yahoo.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com; Bill Lake whl...@gmail.com *Sent:* Thursday, June 13, 2013 9:01 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM tks Pavan, look like not CTL issue, because when i switch this particular phone to UCM DHCP , it get DN but when i move back to SC router DHCP, i did not -- *From:* Pavan K pav.c...@gmail.com *To:* karen.johnson...@yahoo.ca; rrcr...@yahoo.com *Cc:* ccie_voice@onlinestudylist.com; Bill Lake whl...@gmail.com *Sent:* Wednesday, June 12, 2013 5:52:18 PM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM Karen, Check if your phone has an ITL/CTL installed. Erase them and see if the changes take affect Do you see the phone as registered in UCM ? On Jun 12, 2013 5:23 PM, Randall Crumm rrcr...@yahoo.com wrote: Did you try to auto register the phones? Have a great day! Thanks, Randall -- *From:* Karen Johnson karen.johnson...@yahoo.ca *To:* Bill Lake whl...@gmail.com *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Wednesday, June 12, 2013 1:18 PM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM hi Bill, yes i got it from SC router, and i tried to ping UCM as well from the voice int vlan of SC and it works for the ping. but still not get the DN k *From:* Bill Lake whl...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Wednesday, June 12, 2013 11:55:37 AM *Subject:* Re: [OSL | CCIE_Voice] Phone not pick up TFTP or config from UCM Where are you getting your dhcp at sc? If it is sc rtr then you could have it right and just no access to CUCM So try ping (CUCM ip) source vlan (voice #) and confirm the voice vlan has connectivity to CUCM Sent from my iPhone On Jun 12, 2013, at 11:29 AM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi all, Phones are getting the ip address and option 150 from SC router DHCP. However the phones do not show DN or extension. (only blue image background) When I checked the Network setting in phone (ip,subnet, TFTP all showing correctly) - What is the cause phone not able to pick up the DN config from UCM (in UCM show the phones registered with DN) ? - I have also restart the TFTP and UCM. - here is my diagram : phone --- switch--- SC router (DHCP) tks for help ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are
Re: [OSL | CCIE_Voice] CUCM GK Port Number
Thanks, Josh. I was getting caught up because the signaling ports in the
output provided in the task were in the ephemeral range but the task said
that port matching was a requirement. Based on the feedback here, I think
the screen shot in the task was not right and was supposed to show the
ports as 1720. If that was the case, I'd update the service parameter for
the gk trunk, but It seems I can't force a match on the ports that were
provided in the task.
Thanks all for the help.
Marty
On Thursday, June 13, 2013, Josh Petro wrote:
Hi Marty
Bill and Justin are correct in that you can change the port number in the
CUCM and the Gateway to reflect the signaling port (first port) The RAS
port (second port listed on the line) will be in the ephemeral range. The
ephemeral port range can't be changed on the CUCM as far as I know. I also
remember Vic talking about that in a VOD in volume 1 (its in the
4.6 gatekeeper section).
Josh
On Jun 13, 2013 7:25 PM, Bill Lake whl...@gmail.com javascript:_e({},
'cvml', 'whl...@gmail.com'); wrote:
Marty,
You got to the first column and select System parameters then scroll Dow
the the H323 section and there will be your 1720 and there you will put the
name of the gk trunk setup in your trunk config
Reset the trunk and it should start using 1720
Sent from my iPhone
On Jun 13, 2013, at 4:11 PM, Martin Sloan
martinsloa...@gmail.comjavascript:_e({}, 'cvml',
'martinsloa...@gmail.com');
wrote:
Justin,
Thanks for the assist! I'm still lost on the requirement for this task
b/c based on the output supplied, the CUCM's register with GK using ports
40446 and 35246 (gk-trunk_1 and gk-trunk_2). I'm not sure if: 1) I'm
misunderstanding the question and/or output 2) The output in the task was
supposed to show the CUCM's registered with port 1720 3) There is really a
way to register the CUCM's with the ports they have in the output and I
don't know how to do it.
I've attached a SS of the section. Can you make any suggestions? I'm
leaning heavily towards option 1 :-\
Thanks,
Marty
On Thu, Jun 13, 2013 at 4:04 PM, Justin McIntyre
justin.mcint...@blackbox.com javascript:_e({}, 'cvml',
'justin.mcint...@blackbox.com'); wrote:
With regards to your service parameter. Just make sure that the name of
your trunk is listed in this service parameter. Then re-register your UCMs
to the Gatekeeper. Couldn't also hurt to shut / no shut the gatekeeper.
The service parameter is the key to what you are looking for here.
Depending on what time frames you have entered for your registration
timeouts you should see it repair after some time as long as you have the
service parameter configured correctly.
Thanks,
Justin
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restricted content and to monitor all e-mail in general. If you are not the
intended recipient or you have received this email in error, any use,
dissemination or forwarding of this email is strictly prohibited. If you
have received this email in error, please notify the sender by replying to
this email.
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Vol2_Lab6_Task4.2.png
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___
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Are you a CCNP or CCIE and looking for a job? Check out
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[OSL | CCIE_Voice] CUCM GK Port Number
I'm working on task 4.2 of Vol 2 Lab 6 and there's some output from a 'show gatekeeper endpoint' and 'show gatekeeper gw-type-prefix' that we're required to bring up in our configs. As far as gateway registration, tech prefixes, etc my config is looking good but at the very end of the task description it says: 'Port number matching is required for the purpose of this task' The port numbers for the CUCM gk-trunk_1 and gk-trunk_2 trunks are 40446 and 35246, respectively. The solutions guide does not touch on the port numbers in the explanation (that I was able to find). Am I misreading the requirements or is there a way to configure these port numbers? I know of the service parameter to set the GK trunk that will use 1720 but I'm lost on this one. Your help is appreciated. Marty ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] RSVP Problem
Thanks, Vasanth. I appreciate your input! On Tue, Jun 11, 2013 at 2:01 AM, Vasanth vasant...@gmail.com wrote: Yes. In that case you can go with the approach you have taken. I had tough time when doing it in a different lab scenario and then finally removing the QoS policy fixed my problem and then did reverse engineering of the QoS policy(with help of CCM Traces) to find out that fragmentation is causing the problem. If the ask is to configure LFI and RSVP on the same link then you might be using a different version of IOS/CCM. As WAN QoS question I would alway start with auto qos and work through the class-map,map-class,policy-map to meet the requirement. Cheers, Vasanth Regards, Vasanth On Tue, Jun 11, 2013 at 2:40 AM, Martin Sloan martinsloa...@gmail.comwrote: Hi Vasanth, Thanks for the reply. I was able to get this working by removing/re-adding the RSVP MTP association to the MRG in CUCM. Calls are working fine with RSVP now. About the fragmentation, it's required as part of the next task for WAN QoS with LFI between HQ-BR1 so I don't think I can avoid that part. Do you agree? I posted a similar question to the group in regard to setting up the LFI for these tasks. I've been using auto qos because it creates all of the class-maps and calculates the fragment size for me, so there's no digging for the information within Cisco docs. If you're interested, check out my email from 6/1 titled 'Advice or opinions on Vol 2 Lab 4 Task 5.1' and let me know what you think. Thanks, Marty On Mon, Jun 10, 2013 at 3:46 PM, Vasanth vasant...@gmail.com wrote: On Mon, Jun 10, 2013 at 9:30 PM, ccie_voice-requ...@onlinestudylist.com wrote: command under the dspfarm profile as well, it didn't copy over in my email: Hi Martin, You have auto qos enabled for 384 bandwidth frame-relay link. This would bring in frame-relay fragmentation of packets. auto qos would configure frame-relay fragment size of 480 bytes. This causes the ccm not able to parse the rsvp response from the Router (MTP) to call manager. If you can remove the frame-relay fragment command and check RSVP it should work. Regards, Vasanth ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] switch- spanning tree portfast
Singh, It does add the 'spanning tree portfast' command on the HQ switch, but not on the switch modules at the branch routers. You have to add that command manually for the switch modules. Marty On Tue, Jun 11, 2013 at 9:33 AM, singh singh8...@in.com wrote: hi Guys, Is the command spanning tree portfast required under the switch ports on a switch connected to the phone? Doesn't the switchport mode voice vlan take care of this ? Regards, singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] switch- spanning tree portfast
Hi Tony, I would like to know the 'correct' way too (at least in the context of the lab grading) because I want to make sure I get the points on this! When technology evolves, you can usually find conflicting information as noticed here: http://www.cisco.com/en/US/products/hw/modules/ps2797/products_configuration_example09186a00808066b8.shtml#step4 This guide recommends making the ports access ports, which is what I've been practicing. The link you gave says the 802.1Q tagging is vital for 'Cisco AVVID networks', so I'm not sure this is the current best practice but I guess that doesn't say much about getting the points in the lab. Logically for me it makes more sense to use the more current access-port style config, which creates a little conundrum for me if you guys are saying the trunk-style is what they're looking for in the lab. You probably caught the very nice explanation provided by Michael Sears as well, who supports your thoughts on it. Am I off base that the access-style config would be considered the current proper configuration? Thanks, Marty On Tue, Jun 11, 2013 at 1:26 PM, Tony Zunt tony.z...@gmail.com wrote: Marty is correct and we know that config works just fine most of the time. I am curious whether or not CISCO may be looking for something a little different for this element? A wise sage on this forum (certainly not me) pointed out earlier that CISCO recommends use of the 'switchport mode trunk' configuration when supporting ip phones on ESW (which is what we do on site B, site C usually): http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1 That's the method I practice, however in this case it is no longer possible to apply the required ' 'spanning-tree portfast' command on the interface in trunk mode. Therefore the wise engineer also suggested using 'no spanning-tree vlan my voice vlan#' command in global configuration mode in order to allow the phones to boot rapidly. I thought that was brilliant and find it works well too on my lab, but I always wonder what the correct way is. Thanks On Tue, Jun 11, 2013 at 11:29 AM, Martin Sloan martinsloa...@gmail.comwrote: Singh, It does add the 'spanning tree portfast' command on the HQ switch, but not on the switch modules at the branch routers. You have to add that command manually for the switch modules. Marty On Tue, Jun 11, 2013 at 9:33 AM, singh singh8...@in.com wrote: hi Guys, Is the command spanning tree portfast required under the switch ports on a switch connected to the phone? Doesn't the switchport mode voice vlan take care of this ? Regards, singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] switch- spanning tree portfast
