Re: [OSL | CCIE_Voice] Connected number display
Hi: Imho this won't work, I've tested yesterday and phone display has to be manipulated on rp not rl, I haven't test it at called party xformation level but Daniel's aproach seems to be the only working solution Thanks Date: Mon, 21 Jun 2010 23:58:41 -0700 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Daniel, If you do not adjust the called number display on the route pattern, the called number display settings on the route list will go into effect. Have you tried to manipulate the called number on the route list? Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru On 6/21/2010 11:26 PM, Daniel Berlinski wrote: Manipulation at the route list level does not affect how the dialed number is updated on the phone display. I read this as per below: If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. How would manipulation at the route list help in this scenario? I have just tested here by manipulating the dialed number at the route pattern for the first choice gateway (MGCP BR1 - 7Digits) and by using called party xformation pattern for the second choice gateway (MGCP-BR2) In my case I could not do it for 10 digits because my BR2 router is in Spain. The phone display updates as per both transformation configs. If this is not correct please let me know what I'm missing Cheers On Tue, Jun 22, 2010 at 2:20 PM, Berry, Matthew J. mjbe...@krollontrack.com wrote: Daniel, You best bet would be to do the manipulation at the route list level for such a request. - Sent from my Blackberry From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Angel Perez gorr...@hotmail.com Cc: osl osl ccie_voice@onlinestudylist.com Sent: Mon Jun 21 16:04:44 2010 Subject: Re: [OSL | CCIE_Voice] Connected number display Hello Guys Just an idea and please ignore if this is a silly one or let me know if you have already tested this. Could you try to have your manipulation done at route pattern level for BR1 and for BR2 add a called party xformation in order to update the phone display when BR1 is down? As far as my understanding goes ANI manipulations at route pattern and (DNIS) called party transformation patterns applied to egress gateways will also have the cosmetic effect to phones screens. I will give this a go as soon as I have access to equipment again and will update Best Regards Daniel On Mon, Jun 21, 2010 at 11:13 PM, Angel Perez gorr...@hotmail.com wrote: Yes you are right, tested today, ccm engine will not try with another route pattern although controller/gw associated to the first rp is not up. I thought ccm would follow the same behaviour as a h323 gw. Since the only way I know to change phone display number is through route patt, my conclusion is that your requirements are not possible to be satified... Is this an exercise from a workbook or something you want to test? In case it's the first one let us know the solution becouse I can't think a way to make this work with ucm only. Thanks Date: Sun, 20 Jun 2010 17:28:59 +0530 Subject: Re: [OSL | CCIE_Voice] Connected number display From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com i tested bot the RP first.. then i did a no mgcp command on GW1 On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel
Re: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None
Mike: This have been discussed previously in the list (make a search), although phones config is not showed with srst auto none, you can add ephones if you want, if you add ephone 1 and ephone 2 it will match with your srst phones and the privacy will be off. Try it a let us know thx Date: Mon, 21 Jun 2010 06:47:12 -0400 Subject: Re: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None From: 2xcci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Angel, With none the ephone configurations do not show up in the configuration. How can you add the privacy off command to the ephones, if the ephone configuration is not in the config ? Regards, Mike On Mon, Jun 21, 2010 at 3:40 AM, Angel Perez gorr...@hotmail.com wrote: Hi: For srst mode auto none, just add the following ephone 1 privacy off ephone 2 privacy off hth Date: Sun, 20 Jun 2010 20:53:20 -0400 From: 2xcci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None I have BR1 phones configured with a shared line. During normal operation when phone 1 calls a number (ie 911), phone 2 can see the number of the caller who is connected to phone 1. So privacy is off. This was done by changing the service parameter in callmanager and leaving the phones to default for privacy. The problem is when the phones go into SRST (srst mode auto-provision none) this behavior no longer exists. Its as if privacy is enabled. Neither phone can see who the other phones shared line is connected to. Under telephony-service no privacy is configured. Has anyone ran into this issue before ? Is this a proctorlabs limitation somehow ? I am using 7961s, is that a problem ? I do not know how to fix this without changing to srst mode auto-provision all. Is this a limitation of none ? Your thoughts would be greatly appreciated. Mike Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
Yes you are right, tested today, ccm engine will not try with another route pattern although controller/gw associated to the first rp is not up. I thought ccm would follow the same behaviour as a h323 gw. Since the only way I know to change phone display number is through route patt, my conclusion is that your requirements are not possible to be satified... Is this an exercise from a workbook or something you want to test? In case it's the first one let us know the solution becouse I can't think a way to make this work with ucm only. Thanks Date: Sun, 20 Jun 2010 17:28:59 +0530 Subject: Re: [OSL | CCIE_Voice] Connected number display From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com i tested bot the RP first.. then i did a no mgcp command on GW1 On Sun, Jun 20, 2010 at 4:52 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None
Hi: Just add ephone 1 privacy off ephone 2 privacy off You don't need anything else thx Date: Mon, 21 Jun 2010 07:08:49 -0400 Subject: Re: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None From: 2xcci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Awesome I will give it a try and let you know. Thx Mike On Mon, Jun 21, 2010 at 7:05 AM, Angel Perez gorr...@hotmail.com wrote: Mike: This have been discussed previously in the list (make a search), although phones config is not showed with srst auto none, you can add ephones if you want, if you add ephone 1 and ephone 2 it will match with your srst phones and the privacy will be off. Try it a let us know thx Date: Mon, 21 Jun 2010 06:47:12 -0400 Subject: Re: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None From: 2xcci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Angel, With none the ephone configurations do not show up in the configuration. How can you add the privacy off command to the ephones, if the ephone configuration is not in the config ? Regards, Mike On Mon, Jun 21, 2010 at 3:40 AM, Angel Perez gorr...@hotmail.com wrote: Hi: For srst mode auto none, just add the following ephone 1 privacy off ephone 2 privacy off hth Date: Sun, 20 Jun 2010 20:53:20 -0400 From: 2xcci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Privacy - SRST Mode Auto Provision None I have BR1 phones configured with a shared line. During normal operation when phone 1 calls a number (ie 911), phone 2 can see the number of the caller who is connected to phone 1. So privacy is off. This was done by changing the service parameter in callmanager and leaving the phones to default for privacy. The problem is when the phones go into SRST (srst mode auto-provision none) this behavior no longer exists. Its as if privacy is enabled. Neither phone can see who the other phones shared line is connected to. Under telephony-service no privacy is configured. Has anyone ran into this issue before ? Is this a proctorlabs limitation somehow ? I am using 7961s, is that a problem ? I do not know how to fix this without changing to srst mode auto-provision all. Is this a limitation of none ? Your thoughts would be greatly appreciated. Mike Hotmail: Free, trusted and rich email service. Get it now. Hotmail: Free, trusted and rich email service. Get it now. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP phones for CME
Are you working on your own gear? If so check that your phones have the correct fw hth Date: Mon, 21 Jun 2010 13:19:36 +0100 From: naoufal.kerbo...@cbi.ma To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP phones for CME Hi guys, I'm working on lab9 Vol2, and I have 7961 phones registred to SIP CME, but every time the phones unregistred and registred again. Any Ideas? _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RE : SIP phones for CME
Please paste your config Subject: RE : [OSL | CCIE_Voice] SIP phones for CME Date: Mon, 21 Jun 2010 13:54:33 +0100 From: naoufal.kerbo...@cbi.ma To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com No I'm working on PL vRack Message d'origine De: Angel Perez [mailto:gorr...@hotmail.com] Date: lun. 6/21/2010 12:43 À: naoufal.kerboute; osl osl Objet : RE: [OSL | CCIE_Voice] SIP phones for CME Are you working on your own gear? If so check that your phones have the correct fw hth Date: Mon, 21 Jun 2010 13:19:36 +0100 From: naoufal.kerbo...@cbi.ma To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SIP phones for CME Hi guys, I'm working on lab9 Vol2, and I have 7961 phones registred to SIP CME, but every time the phones unregistred and registred again. Any Ideas? _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969 _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCME Unicast MoH
Hi: Unicast is not permited beetween sccp phones for CME (thanks Amy), so no need for Whireshark :) you can only test uni from pstn thx From: ghopk...@wolf-rock.co.uk Date: Mon, 21 Jun 2010 18:11:54 +0100 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUCME Unicast MoH Section in Vol 2 Lab 9 MoH from the CUCME routers (in my own lab) BR2 - multicast - fine to phones and PSTN BR1 - unicast - fine to phones and PSTN, phones are SIP and prefer G729, so transcoder in use. HQ - unicast - fine to PSTN but not to phones, default ephone seems to have multicast-moh, so have turned that off any ideas before I resort to Wireshark ? ephone 1 no multicast-moh device-security-mode none description HQ Phone1 mac-address 0024.14B3.662C type 7965 button 1:1 HQ-RTR#sh telephony-service ephone Number of Configured ephones 2 (Registered 2) ephone 1 Device Security Mode: Non-Secure mac-address 0024.14B3.662C type 7965 button 1:1 keepalive 30 auxiliary 30 max-calls-per-button 8 busy-trigger-per-button 0 Always send media packets to this router: No Preferred codec: g711ulaw conference drop-mode never conference add-mode all conference admin: No privacy: Yes privacy button: No user-locale US network-locale US Regards Graham Hopkins ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