Thanks Bill, et al. It looks like trunk ports are the way to go on lab day. On Tue, Jun 11, 2013 at 4:09 PM, Tony Zunt tony.z...@gmail.com wrote: I think Mr. Sears provided the best response we may see. Our emails passed in midair. If I'd seen his first, I wouldn't have sent mine. :^) Thanks On Tue, Jun 11, 2013 at 2:37 PM, Martin Sloan martinsloa...@gmail.comwrote: Hi Tony, I would like to know the 'correct' way too (at least in the context of the lab grading) because I want to make sure I get the points on this! When technology evolves, you can usually find conflicting information as noticed here: http://www.cisco.com/en/US/products/hw/modules/ps2797/products_configuration_example09186a00808066b8.shtml#step4 This guide recommends making the ports access ports, which is what I've been practicing. The link you gave says the 802.1Q tagging is vital for 'Cisco AVVID networks', so I'm not sure this is the current best practice but I guess that doesn't say much about getting the points in the lab. Logically for me it makes more sense to use the more current access-port style config, which creates a little conundrum for me if you guys are saying the trunk-style is what they're looking for in the lab. You probably caught the very nice explanation provided by Michael Sears as well, who supports your thoughts on it. Am I off base that the access-style config would be considered the current proper configuration? Thanks, Marty On Tue, Jun 11, 2013 at 1:26 PM, Tony Zunt tony.z...@gmail.com wrote: Marty is correct and we know that config works just fine most of the time. I am curious whether or not CISCO may be looking for something a little different for this element? A wise sage on this forum (certainly not me) pointed out earlier that CISCO recommends use of the 'switchport mode trunk' configuration when supporting ip phones on ESW (which is what we do on site B, site C usually): http://www.cisco.com/en/US/docs/ios-xml/ios/lanswitch/configuration/12-4t/lsw-hwic-ethsw-ic.html#GUID-379450C0-5434-4AC3-9BED-396CB7D162C1 That's the method I practice, however in this case it is no longer possible to apply the required ' 'spanning-tree portfast' command on the interface in trunk mode. Therefore the wise engineer also suggested using 'no spanning-tree vlan my voice vlan#' command in global configuration mode in order to allow the phones to boot rapidly. I thought that was brilliant and find it works well too on my lab, but I always wonder what the correct way is. Thanks On Tue, Jun 11, 2013 at 11:29 AM, Martin Sloan martinsloa...@gmail.comwrote: Singh, It does add the 'spanning tree portfast' command on the HQ switch, but not on the switch modules at the branch routers. You have to add that command manually for the switch modules. Marty On Tue, Jun 11, 2013 at 9:33 AM, singh singh8...@in.com wrote: hi Guys, Is the command spanning tree portfast required under the switch ports on a switch connected to the phone? Doesn't the switchport mode voice vlan take care of this ? Regards, singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] RSVP Problem
I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would like some advice on what I might have missed. The requirement is to allow an equal number of calls (2) over redundant links from HQ-BR1 and 4 calls from HQ-BR2. Also, RSVP should use 'video desired' allowing calls to proceed as audio only when there is not enough bandwidth for audio and video. Just using HQ-BR1 as an example, so far I have configured: - bandwidth statements on Serial sub-interfaces: HQ: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.1 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 class AutoQoS-FR-Se0/1/0-201 auto qos voip ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.5 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 211 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 BR1: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.2 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.6 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 111 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 - Software MTP on the router: HQ: sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register HQ-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 rsvp maximum sessions software 8 associate application SCCP BR1: sccp local Loopback0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register BR1-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 maximum sessions software 4 associate application SCCP - Configured both MTP's in CUCM (both are registered) - Placed MTP's into their own MRG - Placed the RSVP MRG at the bottom of the MRGL for each site - Set Locations RSVP settings to mandatory (video desired) - Inter-region settings for HQ/BR1 is set to g729 - HQ Device Pool contains HQ Location and Region - BR1 Device Pool contains BR1 Location and Region - HQ Device Pool is assigned to the phone placing the call to BR1 - BR1 Device Pool is assigned to the phone I'm calling to from HQ With the above settings, I never see any RSVP messaging on the routers. I've done a debug sccp all and debug ip rsvp all and there is nothing sent in regard to RSVP CAC. I've shut/no shut sccp about 100 times, rebooted the routers and rebooted the CUCM servers but still nothing. I pulled some CUCM traces and I can see there is activity for RSVP but it's unclear to me what the issue is. Here are the last few lines, with an 'SsCause' code that I haven't been able to dig up the meaning on: 000618017| 2013/06/07 11:26:50.937| 002| SdlSig| SsUnregisterRelRejInterceptReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 3, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2 SsParty=45758182 handler=0 000618018| 2013/06/07 11:26:50.937| 002| SdlSig| SsUnregisterRelRejInterceptReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 2, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2 SsParty=45758182 handler=0 000618019| 2013/06/07 11:26:50.937| 002| SdlSig| SsClearCallReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 1, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=40 SsNode=2 SsParty=45758182 *SsCause=125 *clearCallRequestor=0 clearCallInstruction=1 FDataType=0opId=0invokeId=0resultExp=F Can someone take a look and let me know if there's a glaring issue with my configs? I've set this up numerous times in the other labs so I'm either blanking on the proper configs or missing a 'gotcha' somewhere. Any help is appreciated. Marty ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] RSVP Problem
Just re-read my configs, please note that BR1 *does* have the 'rsvp' command under the dspfarm profile as well, it didn't copy over in my email: dspfarm profile 1 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 associate application SCCP On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan martinsloa...@gmail.comwrote: I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would like some advice on what I might have missed. The requirement is to allow an equal number of calls (2) over redundant links from HQ-BR1 and 4 calls from HQ-BR2. Also, RSVP should use 'video desired' allowing calls to proceed as audio only when there is not enough bandwidth for audio and video. Just using HQ-BR1 as an example, so far I have configured: - bandwidth statements on Serial sub-interfaces: HQ: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.1 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 class AutoQoS-FR-Se0/1/0-201 auto qos voip ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.5 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 211 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 BR1: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.2 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.6 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 111 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 - Software MTP on the router: HQ: sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register HQ-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 rsvp maximum sessions software 8 associate application SCCP BR1: sccp local Loopback0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register BR1-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 maximum sessions software 4 associate application SCCP - Configured both MTP's in CUCM (both are registered) - Placed MTP's into their own MRG - Placed the RSVP MRG at the bottom of the MRGL for each site - Set Locations RSVP settings to mandatory (video desired) - Inter-region settings for HQ/BR1 is set to g729 - HQ Device Pool contains HQ Location and Region - BR1 Device Pool contains BR1 Location and Region - HQ Device Pool is assigned to the phone placing the call to BR1 - BR1 Device Pool is assigned to the phone I'm calling to from HQ With the above settings, I never see any RSVP messaging on the routers. I've done a debug sccp all and debug ip rsvp all and there is nothing sent in regard to RSVP CAC. I've shut/no shut sccp about 100 times, rebooted the routers and rebooted the CUCM servers but still nothing. I pulled some CUCM traces and I can see there is activity for RSVP but it's unclear to me what the issue is. Here are the last few lines, with an 'SsCause' code that I haven't been able to dig up the meaning on: 000618017| 2013/06/07 11:26:50.937| 002| SdlSig| SsUnregisterRelRejInterceptReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 3, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2 SsParty=45758182 handler=0 000618018| 2013/06/07 11:26:50.937| 002| SdlSig| SsUnregisterRelRejInterceptReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 2, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=0 SsNode=2 SsParty=45758182 handler=0 000618019| 2013/06/07 11:26:50.937| 002| SdlSig| SsClearCallReq| tcc_intercept | Cdcc(2,100,171,82) | Cc(2,100,172,1) | (2,100,37,1).114100-(SEP001E4A928455:10.10.200.72)| [R:NP - HP: 0, NP: 1, LP: 0, VLP: 0, LZP: 0 DBP: 0]SsType=33554437 SsKey=40 SsNode=2 SsParty=45758182 *SsCause=125 *clearCallRequestor=0 clearCallInstruction=1 FDataType=0opId=0invokeId=0resultExp=F Can someone take a look and let me know if there's a glaring issue with my configs? I've set this up numerous times in the other labs so I'm
Re: [OSL | CCIE_Voice] RSVP Problem
Suresh, Thank you for the help! I did have the 'ip rsvp bandwidth' setting under the physical interfaces as well. As a last-ditch effort I removed the HQ-RSVP MTP from the MRG, inserted a generic MTP and saved, then reverted that change and put the HQ-RSVP MTP back in and saved.voila! It now works. I don't know what that accomplished which a system reboot did not, but it's a lesson learned in troubleshooting. Before I hack through system traces, I should try some quicker fixes first. It would have been faster had I just ripped it all out and rebuilt it. It's hard to not get bogged down sometimes when you're in the weeds. Thanks again for the help. Marty On Mon, Jun 10, 2013 at 12:15 PM, Suresh Bhandari bring...@gmail.comwrote: What happens when you make a call from HQ to BR1 phones? If phones ring, check if show sccp connections has expected output - that is, there is a reservation of 40K bandwidth for each ringing phones. You have partial configuration included here. Can you make sure that you have configured physical interfaces, Serial0/1/0 (from your config), on both routers to have ip rsvp bandwidth? On Mon, Jun 10, 2013 at 8:24 PM, Martin Sloan martinsloa...@gmail.comwrote: Just re-read my configs, please note that BR1 *does* have the 'rsvp' command under the dspfarm profile as well, it didn't copy over in my email: dspfarm profile 1 mtp codec pass-through codec g729r8 rsvp maximum sessions software 4 associate application SCCP On Mon, Jun 10, 2013 at 10:28 AM, Martin Sloan martinsloa...@gmail.comwrote: I'm running into an issue getting RSVP to work on Vol 2 Lab 5 and would like some advice on what I might have missed. The requirement is to allow an equal number of calls (2) over redundant links from HQ-BR1 and 4 calls from HQ-BR2. Also, RSVP should use 'video desired' allowing calls to proceed as audio only when there is not enough bandwidth for audio and video. Just using HQ-BR1 as an example, so far I have configured: - bandwidth statements on Serial sub-interfaces: HQ: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.1 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 201 class AutoQoS-FR-Se0/1/0-201 auto qos voip ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.5 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 211 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 BR1: interface Serial0/1/0.1 point-to-point bandwidth 384 ip address 10.10.111.2 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 101 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 ! interface Serial0/1/0.2 point-to-point bandwidth 384 ip address 10.10.111.6 255.255.255.252 ip ospf mtu-ignore snmp trap link-status frame-relay interface-dlci 111 class AutoQoS-FR-Se0/1/0-201 ip rsvp bandwidth 64 - Software MTP on the router: HQ: sccp local Loopback0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register HQ-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 rsvp maximum sessions software 8 associate application SCCP BR1: sccp local Loopback0 sccp ccm 10.10.210.11 identifier 1 priority 1 version 7.0 sccp ccm 10.10.210.10 identifier 2 priority 2 version 7.0 sccp ! sccp ccm group 1 bind interface Loopback0 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register BR1-RSVP ! dspfarm profile 1 mtp codec pass-through codec g729r8 maximum sessions software 4 associate application SCCP - Configured both MTP's in CUCM (both are registered) - Placed MTP's into their own MRG - Placed the RSVP MRG at the bottom of the MRGL for each site - Set Locations RSVP settings to mandatory (video desired) - Inter-region settings for HQ/BR1 is set to g729 - HQ Device Pool contains HQ Location and Region - BR1 Device Pool contains BR1 Location and Region - HQ Device Pool is assigned to the phone placing the call to BR1 - BR1 Device Pool is assigned to the phone I'm calling to from HQ With the above settings, I never see any RSVP messaging on the routers. I've done a debug sccp all and debug ip rsvp all and there is nothing sent in regard to RSVP CAC. I've shut/no shut sccp about 100 times, rebooted the routers and rebooted the CUCM servers but still nothing. I pulled some CUCM traces and I can see there is activity for RSVP but it's unclear to me what the issue is. Here are the last few lines, with an 'SsCause' code that I haven't been able to dig up the meaning on: 000618017| 2013/06/07 11:26:50.937| 002| SdlSig| SsUnregisterRelRejInterceptReq