Hi: Did you test both rp alone first to make sure it working correctly? Did you shutdown controller at br1 before testing backup path? thx Date: Sun, 20 Jun 2010 11:49:27 +0100 From: siddas...@gmail.com To: voip.ccieci...@gmail.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Did you also try what I suggested? masking Called party at RL detail level! cisco voip wrote: I tried this just now. and it is not working, So what i was thinking is correct, it can match only one route pattern and call cannot come back. Is there any other way anyone would think of?? On Sun, Jun 20, 2010 at 12:00 AM, Angel Perez gorr...@hotmail.com wrote: Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Native vlan
Hi: In case that it's not specified, would you set the native vlan? And would you set it for data or for servers vlan in case of hq? Or simply would you let the vlan1 to be the native vlan? Thanks _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Ver.2 Lab 3 Messaging
Hi check Amy reply: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16368.html hth From: engnasse...@hotmail.com To: ccie_voice@onlinestudylist.com Date: Sat, 19 Jun 2010 13:43:23 +0300 Subject: [OSL | CCIE_Voice] Ver.2 Lab 3 Messaging Hello Everyone, I am trying to implement Ver.2 Lab 3 Messaging Part, In Auto Attendant part, I configured the standard greeting in the created call handler approperiatly (having Allow Transfers to Numbers Not Associated with Users or Call Handlers checked), but I am still unable to dial unkown extensions, when I dial a subscriber extension, call is transfered approperiately. Shall be waiting for your help Regards, Mouhammad Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
In case of mgcp and h323 gw as backup one rp would be enough becouse h323 gw calls would display the number as it egress the UCM to h323 gw, so in this case you would set DDI at route pattern to meet mgcp gw phone display requirements, then use rl detail to meet h323 gw phone display requirement plus voice transltion rules at h323 gw to meet pstn requirements wich may be differents to phone display requirements hth Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice #26244
Well done Ash :) Very good job Date: Fri, 18 Jun 2010 19:46:07 +0100 From: siddas...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Voice #26244 Hello all, I went to Brussels yesterday and just an hour before learned that I am now officially CCIE Voice. It was my 2nd attempt but it was worth it. I learned a lot from my first attempt and it helped me build a better strategy for the 2nd. I am thankful to this wonderful list and IPExpert material which I used. Special thanks to Amy Ryan for her help whenever I needed. I am also grateful to my Study Partner Iwan Hoogendoorn, a triple CCIE and I was so lucky to have him as Study partner. I will never forget the way he use to make daily schedules and strictly made me follow those otherwise I am a lazy man..this number is for you Iwan! Few take home points for all those who will be making an attempt in coming days: 1 - Read the lab CAREFULLY (I made it Caps for a reason)..every word in a question is there for a reason! 2 - Do not rush! the mistakes you will make in first one hour will haunt you in the entire lab (unless you are lucky to figure out what went wrong) 3 - Do not spend too much time if something is not working - you can always come back to it. 4 - Note down sections and task which you are working and cross them as soon as you have completed it 5 - Call routing - This is how I did it, not necessarily helpful for you, I did call routing on a page first as what I am going to do at RL level, Pattern level etc..I configured everything first and then tested it one by one..took me 30 minutes to finish call routing 6 - Test everything you have done at least twice and as if it was configured by someone else and you are the proctor..I found one mistake while doing my 2nd check 7 - Save your config often, make sure before you leave that all gateways are up and registered to CUCM. I joined this list for my CCIE studies when I started my CCIE journey back in December 2009 but now I have decided to stick with it as I won't find such a nice bunch of people anywhere.. N.B: Above all, I loved my number..Digit '4' is my lucky number and Cisco made sure that I have enough of them.. :) Thank you all. It's party time now ;) Ashar Siddiqui CCIE#26244 (Voice) _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Native vlan