Re: [OSL | CCIE_Voice] RSVP Problem
Hi Vasanth, Thanks for the reply. I was able to get this working by removing/re-adding the RSVP MTP association to the MRG in CUCM. Calls are working fine with RSVP now. About the fragmentation, it's required as part of the next task for WAN QoS with LFI between HQ-BR1 so I don't think I can avoid that part. Do you agree? I posted a similar question to the group in regard to setting up the LFI for these tasks. I've been using auto qos because it creates all of the class-maps and calculates the fragment size for me, so there's no digging for the information within Cisco docs. If you're interested, check out my email from 6/1 titled 'Advice or opinions on Vol 2 Lab 4 Task 5.1' and let me know what you think. Thanks, Marty On Mon, Jun 10, 2013 at 3:46 PM, Vasanth vasant...@gmail.com wrote: On Mon, Jun 10, 2013 at 9:30 PM, ccie_voice-requ...@onlinestudylist.comwrote: command under the dspfarm profile as well, it didn't copy over in my email: Hi Martin, You have auto qos enabled for 384 bandwidth frame-relay link. This would bring in frame-relay fragmentation of packets. auto qos would configure frame-relay fragment size of 480 bytes. This causes the ccm not able to parse the rsvp response from the Router (MTP) to call manager. If you can remove the frame-relay fragment command and check RSVP it should work. Regards, Vasanth ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 trunk no PLUS in EPNM
Ugh, just saw all of the other correct answers above. I hate google mail but for some reason the OSL would never come through on my yahoo account. Sorry for beating the dead horse. On Fri, Jun 7, 2013 at 9:47 AM, Martin Sloan martinsloa...@gmail.comwrote: Hi Karen, I believe this is expected. The h.323 GW strips the + by default. As a workaround, create a calling party translation profile to put the plus back on and apply it to the outbound dial-peer, you should see the calling number with the + on it. Marty On Thu, Jun 6, 2013 at 5:06 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi, I set up H323 trunk from UCM to my PSTN router (as H323 GW) - EPNM : +15052022XXX - RP : check Use calling party Ext Mask But when i see the call in PSTN phone, it did not have +1505202xxx, but only 15052022XXX questions: - is this expected ? - and if we want +1505xx, what is the workaround ? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] H323 trunk no PLUS in EPNM
Hi Karen, I believe this is expected. The h.323 GW strips the + by default. As a workaround, create a calling party translation profile to put the plus back on and apply it to the outbound dial-peer, you should see the calling number with the + on it. Marty On Thu, Jun 6, 2013 at 5:06 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi, I set up H323 trunk from UCM to my PSTN router (as H323 GW) - EPNM : +15052022XXX - RP : check Use calling party Ext Mask But when i see the call in PSTN phone, it did not have +1505202xxx, but only 15052022XXX questions: - is this expected ? - and if we want +1505xx, what is the workaround ? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE AFTER SRST
Hi Ivan, Check your CUCM configs in the CUE web admin page. Go to 'Configure-Call Manager' and make sure you have all of the required information populated. The CUE setup wizard will let you walk through without specifying a JTAPI user/pass but if you look into the menu I mentioned, it's listed as required information. Verify your CUCM app user information there and reload the CUE module. I had the same problem, with no JTAPI info filled out. The RP's should then register. HTH Marty On Thu, Jun 6, 2013 at 6:49 PM, Ivan Darío Sánchez Calderón ids.calde...@gmail.com wrote: Hi everyone, When I have cue integrated with CUCM and I make some voicemail test when the gateway is in srst everything works fine, but when I come back from srst the CTI RP and the CTI Ports don't register. I restart the cue, the CTI service on CUCM but still shows unregistered, someone knows how to fix this issue.? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] IPMA Assistant Console QoS
Hello, I'm reading through the SRND on the IPMA Assistant console application and it says that it marks all traffic as BE and that an ACL should be setup along the path somewhere to remark that traffic (on port 2912) to DSCP 24. I haven't come across this requirement in the practice labs and I'm wondering, has anyone taken an approach to configuring this and do you have any recommendations? Just a simple ACL on the port which the PC is connected to, or don't even worry about it if it's not specifically asked for in the QoS or IPMA tasks? Thanks, Marty ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab strategy
Let me say, I've never sat the lab so I'm just commenting on my own study experience. - Points 3 4 are probably consuming too much time. Documenting IP's is good but I don't currently go through and check configs. I figure i'll get to it when it's time to configure that piece and if there are issues I'll do my troubleshooting then. - 1.5hrs for Br2 seems like a long time. I think you can get quicker with that part and do it in 45min or less, depending on the complexity. Just with those savings you're getting closer to your 6 hour goal. I use notepad for anything that can be repeated like dial-peers, translation rules/patterns, dspfarm, QoS, etc. I think its MUCH faster than manually typing the configs each time. I always configure BR1 first so most of the time I'm just tweaking some settings and pasting into BR2. Just my input based on my own studies. I'm sure there are veterans out there who have a better insight into saving time. I'm interested to hear their take as well. Marty On Wed, Jun 5, 2013 at 10:58 AM, singh singh8...@in.com wrote: hi Everyone, I need your inputs here. I have been trying to complete mock and practice labs in 8 hours . However their I am unable to finish or I finish with lab with a lot of mistakes with no time for testing. I also realize that I lose thing in the first half and speed up during the end . Generally I take... 1) 10 mins to test if all equipment is working fine ( 10 mins) 2) read the workbook questions for the next 15 mins 3) Make note of the ip addresses and router configuration per site in 25 mins 4) Make note of all cucm configuration , cups , unity connection , uccx and cue for another dial plan 20 mins 5) Now from point 5 - I start with lab configurations from Branch 2 ( site c - which is a mgcp gateway and srst setup) this generally takes me about 1 and a half hour to just complete all configuration ( 1 hour and 30 mins) 6) Then I move to Branch 1 ( site B - which is a H323 gateway with srst) this ge nerally takes about 45 - 50 mins 7) I then move to HQ ( R1 - which is MGCP gateway with srst ) this generally takes 20 mins . 8) Basic setup of DP , css, aar, NTP , service parameters and enterprise para , vlans , dhcp and ip phone registration takes 50 mins 9) CUC integ and other config including recording takes 20 mins 10 ) CUE integration and trafer setup takes 20 mins 11) UCCX integration , One button , script and recording takes 40 mins 12) CUPs integration and client setup takes 20 mins 13 )I then come back to callmanger and do the media resource setup , gateways added cucm , other configuration such as MVA , RSVP , + dial , adding trunks , unassigned dn setup to CUC - this takes 50 mins 14 ) Then I move to the Route pattern setup on callmanger this I do for 3 sites - HQ , Site B and Site C with or without redundancy on callmanger this takes 25 mins 15 ) Then do this such as RTMT log collection and indicating informa tion in seperate files , MGCP debugs this takes another 15 mins 16 ) Switch and WAN QOS this I plan to do only if time permits as this is a complex section Questions : === 1) I barely am able to finish things in time . I have heard from on this forum that there are candidates who finish it is 5 - 6 hours . Would anyone be able to share with me as to how they do this ? 2) Even if the above I complete exercises are complete there are sections where I miss out on configurations. How do I make sure all config for all sections is done correctly? 3) I really wish I am able to finish the lab in 6 hours so that I can test for another 2 hours . Could someone therefore check the above 16 points and let me know about the time I can reduce. 4) As you can see above the router configurations consume a bit more time making ( points 5 , 6 , 7 ) . I have tried with both using a notepad to type the configurations and then paste also with typing on cli but both these methods take around the same time. Please let me know what best method I can use for points 5 , 6 , 7. 5) Other suggestions are most welcome . Thanks guys in advance for all your help! Regards, Singh Get Yourself a cool, short *@in.com* Email ID now!http://www3.in.com/sso/commonregister.php?ref=INutm_source=inviteutm_medium=outgoing ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Thank You
Ben, I just pulled away from a timed practice lab to get the news of the CCIE Voice to CCIE Collaboration transition change about 2 hours late. This is awesome news!! I was a restaurant manager for many years and I know that people can be quick to complain but not so fast to commend, so I wanted to be sure and THANK YOU for changing the policy on the CCIE Voice/Collaboration certifications. Cisco has done the right thing and you have my sincerest gratitude for any part you played in this change. I don't have 'a number' yet, but with this news my enthusiasm to get the certification is restored and I know that you've made a lot folks happy with this change in policy. Thank you again, and hopefully I'll be taking a written exam to get that Collaboration IE instead of the lab. Marty ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] FTP from CUE failed
Filezilla server provides a console log that's typically verbose enough to get a point in the right direction. Do you see a connection attempt with some log messages? On Tue, Jun 4, 2013 at 12:42 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi all, When i try to install license from FTP to CUE, it failed and gave me error message : Download error, Can not download cue-vm-license_12mbx_7.0.1.pkg error code 0 error type 'couldn't connect to host - I have verified the FTP server (Filezilla server) is working using Filezilla client . - username and password is correct - File license is valid What possibility here? tks K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Home Router, switch and Phones for Sale -- I just passed!
Awesome. Congrats, Randall! On Tue, Jun 4, 2013 at 2:05 PM, Randall Crumm rrcr...@yahoo.com wrote: Hi, I have a router, POE switch and phones for sale. Everything is configured and works with Proctorlabs PODS VPN access. This is what I used to pass my lab. PM me if you are interested. Router 26xxXM Switch 35xx POE phones 7960 x 1 for PSTN 7961 x 2 7965 x 1 7070 x2 Have a great day! Thanks, Randall CCIE #39411 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] FTP from CUE failed
Sounds like it's a routing/acl issue then. Can you ping to/from the CUE/FTP server? Are there any devices/ACL's in between that would block this? If the CUE is a few hops away and you have devices that are closer to the FTP server you could start at the closest one and work your way out and telnet to port 21 until it breaks down. If the port is open, it will answer. If you're using IOS devices just make sure that 'transport output telnet' is set under you con/vty line, then 'telnet ip of ftp server 21' and see if it answers. My first step would be standard ping in both directions though. On Tue, Jun 4, 2013 at 6:16 PM, Karen Johnson karen.johnson...@yahoo.cawrote: yes i did not see when I do from CUE but when i do from Filezilla client, I see *From:* Martin Sloan martinsloa...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 4, 2013 12:01:01 PM *Subject:* Re: [OSL | CCIE_Voice] FTP from CUE failed Filezilla server provides a console log that's typically verbose enough to get a point in the right direction. Do you see a connection attempt with some log messages? On Tue, Jun 4, 2013 at 12:42 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi all, When i try to install license from FTP to CUE, it failed and gave me error message : Download error, Can not download cue-vm-license_12mbx_7.0.1.pkg error code 0 error type 'couldn't connect to host - I have verified the FTP server (Filezilla server) is working using Filezilla client . - username and password is correct - File license is valid What possibility here? tks K ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] FTP from CUE failed
It's pretty strange that you have ping connectivity but it's not answering. Can you telnet to the FTP server from the switch? Also, BR2 is directly connected to 3750? No WAN setup to route traffic from HQ servers to BR2? On Tue, Jun 4, 2013 at 7:00 PM, Karen Johnson karen.johnson...@yahoo.cawrote: I see nothing in Filezilla Server *From:* Bill whl...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* Martin Sloan martinsloa...@gmail.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 4, 2013 4:59:03 PM *Subject:* Re: [OSL | CCIE_Voice] FTP from CUE failed Sounds like you are dropping the return packets but not sure why What do you see in the FileZilla server? Sent from my iPad On Jun 4, 2013, at 5:16 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: yes i did not see when I do from CUE but when i do from Filezilla client, I see *From:* Martin Sloan martinsloa...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 4, 2013 12:01:01 PM *Subject:* Re: [OSL | CCIE_Voice] FTP from CUE failed Filezilla server provides a console log that's typically verbose enough to get a point in the right direction. Do you see a connection attempt with some log messages? On Tue, Jun 4, 2013 at 12:42 PM, Karen Johnson karen.johnson...@yahoo.cawrote: hi all, When i try to install license from FTP to CUE, it failed and gave me error message : Download error, Can not download cue-vm-license_12mbx_7.0.1.pkg error code 0 error type 'couldn't connect to host - I have verified the FTP server (Filezilla server) is working using Filezilla client . - username and password is correct - File license is valid What possibility here? tks K ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] FTP from CUE failed
If the FTP server answers when plugged into BR2 it sounds like the FTP part is good. Does the server answer when you telnet to it on port 21 from the switch? Eliminate every possible 'in between' issue and work out until it fails. To me, it does seem like a different setup to not have any WAN connection in between the 3750 and BR2 though. Why is it connected directly to the switch and the HQ servers? Where's Frame Relay!? On Tue, Jun 4, 2013 at 8:39 PM, Bill Lake whl...@gmail.com wrote: It should not be a VMware setting Where are you running the FileZilla server? Can the cue ping the ip of this and do you see the packets actually get to this server? What version of windows are you running the FileZilla on? What is the host system? Can you post your configs of the br2 and 3750? Sent from my iPhone On Jun 4, 2013, at 5:40 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: ping wotk both ways and no firewall block issue. most likely my VM setting, I choose Bridged in VM workstation. is that ok. here is my diagram VM workstation server (PUB,SUB,UCCX,CUC) --- 3750 --- BR2 router (CUE) *From:* Martin Sloan martinsloa...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 4, 2013 4:31:29 PM *Subject:* Re: [OSL | CCIE_Voice] FTP from CUE failed Sounds like it's a routing/acl issue then. Can you ping to/from the CUE/FTP server? Are there any devices/ACL's in between that would block this? If the CUE is a few hops away and you have devices that are closer to the FTP server you could start at the closest one and work your way out and telnet to port 21 until it breaks down. If the port is open, it will answer. If you're using IOS devices just make sure that 'transport output telnet' is set under you con/vty line, then 'telnet ip of ftp server 21' and see if it answers. My first step would be standard ping in both directions though. On Tue, Jun 4, 2013 at 6:16 PM, Karen Johnson karen.johnson...@yahoo.cawrote: yes i did not see when I do from CUE but when i do from Filezilla client, I see *From:* Martin Sloan martinsloa...@gmail.com *To:* Karen Johnson karen.johnson...@yahoo.ca *Cc:* ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 4, 2013 12:01:01 PM *Subject:* Re: [OSL | CCIE_Voice] FTP from CUE failed Filezilla server provides a console log that's typically verbose enough to get a point in the right direction. Do you see a connection attempt with some log messages? On Tue, Jun 4, 2013 at 12:42 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi all, When i try to install license from FTP to CUE, it failed and gave me error message : Download error, Can not download cue-vm-license_12mbx_7.0.1.pkg error code 0 error type 'couldn't connect to host - I have verified the FTP server (Filezilla server) is working using Filezilla client . - username and password is correct - File license is valid What possibility here? tks K ___ For more information regarding industry leading CCIE Lab training, please visit http://www.ipexpert.com/ Are you a CCNP or CCIE and looking for a job? Check out http://www.platinumplacement.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Voice Retirement
Awesome! On Sat, Jun 1, 2013 at 11:10 PM, Bill Lake whl...@gmail.com wrote: Dear Ben, In the last several years I have installed many Cisco Telepresence 3200/3000, 1300, 500 and 9200/9000 series systems. I have done work with personal video conferencing systems on executive desktops. I have also been involved in many installations of VoIP systems from around 50 to over 3000 phones at one location. I have also installed more software clients that support voice and video than I care to remember. I hope I am the very type of person that Cisco wants to earn their CCIE Voice. I earned that CCIE Voice this year to open doors for me and afford me some opportunities that were weren't there before. Now just a short time after earning my CCIE with very relevant skills in today's market, Cisco has decided to close the door on CCIE Voice and introduce CCIE Collaboration. This clearly is within Cisco's right to do and as a name, it better fits what communications have evolved into. That said, the underlying technology and ability to work on it have not changes. Yes video has different requirements than voice but once you learn to provide good voice quality, you can leverage that to provide good video quality. The same is true of almost all of the new technologies, from dialing to software clients that are introduced in CCIE Collaboration. So when Cisco has decided in the past to retire a CCIE exam, it was due to the massive technology changes that in general left the CCIE track untenable. If Cisco had decided to leave the name CCIE Voice and included the new tasks, it would still be relevant as it is with the new name. You can not say the same for CCIE ISP Dial, CCIE SNA/IP integration and so on. As a technology they are not dead but are completely untenable as a CCIE track but . Cisco can not logically take the same stance that Voice is a dead technology and is not integral to the new CCIE Collaboration. Since voice is so integral to the CCIE Collaboration, I would consider it to be more a change in name than technology. In retiring CCIE Voice and introducing CCIE Collaboration, Cisco has punished CCIE Voice holders like never before. Even with their skills present and relevant to the CCIE track, they have been told that the only way to achieve the new CCIE Collaboration title is to pass the lab. This is hard to believe as other tracks have changed far more over their lives and especially for those that passed CCIE Voice V3. A perfect example of this is CCIE RS from the early 2000's. During that exam candidates had to earn their strips on technologies like token ring, IPX and other similarly dead protocols. They are allowed to remain CCIE RS by passing every 2 years a CCIE level written exam, any exam, so they don't even need to prove they are keeping current on RS. So it seems that Cisco is interested mostly in CCIE's keeping current in today's technology and not so much with your CCIE track. That seems completely tossed out the window with CCIE voice. CCIE Voice is so integral to CCIE Collaboration that you can't logically argue that voice is a dead technology and you must earn your CCIE Collaboration by passing another Voice centric lab in the CCIE Collaboration. It is completely within Cisco's right to demand that anyone pass the CCIE Collaboration to earn the title. It is however with great hope that the logical argument laid out here will help Cisco change paths on this and offer a different path to current holders of CCIE Voice. Cisco could easily create or use the CCIE Collaboration written exam to ensure that people who have earned their CCIE Voice continue to keep up with the ever changing technology. Cisco could also make a one time exam that is perhaps more challenging than the normal written exam but less demanding of time, travel and expense than a full blown CCIE lab. I believe that Cisco could easily integrate simulations into either exam type that would ensure that those who have earned their CCIE voice are keeping up with technology. That is what this change to CCIE Collaboration is about, better reflecting the requirements in the field and the technology we deal with. Sincerely, Bill Lake On Fri, May 31, 2013 at 8:02 AM, Martin Sloan martinsloa...@gmail.comwrote: Ben, I'm writing you this morning to express my great disappointment in regard to Cisco's recent announcement to retire the CCIE Voice track with no reasonable upgrade path to the CCIE Collaboration. I know that you're well informed as to all the arguments which are being made against this decision on Facebook, Twitter and other social media outlets, so I won't go into any detail on why I think this is a bad decision. At this point, the facts are well laid out for everyone to see. I'd like to ask you to please reconsider this decision and provide a reasonable upgrade path for the certified CCIE Voice candidates. I, myself, am not yet
Re: [OSL | CCIE_Voice] Collaboration
Victor, I agree, thanks very much to John for setting up the petition. Also, let me apologize if you or anybody else considers my own email to Ben Ng yesterday as being hasty. My intention was start one of many in a line that communicates directly to him the sentiments of this community heard from the perspective of the individuals. We're all affected by this decision and I've been in a state of disbelief since hearing it. I'm eager to do anything to be heard. Thanks, Marty On Sat, Jun 1, 2013 at 12:39 AM, Victor Voice vie.c...@gmail.com wrote: Hello everyone, First of all, I would like to take this opportunity to express my heartfelt thanks to John Welsh for creating the petition. I need to discuss with other members the following ideas: I suggest sending up to 10 e-mails to: Ben Ng (CCIE Voice Program Manager) and Cc some decision makers in Cisco like: John Chambers. If you know the name of other decision makers just mention the name and we will work together to bring their e-mails. E-mail subjects should be attractive so please suggest some subjects. The contents of the e-mails should be suitable for higher management. So we have to determine the headlines that we have to mention. Create a page on Facebook. It will be better if we can add some graphs. Post more complaints and response to the existing in Cisco Learning Network under CCIE Collaboration and CCIE Voice Add the links of the petition and face-book page to the e-mail. Please do not do any action unless we agree on it. Regards, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Advice or opinions on Vol 2 Lab 4 Task 5.1
The task involves configuring LLQ, FRF.12, LFI and cRTP from HQ-BR1 while using class-based shaping and cRT. The solutions guide and video walk through recommend not using Auto QoS and creating the configs manually instead since Auto QoS will enable FRTS and shaping in the map class. I'm doing this both ways and, for me, it seems that turning on Auto QoS (auto qos voip) and then modifying is a little quicker. Using this technique, I would: 1) Create my class-based shaper policy map and assign the Auto QoS policy map to it (after tweaking bandwidth amounts and cRTP, etc) policy-map CB-SHAPE class class-default shape average 365000 3560 0 service-policy AutoQoS-Policy-UnTrust 2) Remove FRTS from the interface interface Serial0/1/0 no frame-relay traffic-shaping 3) Remove shaping commands from the map class, remove the Auto QoS output policy and add mine from above map-class frame-relay AutoQoS-FR-Se0/1/0-201 no frame-relay cir 365000 no frame-relay bc 3650 no frame-relay be 0 no frame-relay mincir 365000 no service-policy output AutoQoS-Policy-UnTrust service-policy output CB-SHAPE One could argue 'six in one, half dozen in another' from a speed standpoint, but based on just getting the points, are there any issues with this approach? Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration
Thanks, Suresh. Counts went up by 100 overnight. Keep it going!! On Fri, May 31, 2013 at 12:40 AM, Suresh Bhandari bring...@gmail.comwrote: *Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration *- Sign the Petition! For the interested candidates... Please join this campaign: http://chn.ge/17A0zXE I already did. Now its your turn. Initially shared by Martin Sloan (martinsloa...@gmail.com) -- Suresh Bhandari ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] uccx login showing unauthorized...