Thanks for your answers Regards From: salman.shaik...@gmail.com Date: Sat, 19 Jun 2010 10:46:30 -0400 Subject: Re: [OSL | CCIE_Voice] Native vlan To: sc...@meganandscott.com CC: gorr...@hotmail.com; ccie_voice@onlinestudylist.com rite scott, don't look at vlan 1, i just gave example in production perspective view for management purpose and for lower vlan advise coz of when something happened to your network or like power failure or something else.. well network will take a small bit time to re-convergence . so it's better to use for always lower vlan's for those which required to be up ASAP that's why the above advise not for lab point of view... so pls take out vlan 1 and rest of things are fine for me . once again if i am wrong pls correct me that would be really appreciable ... thanks On Sat, Jun 19, 2010 at 10:07 AM, Scott Newberry sc...@meganandscott.com wrote: To be honest, I think I'd just leave vlan 1 as the native. Not because of any specific knowledge of the lab, but because I don't want to do anything I'm not required to do. It takes no time to set the native vlan, but if my mind is moving on to the next task and I type the wrong command or something along those lines, I don't want my doing something extra that wasn't required to cost me troubleshooting time. Same thing with setting allowed vlans on a trunk. If I'm not required to restrict which vlans are allowed on the trunk, they're all getting trunked. If I fat-finger a vlan number... Not that any of that should be hard to troubleshoot, but on test day, I just don't want any extra, self-induced stress. Just my two cents! Scott http://ccie.meganandscott.com Blogging my way to my 8/16/2010 lab exam date On Sat, Jun 19, 2010 at 6:05 AM, Angel Perez gorr...@hotmail.com wrote: Hi: In case that it's not specified, would you set the native vlan? And would you set it for data or for servers vlan in case of hq? Or simply would you let the vlan1 to be the native vlan? Thanks Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Connected number display
Hi Ash, I think that to change calling number at phone display you may do transformation at rp level, correct me if i'm wrong thx Date: Sat, 19 Jun 2010 12:34:08 +0100 From: siddas...@gmail.com To: gorr...@hotmail.com CC: voip.ccieci...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Connected number display Sorry Ignore my last post, I thought you are asking about Calling party number (ANI). The one Angel mentioned is a possible solution or try this one...make one route pattern, Create two RG in the RL, then place mask under Called party like XXX and XX under Route list detail level. I have not tested it so give it a try and let us know how it works. Ash Angel Perez wrote: Hi: The only way I can imagine to make this work is with to different route patterns, instead with one route pattern and a route list with two options, something like this: rp1: 91[2-9]XX.[2-9]XX DDI PREDOT, PT=br1-local-first-option rp2: 91.[2-9]XX[2-9]XX DDI PREDOT, PT=br1-local-sec-option br1 phone 1: css (phones,911,br1-local-first-option, br1-local-sec-option, ld, ...) Becouse rp1 and rp2 are and equal match for UCM call processing engine, the pt orther will be the tie breaker, so the first choice would be rp1, and second choice would be rp2. Let us know how it goes Regards Date: Sat, 19 Jun 2010 16:01:09 +0530 From: voip.ccieci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Connected number display Hi Experts, If I have two MGCP gateways BR1 and BR2, and call should go thru BR1 and if it fails it should go thru BR2. Requirement is if call goes through BR1, called number on my display should be 7 digits. If it goes thru BR2, called number should be 10 digits. From what i understand, display number is the manipulated number in Route Pattern. So I am not really sure how to change the display number on the basis of what gateway call is going out. Any Suggestions? Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab 3 Volume 2 SRST CUE Unknown caller
Hi: from CUE: voicemail callerid hth Date: Thu, 17 Jun 2010 07:24:17 +1200 From: dberlin...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab 3 Volume 2 SRST CUE Unknown caller Hello In the following scenario: Phone 1002 rings 3002 in SRST mode, calls are unanswered and forwarded to CUE. I leave a msg for 3002 and when collecting it the following is played by CUE “from unknown caller”. I see the call is sent to CUE as follows: From: BR1PH2 sip:+16178631...@10.10.202.1 To: sip:3...@10.10.202.2 I would like to configure it so that CUE plays “from 1002” instead. What configuration is required to achieve this? Thanks _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi: I was wondering how can you add privacy/privacy off to the ephone if you are setting srst auto none? The only way I can imagine is changing from srst auto all to auto none once the ephone are configured. Correct me if i'm wrong thanks Date: Mon, 14 Jun 2010 18:06:15 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Angel, Yes I made it work..its been quite few days now.. I just explicitly included privacy off commands under ephones and it worked. There is no need for srst auto prov all and dialpeer hunt 3 etc... hth On Mon, Jun 14, 2010 at 3:20 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you manage to make this work? Finally I got some time to relab it, if you are interested let me know and I'll post my working config thx Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] QOS FRF.12 MLPP