Amit, You could check the user account credentials by trying to log in to the CUCM end user page with that account. If you're not able to fix the issue I know you can use cet.bat on the UCCX server to set the install state back to 'FRESH_INSTALL' and go through the setup process again. It's a pain but it will allow you to set the admin user again. Marty On Fri, May 31, 2013 at 10:31 AM, Amit Sharma aryan231...@gmail.com wrote: please help how can fix it? when i login uccx with any browser it giving unauthorized message... -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Cisco: Provide a reasonable transition path from CCIE Voice to CCIE Collaboration - Sign the Petition! Please join this campaign: http://chn.ge/17A0zXE On Thu, May 30, 2013 at 2:01 PM, probert...@gmail.com probert...@gmail.comwrote: Watching this video is ironic: Ben Ng at Cisco Live 2012: http://www.ustream.tv/recorded/23271405 But you can hear where the idea for the new name came from. On Thu, May 30, 2013 at 9:29 AM, Brian Schear brian.sch...@vitalsite.comwrote: It is Ben Ng. Found his linked in profile below which describes his position in Cisco. www.linkedin.com/pub/ben-ng/3/509/940 Profile on the Cisco site. https://www.cisco.com/web/learning/le31/communities/netpro/bios/benng.html Anyone have better contact info to send him respectful and thoughtful arguments on this? Brian -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Rrcrumm Sent: Wednesday, May 29, 2013 11:11 AM To: Leslie Meade Cc: ccie_voice@onlinestudylist.com; vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto: whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_ tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom http://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.com mailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't big buck makers at Cisco just rename a Cert rather than do something completely rubbish. With just one announcement, they have made many people lose faith in Certification
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
I'd love to get some traction on an effort to fix this. Please count me in for anything I can do to let our concerns be heard. I'm sure I'm not alone here but I'm embarrassed to tell my wife that I've invested our families time and money into something that's EOL. Not a very wise decision on my part. Epic fail! On Wed, May 29, 2013 at 11:39 AM, m george m.george00...@gmail.com wrote: VIk/IPExpert Team, Does anyone knows who the Program Manager is for Upcoming Collaboration Track ? We need to voice our concerns to PM's direct email address on Cisco Support Community. That's only way, we will be heard. Let's hope Cisco listens to us. Regards, On Wed, May 29, 2013 at 8:29 PM, Bill Lake whl...@gmail.com wrote: I agree that in retiring the exam and requiring that you retake the lab portion again is incomprehensible. They can't tell me that the RS hasn't changed as much or more over its lifetime. It is still the same but they did not retire it (well maybe that is the plan, retire them all and make you earn new) so if you got your RS 10 years or 10 days ago you are CCIE RS. You can easily say the same for others but you get the idea. I think that this is marketing and even so they could have easily done exactly what they did with CCVP to CCNP Voice. When you renew, you do so by passing the CCIE Collaboration written exam (which they make more like the others with some interactive tasks) and you then renew as a CCIE Collaboration. I just think we should stop complaining, organize the CCIE voice community and ask nicely, demand persuasively and argue smartly to get them to change their minds about having to take the lab again to move to CCIE Collaboration. What they have done is weaken in my mind what I strove so hard to earn Bill On 5/29/13, Mark Holloway m...@markholloway.com wrote: Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom acquisition) ISP Dial CCIE SNA/IP Integration CCIE (aka CCIE Blue) Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto: whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom http://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.com mailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't big buck makers at Cisco just rename a Cert rather than do something completely rubbish. With just one announcement, they have made many people lose faith in Certification process. I am sure Voice labs will be the most deserted labs until Feb 2014. At the end of day, we can only request Cisco to re-consider this decision. I hope folks concerned collaborate put their suggestions forward on Cisco Support Community direct to Cisco Certification teams so they realize what they are doing is NOT right. I will take some months for us to digest this news. Thanks On Wed, May 29, 2013 at 12:27 PM, Vik Malhi vma...@ipexpert.commailto: vma...@ipexpert.com wrote: As I said before - I would think product marketing had something to say about this. Just my opinion. Why for the last 4 years has there been a lack of Microsoft products in
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
I couldn't give you a +1 on the Cisco site so let me offer the +1 here. Well put, very concise and totally accurate. I completely agree with you. I vote that you are 'The voice of The Voice'. Bitching may not work, but it makes me feel better :-D On Wed, May 29, 2013 at 1:07 PM, William Bell b...@ucguerrilla.com wrote: When you have a group of people that share an opinion, you need to organize that group of people so that they can speak as one voice. It is called Unified Communications for a reason! The key is to have this group opinion communicated across multiple mediums in a consistent and persistent manner. Basically, you have to market your message. Twitter, FB, and the Cisco Communities are good target mediums if you want to get Cisco's attention. Finding out who is in charge of the IE Voice/Collaboration program and getting their email is another medium. Though, the recipient of said email bomb won't look on that with favorable eyes and it may be counterproductive. Bitching for the sake of bitching won't work. You also have to make sure your argument is one that has a chance of appealing to the other party's willingness or ability to make a compromise. For instance, bitching at Cisco and saying they should rethink retiring the IE voice and grandfather us in may not work. However, launching a campaign to convince them that there should be an alternate path for the IE voice to upgrade their IE may provide a more workable compromise. Thus far I have spoken about organizing our complaints to get attention and putting out a message that provides a reasonable and workable compromise. Cisco has and will listen to that messaging. It has a chance if you say it loud and often. The whole squeaky wheel thing. If you had a way to show that this move costs Cisco money then you would have an even more effective weapon. This is a little harder to conceptualize and even harder to convince everyone to do what would need to be done. -Bil -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 29, 2013, at 10:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.com mailto:whl...@gmail.com whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom http://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.commailto:
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
I thought so too. I could see if it was getting obnoxious but all of the comments were pretty professional. On Wed, May 29, 2013 at 4:12 PM, William Bell b...@ucguerrilla.com wrote: They did disable commenting. That's interesting. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On May 29, 2013, at 2:24 PM, Martin Sloan wrote: Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto: whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom http://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george m.george00...@gmail.com mailto:m.george00...@gmail.com wrote: Vik, A 2nd grader can pass RS/Sec/SP much much easier than Voice IE. Voice is still the toughest one i know some double IE's who couldn't pass Voice. If Cisco has lost faith in re-cert, that should apply to every track, not just Voice. Naturally, they should have renamed Certification to Voice/Collaboration or Voice/Video etc introduced new version. If they had to do this retiring thing, why didn't they do when they introduced V3 from V2 ? Old days of Call Manager based on Windows literally everything based on windows, Analog endpoints/VGs/ATAs etc. Retiring CCIE Voice makes no sense whatsoever whether be it from marketing point of view or any other. Why can't big buck makers at Cisco just rename a Cert rather than do something completely rubbish. With just one announcement, they have made many people lose faith in Certification process. I am sure Voice labs will be the most deserted labs until Feb 2014. At the end of day, we can only request Cisco to re-consider this decision. I hope folks concerned collaborate put their suggestions forward on Cisco Support Community direct to Cisco Certification teams so
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Thanks, Bill. Just posted my first Tweet! Daniel - I had to borrow your great point about the changes from v2 to v3 being more significant than the current changes. Awesome insight. I joked when Vik first posted about collaboration that I didn't want to be a CCIE-Ambigouos Marketing Jargon. I'm eating my words and they taste terrible. On Wed, May 29, 2013 at 4:40 PM, Daniel Pagan dpa...@fidelus.com wrote: Not just you – I also cannot post a response. ** ** Great posting, Bill. I appreciate that you expressed what many of us are feeling right now in a very articulate and logical manner. ** ** I’m in complete agreement with nearly every response in this thread. It’s rather upsetting to see this entire track get retired when its replacement’s blueprint is simply a needed refresh. In fact, it seems the blueprint changes made during the transition from lab v2 to v3 were greater in comparison to this (CUPS added, UC v7 platforms added incl. UnityCx, ISRs added and 6608s and VG248 removed, core knowledge questions added, etc.). I’m reading the lab topics for v4 and see nothing that couldn’t be included in a “migration exam” for current voice CCIEs. ** ** Daniel Pagan, CCIE #25689 ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Martin Sloan *Sent:* Wednesday, May 29, 2013 2:25 PM *To:* Rrcrumm *Cc:* ccie_voice@onlinestudylist.com; vma...@ipexpert.com *Subject:* Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced* *** ** ** Is it just me or did they disable commenting on the v4 lab topics post? https://learningnetwork.cisco.com/docs/DOC-20804 I wanted to commend William Bell on hitting the nail on the head and put my own 2 cents in. I'm able to place a comment in the equipment list post here: https://learningnetwork.cisco.com/docs/DOC-20804 but not the other. Does anyone have comment options on the exam topics page? ** ** On Wed, May 29, 2013 at 12:10 PM, Rrcrumm rrcr...@yahoo.com wrote: Ben Ng comes to mind On May 29, 2013, at 7:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.commailto: whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several others. See here with the snippet. Now the second is a wiki so we would want official confirmation. https://learningnetwork.cisco.com/docs/DOC-17226 http://en.wikipedia.org/wiki/Cisco_Career_Certifications#Retired_CCIE_tracks Retired CCIE tracks Some previously awarded CCIE specializations have been retired by Cisco. These are: * WAN Switching CCIE (Essentially a specialisation focusing on the IGX/BPX switch products, which had been acquired as part of the StrataCom http://en.wikipedia.org/wiki/StrataCom acquisition) * ISP Dial CCIE * SNA/IP Integration CCIE (aka CCIE Blue) * Design CCIE (NOTE: The CCIE Design and CCDE are completely different design tests in format and subjects examined) People who hold these now-retired certifications can remain CCIEs, provided they continue to take recertification exams. They now hold the title CCIE, rather than CCIE Security, or some other specialization. So if we can get official confirmation that we won't be stripped of CCIE if you pass the voice lab, it might be good, for those of us that have already passed we don't get a chance to change our minds for those that have yet to pass, this might be incentive to change your goal. Bill On Wed, May 29, 2013 at 3:29 AM, m george
Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced
Great pep talk Josh, I'm right before you on 7/29 in RTP and will keeping focused on it. I hope they reconsider or provide an alternate update path but if not I'm already paid up for the CCIE-V, might as well go for it. Good luck on your lab. On Wed, May 29, 2013 at 6:25 PM, Josh Petro josh.pe...@gmail.com wrote: I wasn't going to chime in, but given some of the responses I feel led to put in my two cents. My plan is to sit the exam August 1st (already paid for) and get CCIEVoice. I agree that it's a shame Cisco is changing the name, but like many of you said, you still have the number. I would also bet that most if not all of you have had some video experience, so if you decide to try for the Collaboration cert and already have a CCIE Voice, it *should* be fairly straight forward. If you're in the same boat as I am, my advice is to keep on going. You've already spent X amount of months studying (and time away from family), so keep going, don't procrastinate and get it done! I hope that reads more of a pep talk for my fellow candidates, rather than a rant. Oh, and I'm up for voicing an opinion to Cisco about this, but I would doubt they would shift policy because of us - but who knows. Josh On Wed, May 29, 2013 at 3:15 PM, Martin Sloan martinsloa...@gmail.comwrote: I couldn't give you a +1 on the Cisco site so let me offer the +1 here. Well put, very concise and totally accurate. I completely agree with you. I vote that you are 'The voice of The Voice'. Bitching may not work, but it makes me feel better :-D On Wed, May 29, 2013 at 1:07 PM, William Bell b...@ucguerrilla.comwrote: When you have a group of people that share an opinion, you need to organize that group of people so that they can speak as one voice. It is called Unified Communications for a reason! The key is to have this group opinion communicated across multiple mediums in a consistent and persistent manner. Basically, you have to market your message. Twitter, FB, and the Cisco Communities are good target mediums if you want to get Cisco's attention. Finding out who is in charge of the IE Voice/Collaboration program and getting their email is another medium. Though, the recipient of said email bomb won't look on that with favorable eyes and it may be counterproductive. Bitching for the sake of bitching won't work. You also have to make sure your argument is one that has a chance of appealing to the other party's willingness or ability to make a compromise. For instance, bitching at Cisco and saying they should rethink retiring the IE voice and grandfather us in may not work. However, launching a campaign to convince them that there should be an alternate path for the IE voice to upgrade their IE may provide a more workable compromise. Thus far I have spoken about organizing our complaints to get attention and putting out a message that provides a reasonable and workable compromise. Cisco has and will listen to that messaging. It has a chance if you say it loud and often. The whole squeaky wheel thing. If you had a way to show that this move costs Cisco money then you would have an even more effective weapon. This is a little harder to conceptualize and even harder to convince everyone to do what would need to be done. -Bil -- William Bell, CCIE #38914 blog: http://ucguerrilla.com Follow me on twitter @ucguerrilla On May 29, 2013, at 10:28 AM, Leslie Meade leslie.me...@lvs1.com wrote: The question is... what if anything can we do ? Where would we start.. Original message From: Mark Holloway m...@markholloway.com Date: To: Bill Lake whl...@gmail.com Cc: OSL Group ccie_voice@onlinestudylist.com,Vik Malhi vma...@ipexpert.com Subject: Re: [OSL | CCIE_Voice] CCIE Collaboration officially announced Granted we all know that taking any CCIE Written will allow us to remain CCIE's even if Voice is retired, but I think the frustration is Voice and Collaboration are not THAT far apart and no matter how you look at it, it all falls under Cisco Unified Communications, which is what the name of the new CCIE really should be anyway. The core of the Voice blueprint is still there. The Collaboration equipment list looks like a refresh of current products, not a forklift of one technology replacing another. In my opinion this was too harsh of a move to retire Voice and start over again with Collaboration. There are too many similarities between the two. On May 29, 2013, at 7:10 AM, Bill Lake whl...@gmail.com mailto:whl...@gmail.com whl...@gmail.com wrote: Ranting about it won't change anything. I read on line that when they retire your CCIE, you can still renew by passing a CCIE level written or lab. If this is true then you do not loose your CCIE just the voice tag. That seems to be a difficult pill to swallow but it would not be the first from my reading. Storage had this happen earlier this year as have several
[OSL | CCIE_Voice] Voice translation issue
I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice translation issue
Thank you for the help. I can get it to work with a no strip or forward digits all but from a logical standpoint, the config should work, right? I have all my other dial-peers and translations working for local, ld and intl calls without digit manipulation on the dial-peer directly, but this particular one is not working. On a totally different topic, should I be adding '[OSL | CCIE_Voice]' to my email? I figured that was tacked on automatically but I don't see it on my posts. Thanks. On Sun, May 19, 2013 at 10:07 AM, Ravindra Lakpriya lakpr...@gmail.comwrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice translation issue
Here's the config from another dial-peer that's working without digit manipulation on the dial-peer itself. I guess my question is why would I need to perform digit manipulation directly on the dial-peer for 9911 and not for the LD dial-peer below? dial-peer voice 10 pots translation-profile outgoing LD destination-pattern 91[2-9]..[2-9]..$ port 0/0/0:23 voice translation-profile LD translate calling 5 translate called 5 voice translation-rule 5 rule 1 /^91/ /1/ type any national plan any isdn rule 2 /^1...$/ /617394\0/ type any national plan any isdn On Sun, May 19, 2013 at 10:07 AM, Ravindra Lakpriya lakpr...@gmail.comwrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice translation issue
Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice translation issue
If I change the translation to this: voice translation-rule 8 rule 1 /^9911$/ /99911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn It sends '911' to the PSTN! The voice translation debug looks like this: *May 19 14:39:44.716: //-1/C76CA681803C/RXRULE/regxrule_dp_translate: calling_number=6173941002 calling_octet=0x41 called_number=99911 called_octet=0x1 I just don't understand the logic on this one. I know there's more than 1 way to skin this cat but it bugs the heck out of me to not understand this. I look at translations as a /MATCH/ /REPLACE/ setup with '/^9911$/' matching the dialed digits and then replacing the entire matched string with the replacement string '/911/'. I can't wrap my head around what's going on with this one. I tried it on the BR2 and HQ gateways and got the same results. On Sun, May 19, 2013 at 10:41 AM, Martin Sloan martinsloa...@gmail.comwrote: Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Voice translation issue
Bruno, thanks for giving it a try. I feel a little better about my sanity now :-) It is very strange and I also had the same results as you, with only the '9' exhibiting this behavior in the replace string. I was hoping to use translation rules for all my dial-peer digit manipulation but with this issue coming up, I think I'll use a forward digits command on the dial peer for 911/9911. I would love to know what's going on here though! On Sun, May 19, 2013 at 1:26 PM, Bruno Takahashi brun...@gmail.com wrote: That´s a good one! I've tried it out here and indeed does behaviour like that.. and I still can't figure out what on earth is happening that the Outpulsed digits are just 11 If you replace with /9911/, it sends only 1 !! It seems something is wrong with the 9's, because if you try to replace with /123/ for example, it works just fine. But when the replace pattern leads with a 9 something goes wrong.. Something misterious is happening in this translation-rule.. please let us know if you find out! I gave up on this one already On Sun, May 19, 2013 at 12:13 PM, Martin Sloan martinsloa...@gmail.comwrote: If I change the translation to this: voice translation-rule 8 rule 1 /^9911$/ /99911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn It sends '911' to the PSTN! The voice translation debug looks like this: *May 19 14:39:44.716: //-1/C76CA681803C/RXRULE/regxrule_dp_translate: calling_number=6173941002 calling_octet=0x41 called_number=99911 called_octet=0x1 I just don't understand the logic on this one. I know there's more than 1 way to skin this cat but it bugs the heck out of me to not understand this. I look at translations as a /MATCH/ /REPLACE/ setup with '/^9911$/' matching the dialed digits and then replacing the entire matched string with the replacement string '/911/'. I can't wrap my head around what's going on with this one. I tried it on the BR2 and HQ gateways and got the same results. On Sun, May 19, 2013 at 10:41 AM, Martin Sloan martinsloa...@gmail.comwrote: Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.comwrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling Party Number i = 0x4181, '6173941002' Plan:ISDN, Type:Subscriber(local) Called Party Number i = 0x81, '11' Plan:ISDN, Type:Unknown -Debug voice translation- *May 19 13:38:29.615: //-1/394E525380FD/RXRULE/regxrule_vp_translate: calling_number=6173941002 calling_octet=0x41 called_number=911 called_octet=0x1 From testing the voice translation and checking the translation debugs, it looks like everything works but the gw sends only '11' to the PSTN. Can someone please school me on this one? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Ravindra Lakpriya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE
Re: [OSL | CCIE_Voice] Voice translation issue