Hi: Just to add something to Matthew's reply, be sure that you set the correct compression method either frame relay (activated by default with auto qos voip trust in links with 768k bandwith or less) or class based (compress header ip rtp at desired class) . You can't have both at the same time hth Date: Tue, 15 Jun 2010 05:28:59 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] QOS FRF.12 MLPP Kobel, In my opinion, you should only retain the frame-relay ip rtp header-compression under the frame-relay DLCI if you are asked to compress the rtp packets. Because we're dealing with a slow-speed link, auto qos tries to be helpful by adding in this command. My general stance when it comes to answering the QoS lab questions is to only configure what they ask you to setup. Using auto qos is helpful to rough-in a configuration, but leaving in unnecessary elements does not demonstrate a mastery of the knowledge you are being tested on. I will provide another example: When you type auto qos voip several classes will be created. One of those classes, called something like remark, will set DSCP values on so-called rogue traffic masquerading as media or signaling traffic. If the question does not ask you to perform that task, you'll want to remove the remark class. I'm not sure if this helps, but it's my take on the subject. My guess is that the lab would be specific whether they wanted class-based cRTP or not. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 4:23 AM, kobel wrote: Also, after the Auto QOS generates a lot of classes etc. We do edit few things here and there. Just wanted to confirm that is it a good practice to remove rtp header compression? I use to remove it always but now I am getting conflicting feedback that should we remove it or not? interface Serial0/2/0.1 point-to-point bandwidth 256 frame-relay interface-dlci 301 CISCO class AutoQoS-FR-Se0/2/0-301 auto qos voip trust frame-relay ip rtp header-compression I would appreciate any input in this regard. you can configure cRTP in two ways. if the task doesn't explicitly ask for CB cRTP, I keep auto qos config - why waste time? I'm not aware of any drawback of this method. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi: These are my observations for srst and cbarge First of all you will need a cnf bridge configured, the best way is adding srst ip add as a third option in sccp ccm group, once your cnf bridge is registered to srst router (it take some more times than phones) you will need a dn octo line (recomended) configured as conference ad-hoc 1: srst auto all: Then once ephones are registered to srst these combinations worked for me: == telephony-service privacy ! (default) ephone 1 no privacy privacy-button ! from the button disable or enable it == telephony-service privacy / no privacy ! you can also manage from here ephone 1 no privacy privacy-button ! === telephony-service privacy ! (default) ephone 1 privacy-button privacy on / privacy off ! enable and disable from ephone === If you enable/disable localy you can't enable/disable globaly ephone 1 privacy on/off ! enable/disable privacy telephony-service privacy/no privacy! this won't enable/disable privacy becouse you have enable/disable it localy at ephone 2: srst auto none: follow vc approach described above hth Date: Tue, 15 Jun 2010 06:09:52 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Angel - I think you are right. The only way I can see of configuring privacy on/off would be through the ephone section itself. Privacy isn't an option with an ephone-template, otherwise you could have set it there. You could possibly set no privacy under telephony-service, but that would be a global setting. I am not at my lab right now so I cannot verify if that would actually propagate down to SRST-provisioned phones. Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 On 6/15/2010 3:37 AM, Angel Perez wrote: Hi: I was wondering how can you add privacy/privacy off to the ephone if you are setting srst auto none? The only way I can imagine is changing from srst auto all to auto none once the ephone are configured. Correct me if i'm wrong thanks Date: Mon, 14 Jun 2010 18:06:15 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Angel, Yes I made it work..its been quite few days now.. I just explicitly included privacy off commands under ephones and it worked. There is no need for srst auto prov all and dialpeer hunt 3 etc... hth On Mon, Jun 14, 2010 at 3:20 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Did you manage to make this work? Finally I got some time to relab it, if you are interested let me know and I'll post my working config thx Hotmail: Free, trusted and rich email service. Get it now. Hotmail: Trusted email with powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Configuring H.323 Call Preserve
For h323 call preservation adding voice service voip h323 call-preserve And: Allow Peer to Preserve H.323 Call at ucm call manager service param advanced would be enough hth Date: Tue, 15 Jun 2010 06:45:33 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com; ciscovoiceg...@gmail.com Subject: [OSL | CCIE_Voice] Configuring H.323 Call Preserve When configuring call preservation for an H.323 gateway, I am using the following command: voice service voip h323 call-preserve As soon as I hit ENTER, the IOS spits back this warning/notice to me: Warning: Configuring media inactivity detection to avoid hung calls is highly recommended. Does anyone know what I need to do in order to configure media inactivity detection? I want to make sure that I am entering the proper commands to ensure that H.323 call preservation is enabled. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] OSPF Error CUE Module
Hi: This is becouse you are setting ip unnumbered, there is another method with ip address, with it you won't get this error But the error it's just cosmetic hth Date: Tue, 15 Jun 2010 06:48:58 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com; ciscovoiceg...@gmail.com Subject: [OSL | CCIE_Voice] OSPF Error CUE Module I am getting an odd OSPF error after having configured my service-engine for the CUE module: Jun 14 05:46:22.401: %OSPF-4-NO_IPADDRESS_ON_INT: No IP address for interface Service-Engine0/0 Everything appeared to function properly even with this error being reported. Below is my example config that I use to configure the CUE module's IP and connectivity: interface FastEthernet 0/0.101 ip address X.X.X.X 255.255.255.0 interface Service-Engine 0/0 ip unnumered FastEthernet 0/0.101 service-module ip address X.X.X.X 255.255.255.0 service-module ip default-gateway Y.Y.Y.Y no shut ip route X.X.X.X 255.255.255.255 Service-Engine 0/0 -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] service redundancy
Hi: There are certain services: em, ipma, ac, axl or even dhcp and tftp that you can activate at pub or sub. If it is not specified you can doubt if you may activate it at pub, sub or both, my question is what do you think is the best practice to use pub or sub, or it is the same becouse it's not specified. For example if you have to add em service for phones, should you add two services one for each server, just pub or just sub? Thanks in advance _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUE caller id
Hi: I was trying to setup CUE to say voicemail user name instead of phone number when somebody left a message at voicemail, (like in CUC) but the most i can do is just to hear phone number (voicemail callerid), after some tests my conclusions is that it is not possible Anybody has tried this? Regards _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] service redundancy
Hi: Are you sure? I'm logged right know to UCM cluster and I can activate the service at both pub and sub... Anyway for ipma example if redundancy is not required, would you use pub or sub when adding the service url... that is the big question thanks Date: Tue, 15 Jun 2010 13:21:22 +0100 From: naoufal.kerbo...@cbi.ma To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] service redundancy CUCM don't provide redundancy for EM. For IPMA you can activate the service on sub or on pup also both if u want redundoncy On 06/15/2010 12:59 PM, Angel Perez wrote: Hi: There are certain services: em, ipma, ac, axl or even dhcp and tftp that you can activate at pub or sub. If it is not specified you can doubt if you may activate it at pub, sub or both, my question is what do you think is the best practice to use pub or sub, or it is the same becouse it's not specified. For example if you have to add em service for phones, should you add two services one for each server, just pub or just sub? Thanks in advance Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] service redundancy
Hi: You can't configure redundancy like with tftp but you can configure two services one with pub ip address and other one with sub ip address, this way if pub is down you can let the user to activate em from sub service thanks From: pav.c...@gmail.com To: gorr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] service redundancy Date: Tue, 15 Jun 2010 08:15:11 -0500 CC: naoufal.kerbo...@cbi.ma; ccie_voice@onlinestudylist.com I dont think There is a way to configure redundancy for em. You can activate on pub/ sub but only use one of tgem. Let me know if i am mistaken. Sent from my phone On Jun 15, 2010, at 7:26 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Are you sure? I'm logged right know to UCM cluster and I can activate the service at both pub and sub... Anyway for ipma example if redundancy is not required, would you use pub or sub when adding the service url... that is the big question thanks Date: Tue, 15 Jun 2010 13:21:22 +0100 From: naoufal.kerbo...@cbi.ma To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] service redundancy CUCM don't provide redundancy for EM. For IPMA you can activate the service on sub or on pup also both if u want redundoncy On 06/15/2010 12:59 PM, Angel Perez wrote: Hi: There are certain services: em, ipma, ac, axl or even dhcp and tftp that you can activate at pub or sub. If it is not specified you can doubt if you may activate it at pub, sub or both, my question is what do you think is the best practice to use pub or sub, or it is the same becouse it's not specified. For example if you have to add em service for phones, should you add two services one for each server, just pub or just sub? Thanks in advance Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Calling Name