I'm running c2800nm-adventerprisek9_ivs-mz.124-24.T8.bin on BR1 and also tried it on HQ with c3825-adventerprisek9_ivs-mz.124-24.T8.bin. I have the same issue with regular 911 calls as well. I can get it to work by padding more 9's though. So the translations that work look like: voice translation-rule 7 rule 1 /^911/ /9911/ type any unknown plan any isdn voice translation-rule 8 rule 1 /^9911/ /99911/ type any unknown plan any isdn The Cisco learning network says about the IOS All routers use IOS version 12.4T Train. I hope it's Bill's version! On Sun, May 19, 2013 at 7:44 PM, Bruno Takahashi brun...@gmail.com wrote: flash0:c2900-universalk9-mz.SPA.152-4.M1.bin here On Sun, May 19, 2013 at 6:20 PM, Bill whl...@gmail.com wrote: What ios are you guys running? I don't see this happening in ios I am running flash:c2800nm-adventerprisek9_ivs_li-mz.124-20.T.bin Sent from my iPad On May 19, 2013, at 12:54 PM, Martin Sloan martinsloa...@gmail.com wrote: Bruno, thanks for giving it a try. I feel a little better about my sanity now :-) It is very strange and I also had the same results as you, with only the '9' exhibiting this behavior in the replace string. I was hoping to use translation rules for all my dial-peer digit manipulation but with this issue coming up, I think I'll use a forward digits command on the dial peer for 911/9911. I would love to know what's going on here though! On Sun, May 19, 2013 at 1:26 PM, Bruno Takahashi brun...@gmail.comwrote: That´s a good one! I've tried it out here and indeed does behaviour like that.. and I still can't figure out what on earth is happening that the Outpulsed digits are just 11 If you replace with /9911/, it sends only 1 !! It seems something is wrong with the 9's, because if you try to replace with /123/ for example, it works just fine. But when the replace pattern leads with a 9 something goes wrong.. Something misterious is happening in this translation-rule.. please let us know if you find out! I gave up on this one already On Sun, May 19, 2013 at 12:13 PM, Martin Sloan martinsloa...@gmail.comwrote: If I change the translation to this: voice translation-rule 8 rule 1 /^9911$/ /99911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn It sends '911' to the PSTN! The voice translation debug looks like this: *May 19 14:39:44.716: //-1/C76CA681803C/RXRULE/regxrule_dp_translate: calling_number=6173941002 calling_octet=0x41 called_number=99911 called_octet=0x1 I just don't understand the logic on this one. I know there's more than 1 way to skin this cat but it bugs the heck out of me to not understand this. I look at translations as a /MATCH/ /REPLACE/ setup with '/^9911$/' matching the dialed digits and then replacing the entire matched string with the replacement string '/911/'. I can't wrap my head around what's going on with this one. I tried it on the BR2 and HQ gateways and got the same results. On Sun, May 19, 2013 at 10:41 AM, Martin Sloan martinsloa...@gmail.com wrote: Thanks, Bill. Any thoughts on why the gw is only sending '11' to the PSTN? If the dial-peer is stripping explicitly matched digits it should strip all of the digits. It just doesn't make any sense to me that the voice translation debug and test shows that the digit manipulation happens correctly but the gw sends only '11'. I'm really confused about that! On Sun, May 19, 2013 at 10:33 AM, Bill whl...@gmail.com wrote: voice translation-rule 8 *rule 1 // // type any unknown plan any isdn* rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn Sent from my iPad On May 19, 2013, at 9:07 AM, Ravindra Lakpriya lakpr...@gmail.com wrote: In the dial peer configure no digit strip. :) On Sun, May 19, 2013 at 5:47 PM, Martin Sloan martinsloa...@gmail.com wrote: I have a voice translation rule in place for '9911' calls on BR1 during SRST. I'm running into some odd behavior (from my perspective) and I'm hoping it's a config issue I'm just not spotting. I have the translation profile applied to the dial peer and the only other translation that would be in the calling path is on the voice port but even that one is applied to inbound calls for stripping down to 4 digits. Here's the config related to this dial-peer: voice translation-rule 8 rule 1 /^9911$/ /911/ type any unknown plan any isdn rule 2 /^1...$/ /617394\0/ type any subscriber plan any isdn voice translation-profile 9911 translate calling 8 translate called 8 dial-peer voice 9911 pots translation-profile outgoing 9911 destination-pattern 9911$ port 0/0/0:23 BR1-RTR#test voice translation-rule 8 9911 Matched with rule 1 Original number: 9911 Translated number: 911 Original number type: none Translated number type: unknown Original number plan: none Translated number plan: isdn -Debug ISDN q931- Calling
Re: [OSL | CCIE_Voice] CCIE Voice renamed CCIE Collaboration available Nov 2013
Thanks,Vik. Hopefully I can pass on the current blueprint so I'm a 'CCIE - Voice' and not a 'CCIE Ambiguous Marketing Jargon' :-) On Wed, May 15, 2013 at 12:31 PM, Vik Malhi vma...@ipexpert.com wrote: More info to come- but we've all been waiting a long time to hear some news. People in the middle of their studies hoping to pass on the current blueprint- your countdown begins now. Vik ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] shortcut command
How about: sh run | s ephone-dn 1 sh run | s ephone 2 Note there are 2 spaces in between ephone-dn and 1 2 spaces between ephone and 2. On Tue, May 14, 2013 at 6:16 PM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi some time i need to see only particular ephone setting/or ephone-dn setting on CME.is there any shortcut command like show ephone | like that ...on CME please share.. -- Regards, Dharambir Kumar ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab IP Routing
Ali, Great question and I'm not an expert here but as I understand it, in this instance, 255's and 0's accomplish the same thing but 255's are the 'proper' method and 0's are an alternate way to do it. Marty On Wed, May 1, 2013 at 7:02 PM, ali raza ccie2...@gmail.com wrote: shouldn't it be? ** ** router ospf 1 network 0.0.0.0 0.0.0.0 ar 0 ** ** regards, ** ** ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Josh Petro *Sent:* Wednesday, May 01, 2013 10:57 PM *To:* Martin Sloan *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab IP Routing ** ** No worries, but thanks for the correction and follow up! On May 1, 2013 9:42 AM, Martin Sloan martinsloa...@gmail.com wrote:*** * Josh, ** ** I realized that my config is bunk but you prob spotted that. I didn't include the mask, my bad. ** ** router ospf 1 network 0.0.0.0 255.255.255.255 area 0 ** ** HTH ** ** On Wed, May 1, 2013 at 8:35 AM, Josh Petro josh.pe...@gmail.com wrote:** ** Thanks everyone. I have a pretty good OSPF base understanding, so I'll brush up and make sure I've got it all down. I'm sure I'm like most of you and have more EIGRP experience, but I have run into OSPF installs here an there. Thanks for the explanation on the telnet. I wasn't sure if they limited the vty access or limited which commands you could run. Some of the practice labs say things like 'do this or that without using automated commands'. I'm assuming that's to hone our manual config skills, rather than prep us for potential lab requirements. Anyhow, thanks again. Josh ** ** On Wed, May 1, 2013 at 5:35 AM, Jamie Parr (jamparr) jamp...@cisco.com wrote: Hi Josh The network in the lab is as real as it can be, no reason you couldn’t telnet to any of the routers. The quickest way I found to complete the lab is to write everything out in notepad first for all devices, so the speed of the connection isn’t an issue. I barely touched the network devices after pasting in the config HTH *Jamie Parr* CCIE #38633 (voice) Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641 %2B44%2020%208824%202641* Mobile: *+44 7590622049* *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Martin Sloan *Sent:* 01 May 2013 03:43 *To:* Josh Petro *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab IP Routing Josh, For the OSPF question, my current plan is to issue the below as part of a base config on all routers to make sure I'm advertising the local networks and hopefully negate any built-in troubleshooting for the routing piece:** ** router ospf 1 network 0.0.0.0 area 0 I'm interested to hear others opinions though. Marty On Tue, Apr 30, 2013 at 10:12 PM, Josh Petro josh.pe...@gmail.com wrote: Hi everyone, I'm sorry for asking this question, but it hit me a little while ago that for the most part, routing via OSPF is setup on most (if not all) of the practice labs. Is that the case in the real lab? I'm trying to make sure I don't need to brush up on my OSPF commands prior to my attempt. I know the previous versions of the lab had more IP routing in them, but I wasn't sure what this version was like. Also, I'm assuming our connection to the lab is via console cable and there is no way to telnet to the gateways, correct? Reason I ask is because we're always talking about speed and whatnot and 9600 baud isn't exactly what you'd want so I've been practicing my command line switches for finding the correct running-config commands. Any info (without breaking NDA) is appreciate. Josh ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab IP Routing
Josh, I realized that my config is bunk but you prob spotted that. I didn't include the mask, my bad. router ospf 1 network 0.0.0.0 255.255.255.255 area 0 HTH On Wed, May 1, 2013 at 8:35 AM, Josh Petro josh.pe...@gmail.com wrote: Thanks everyone. I have a pretty good OSPF base understanding, so I'll brush up and make sure I've got it all down. I'm sure I'm like most of you and have more EIGRP experience, but I have run into OSPF installs here an there. Thanks for the explanation on the telnet. I wasn't sure if they limited the vty access or limited which commands you could run. Some of the practice labs say things like 'do this or that without using automated commands'. I'm assuming that's to hone our manual config skills, rather than prep us for potential lab requirements. Anyhow, thanks again. Josh On Wed, May 1, 2013 at 5:35 AM, Jamie Parr (jamparr) jamp...@cisco.comwrote: Hi Josh ** ** The network in the lab is as real as it can be, no reason you couldn’t telnet to any of the routers. The quickest way I found to complete the lab is to write everything out in notepad first for all devices, so the speed of the connection isn’t an issue. I barely touched the network devices after pasting in the config ** ** HTH ** ** *Jamie Parr* CCIE #38633 (voice) Engineer - IT jamp...@cisco.com Phone: *+44 20 8824 2641* Mobile: *+44 7590622049* ** ** *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Martin Sloan *Sent:* 01 May 2013 03:43 *To:* Josh Petro *Cc:* ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Lab IP Routing ** ** Josh, ** ** For the OSPF question, my current plan is to issue the below as part of a base config on all routers to make sure I'm advertising the local networks and hopefully negate any built-in troubleshooting for the routing piece:* *** ** ** router ospf 1 network 0.0.0.0 area 0 ** ** I'm interested to hear others opinions though. ** ** Marty ** ** ** ** On Tue, Apr 30, 2013 at 10:12 PM, Josh Petro josh.pe...@gmail.com wrote: Hi everyone, I'm sorry for asking this question, but it hit me a little while ago that for the most part, routing via OSPF is setup on most (if not all) of the practice labs. Is that the case in the real lab? I'm trying to make sure I don't need to brush up on my OSPF commands prior to my attempt. I know the previous versions of the lab had more IP routing in them, but I wasn't sure what this version was like. Also, I'm assuming our connection to the lab is via console cable and there is no way to telnet to the gateways, correct? Reason I ask is because we're always talking about speed and whatnot and 9600 baud isn't exactly what you'd want so I've been practicing my command line switches for finding the correct running-config commands. Any info (without breaking NDA) is appreciate. Josh ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab IP Routing
Josh, For the OSPF question, my current plan is to issue the below as part of a base config on all routers to make sure I'm advertising the local networks and hopefully negate any built-in troubleshooting for the routing piece: router ospf 1 network 0.0.0.0 area 0 I'm interested to hear others opinions though. Marty On Tue, Apr 30, 2013 at 10:12 PM, Josh Petro josh.pe...@gmail.com wrote: Hi everyone, I'm sorry for asking this question, but it hit me a little while ago that for the most part, routing via OSPF is setup on most (if not all) of the practice labs. Is that the case in the real lab? I'm trying to make sure I don't need to brush up on my OSPF commands prior to my attempt. I know the previous versions of the lab had more IP routing in them, but I wasn't sure what this version was like. Also, I'm assuming our connection to the lab is via console cable and there is no way to telnet to the gateways, correct? Reason I ask is because we're always talking about speed and whatnot and 9600 baud isn't exactly what you'd want so I've been practicing my command line switches for finding the correct running-config commands. Any info (without breaking NDA) is appreciate. Josh ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUE CLI/GUI ?