I guess that you won't forget this one :) Date: Tue, 15 Jun 2010 12:56:46 -0400 From: daniyal.vo...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME Calling Name Hi could some one pls help to resolve this issue in CME i don't want send the Calling Name on specific dial-peer but Number suppose to go under D channel i have configured Isdn out display ie that affecting on all calls but requirement is that i just want to block or restrict one person/dial-peer to don't show the calling Name comments/advise appreciated Dani _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi: Did you manage to make this work? Finally I got some time to relab it, if you are interested let me know and I'll post my working config thx _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME CUCM call hold problems
Hi Daniel: Let my check it during the day thx Date: Fri, 11 Jun 2010 18:35:14 +1200 Subject: Re: [OSL | CCIE_Voice] CME CUCM call hold problems From: dberlin...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Angel Thanks a lot for this it has worked by configuring IOS MTP. May I ask you if call transfers worked fine for you as well? In my setup call transfers are not working properly. If for instance I send a call from a CME phone to a CUCM phone then press transfer, the CME phone remains on hold after call is completed with the transfer-to party. The only way to complete transfer is by pressing hold twice on the CME phone. Anyone got call transfers to work perfectly? Same behaviour seen with Supervised or Blind xfer. My CME configs as follows: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 This was added in an attempt to get call xfers to work flawlessly h323 emptycapability This was added in an attempt to get call xfers to work flawlessly h225 id-passthru This was added in an attempt to get call xfers to work flawlessly h225 connect-passthru This was added in an attempt to get call xfers to work flawlessly no call service stop h245 passthru tcsnonstd-passthru This was added in an attempt to get call xfers to work flawlessly sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 br2-xcoder no auto-reg-ephone load 7960-7940 P00308000500 load 7965 SCCP45.8-3-3S max-ephones 3 max-dn 6 no-reg ip source-address 10.10.110.3 port 2000 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Jun 11 2010 08:11:35 ! sccp local Vlan400 sccp ccm 10.10.110.3 identifier 1 version 5.0.1 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register br2-xcoder signaling dscp af31 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 19 associate application SCCP Cheers On Thu, Jun 10, 2010 at 8:12 PM, Angel Perez gorr...@hotmail.com wrote: Hi: You need software mtp from ios not from ucm, make sure that ios mtp are configured and registered, to be sure that mtp is working verify it with sh sccp or from ucm. Once you have ios mtp registered add a mrg and include all ucm software mtp and cnf, then do not include this mrg to any mrgl, this way you will be sure that this resources are not available for your trunk/phones. Also be sure that in the trunk/phones mrgl the ios mtp rosource is above other ucm software resources. Then place a call, press hold and verify with sh sccp con For more information check: CUCM 7 SRND page 5-11 (H.323 Trunks with Media Termination Points ) http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15970.html hth Date: Thu, 10 Jun 2010 19:17:48 +1200 From: dberlin...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME CUCM call hold problems Hello all After completing lab 2 of volume 2 Gatekeeper section I found the following behaviour when testing call hold between phones registered to CUCM and CME respectively: By saying successful I mean the ability to place call on hold and resume Calls from CUCM phones bound for CME phones placed on hold by either CUCM or CME phones are successful Calls from CME phones bound for CUCM phones placed on hold by CUCM phones are not successful. The problem manifestates as not allowing me to resume the call. Same scenario but pushing hold from a CME phone is successful. With this scenario in mind the following was done: MTP required checkbox in trunk is checked and added to MRL of trunk's device pool and the trunk page itself, software MTPs and Hardware IOS xcoders While testing with these Media Resources configured show perf query class counters were not incrementing at all when I pushed hold on the CUCM phone - I was expecting to see MTP usage once pushing the hold.button - Am I right to expect it to happen? show sccp connections did not show anything either as I thought that the xcoder was being used instead. In addition, wait for TCS on trunk were unchecked and outbound faststart was also configured as last resort to see if any difference could be seen in behaviour. rebooting servers did not help either. Anyone experienced this? Cheers Hotmail: Powerful Free email with security by Microsoft. Get it now. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail
Re: [OSL | CCIE_Voice] CME CUCM call hold problems