I've been wondering the same thing myself. I picked up Kevin Wallace's IE Voice Alchemy (1examamonth.com - highly recommended for developing an exam strategy) and point 6 of his '12 strategies' is to use the CUE CLI instead of the GUI as a time saver. I use the CLI for most things in my work so I'd prefer to go that route. I figure once I get familiar with the syntax of the different config options, it would be faster. Anyone recommend otherwise? On Mon, Apr 29, 2013 at 11:13 AM, Nicolas MICHEL mcl.nico...@gmail.comwrote: Hey Guys. I have currently struggling with CUE integration / installation and configuration. What would you use in the Lab ? CLI or GUi ? Because in the Workbooks, the GUI is used approx all the times ... Just wanted to have your thoughts Thanks for the help Nicolas __**_ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME BACD dial-peer
Just wanted to post the fix that I found for this one in case it helps. Thanks to the OSL archives for this one. Specifying the port after the ip on the dial-peer fixed this. In case both scenarios are presented (non-default H225 listen address with BACD), use something like the below on the AA dial-peer. session target ipv4:10.10.110.3:1820 Thanks for the help On Wed, Apr 24, 2013 at 8:55 AM, Martin Sloan martinsloa...@gmail.comwrote: Awesome, that fixed the problem and many thanks for the help. I had the listen port command in there from a previous lab and was so focused on the dial-peer I was missing other opportunities to fix this. Do you mind commenting on how you knew this was the issue and whether you think I could expect to see both of these requirements in the real lab. If so, is there a workaround? On Tue, Apr 23, 2013 at 11:38 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Please remove port 1820 from VoIP service it will work Sent from my iPhone On Apr 23, 2013, at 9:54 PM, Martin Sloan martinsloa...@gmail.com wrote: Rafael, Thanks for the assist. I've attached the sh run and debug. Marty On Tue, Apr 23, 2013 at 10:10 PM, Rafael Chavantes raf...@chavantes.comwrote: Hello Martin, Can you please send the sh run and the output for debug ccapi inout? On Tuesday, April 23, 2013, Martin Sloan wrote: Hello experts, I'm having some trouble with the BACD dial-peer in vol 1 WB section 9.2. I'm following the CUCME BACD 'tcl in flash mem' guide as recommended by Vik but I'm getting fast busy when dialing into the aa. If I bring up a POTS dial-peer for PSTN-AA or if I modify session target to ras in the voip dial-peer for IP Phone-AA, the aa works so I at least know that part is solid. Here's the dial-peer config I'm using that isn't working: dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Also: allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip And a snip from the end of a voip dialpeer debug: Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3500, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3006, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40001-dial-peer)# I see in the debug it matches 222 as the inbound dial-peer but then it also matches 40001 for the SIP CME phone. Any help is much appreciated. Marty -- Rafael Chavantes BR2_sh_run.txt BACD_ccapi_inout.txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME BACD dial-peer
Awesome, that fixed the problem and many thanks for the help. I had the listen port command in there from a previous lab and was so focused on the dial-peer I was missing other opportunities to fix this. Do you mind commenting on how you knew this was the issue and whether you think I could expect to see both of these requirements in the real lab. If so, is there a workaround? On Tue, Apr 23, 2013 at 11:38 PM, Ramcharan Arya ramcharan.a...@gmail.comwrote: Please remove port 1820 from VoIP service it will work Sent from my iPhone On Apr 23, 2013, at 9:54 PM, Martin Sloan martinsloa...@gmail.com wrote: Rafael, Thanks for the assist. I've attached the sh run and debug. Marty On Tue, Apr 23, 2013 at 10:10 PM, Rafael Chavantes raf...@chavantes.comwrote: Hello Martin, Can you please send the sh run and the output for debug ccapi inout? On Tuesday, April 23, 2013, Martin Sloan wrote: Hello experts, I'm having some trouble with the BACD dial-peer in vol 1 WB section 9.2. I'm following the CUCME BACD 'tcl in flash mem' guide as recommended by Vik but I'm getting fast busy when dialing into the aa. If I bring up a POTS dial-peer for PSTN-AA or if I modify session target to ras in the voip dial-peer for IP Phone-AA, the aa works so I at least know that part is solid. Here's the dial-peer config I'm using that isn't working: dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Also: allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip And a snip from the end of a voip dialpeer debug: Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3500, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3006, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40001-dial-peer)# I see in the debug it matches 222 as the inbound dial-peer but then it also matches 40001 for the SIP CME phone. Any help is much appreciated. Marty -- Rafael Chavantes BR2_sh_run.txt BACD_ccapi_inout.txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME BACD dial-peer
Hi Bill, Thanks for the help. This is on a CME so I'm not following how the CUCM plays into this. Is there something to change within telephony-service on the IOS? Thanks, Marty On Wed, Apr 24, 2013 at 9:16 AM, Bill Lake whl...@gmail.com wrote: If you are required to use a different port you must change the setting in CUCM Sent from my iPhone On Apr 24, 2013, at 7:55 AM, Martin Sloan martinsloa...@gmail.com wrote: Awesome, that fixed the problem and many thanks for the help. I had the listen port command in there from a previous lab and was so focused on the dial-peer I was missing other opportunities to fix this. Do you mind commenting on how you knew this was the issue and whether you think I could expect to see both of these requirements in the real lab. If so, is there a workaround? On Tue, Apr 23, 2013 at 11:38 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Please remove port 1820 from VoIP service it will work Sent from my iPhone On Apr 23, 2013, at 9:54 PM, Martin Sloan martinsloa...@gmail.com wrote: Rafael, Thanks for the assist. I've attached the sh run and debug. Marty On Tue, Apr 23, 2013 at 10:10 PM, Rafael Chavantes raf...@chavantes.comwrote: Hello Martin, Can you please send the sh run and the output for debug ccapi inout? On Tuesday, April 23, 2013, Martin Sloan wrote: Hello experts, I'm having some trouble with the BACD dial-peer in vol 1 WB section 9.2. I'm following the CUCME BACD 'tcl in flash mem' guide as recommended by Vik but I'm getting fast busy when dialing into the aa. If I bring up a POTS dial-peer for PSTN-AA or if I modify session target to ras in the voip dial-peer for IP Phone-AA, the aa works so I at least know that part is solid. Here's the dial-peer config I'm using that isn't working: dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Also: allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip And a snip from the end of a voip dialpeer debug: Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3500, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3006, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40001-dial-peer)# I see in the debug it matches 222 as the inbound dial-peer but then it also matches 40001 for the SIP CME phone. Any help is much appreciated. Marty -- Rafael Chavantes BR2_sh_run.txt BACD_ccapi_inout.txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME BACD dial-peer
Hello experts, I'm having some trouble with the BACD dial-peer in vol 1 WB section 9.2. I'm following the CUCME BACD 'tcl in flash mem' guide as recommended by Vik but I'm getting fast busy when dialing into the aa. If I bring up a POTS dial-peer for PSTN-AA or if I modify session target to ras in the voip dial-peer for IP Phone-AA, the aa works so I at least know that part is solid. Here's the dial-peer config I'm using that isn't working: dial-peer voice 222 voip service aa destination-pattern 3500 session target ipv4:10.10.110.3 incoming called-number 3500 dtmf-relay h245-alphanumeric codec g711ulaw no vad Also: allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip And a snip from the end of a voip dialpeer debug: Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3500, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.603: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=222 Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Calling Number=3006, Called Number=, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH Apr 24 00:52:24.623: //-1//DPM/dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=40001-dial-peer)# I see in the debug it matches 222 as the inbound dial-peer but then it also matches 40001 for the SIP CME phone. Any help is much appreciated. Marty ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