Hi again: I'm not sure of your exact topolgy, are using cube with gatekeepers? or just two gateways (ucme, ucm) registered to gw? thx From: gorr...@hotmail.com To: dberlin...@gmail.com Date: Fri, 11 Jun 2010 08:52:59 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME CUCM call hold problems Hi Daniel: Let my check it during the day thx Date: Fri, 11 Jun 2010 18:35:14 +1200 Subject: Re: [OSL | CCIE_Voice] CME CUCM call hold problems From: dberlin...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hello Angel Thanks a lot for this it has worked by configuring IOS MTP. May I ask you if call transfers worked fine for you as well? In my setup call transfers are not working properly. If for instance I send a call from a CME phone to a CUCM phone then press transfer, the CME phone remains on hold after call is completed with the transfer-to party. The only way to complete transfer is by pressing hold twice on the CME phone. Anyone got call transfers to work perfectly? Same behaviour seen with Supervised or Blind xfer. My CME configs as follows: voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip supplementary-service h450.12 This was added in an attempt to get call xfers to work flawlessly h323 emptycapability This was added in an attempt to get call xfers to work flawlessly h225 id-passthru This was added in an attempt to get call xfers to work flawlessly h225 connect-passthru This was added in an attempt to get call xfers to work flawlessly no call service stop h245 passthru tcsnonstd-passthru This was added in an attempt to get call xfers to work flawlessly sip bind control source-interface Vlan400 bind media source-interface Vlan400 registrar server ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 3 sdspfarm tag 1 br2-xcoder no auto-reg-ephone load 7960-7940 P00308000500 load 7965 SCCP45.8-3-3S max-ephones 3 max-dn 6 no-reg ip source-address 10.10.110.3 port 2000 time-format 24 date-format dd-mm-yy max-conferences 8 gain -6 call-forward pattern .T transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Jun 11 2010 08:11:35 ! sccp local Vlan400 sccp ccm 10.10.110.3 identifier 1 version 5.0.1 sccp ip precedence 3 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register br2-xcoder signaling dscp af31 ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 19 associate application SCCP Cheers On Thu, Jun 10, 2010 at 8:12 PM, Angel Perez gorr...@hotmail.com wrote: Hi: You need software mtp from ios not from ucm, make sure that ios mtp are configured and registered, to be sure that mtp is working verify it with sh sccp or from ucm. Once you have ios mtp registered add a mrg and include all ucm software mtp and cnf, then do not include this mrg to any mrgl, this way you will be sure that this resources are not available for your trunk/phones. Also be sure that in the trunk/phones mrgl the ios mtp rosource is above other ucm software resources. Then place a call, press hold and verify with sh sccp con For more information check: CUCM 7 SRND page 5-11 (H.323 Trunks with Media Termination Points ) http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15970.html hth Date: Thu, 10 Jun 2010 19:17:48 +1200 From: dberlin...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME CUCM call hold problems Hello all After completing lab 2 of volume 2 Gatekeeper section I found the following behaviour when testing call hold between phones registered to CUCM and CME respectively: By saying successful I mean the ability to place call on hold and resume Calls from CUCM phones bound for CME phones placed on hold by either CUCM or CME phones are successful Calls from CME phones bound for CUCM phones placed on hold by CUCM phones are not successful. The problem manifestates as not allowing me to resume the call. Same scenario but pushing hold from a CME phone is successful. With this scenario in mind the following was done: MTP required checkbox in trunk is checked and added to MRL of trunk's device pool and the trunk page itself, software MTPs and Hardware IOS xcoders While testing with these Media Resources configured show perf query class counters were not incrementing at all when I pushed hold on the CUCM phone - I was expecting to see MTP usage once pushing the hold.button - Am I right to expect it to happen? show sccp connections did not show anything either as I thought that the xcoder was being used instead. In addition, wait for TCS on trunk were unchecked and outbound faststart was also configured as last resort to see if any difference could be seen
Re: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199)
Very very good job Roger :) From: roger.kallb...@cygate.se To: ccie_voice@onlinestudylist.com Date: Fri, 11 Jun 2010 14:13:23 +0200 Subject: [OSL | CCIE_Voice] I passed CCIE Voice (# 26199) I took my lab yesterday, first attempt, just got the score report. I passed :-) I will write down my strategy once I have landed from the cloud that I'm currently flying on. :-D Roger Källberg Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VPIM problem
Hi: I've the following DNS configuration: cue- cue.cisco.lab cuc- cuc.cisco.lab (ip address 150.200.30.13) Both cue and cuc have properly dns address and domain configured Cue config: network location id 440 email domain cue.cisco.lab name cue end location network location id 330 email domain cuc.cisco.lab name cuc end location network local location id 440 Cuc config: -smtp server addres cuc.cisco.lab -vpim location added for cue (440) -converstion manager reloaded -smtp server reloaded -remote users added (to dial by number to remote ext) -If I check cuc user I can see: hq2 @ cuc.cisco.lab (extension 6002) -Also I reloaded the box At this point I can send vpim messages from cue to cuc but when I try to send it in the opposite direction (cuc to cue) I get this error on cue: cue# show trace buffer tail Press CTRL-C to exit... 4402 06/11 20:14:05.584 netw smtp 2 4402 06/11 20:14:05.601 netw smtp 3 socket hostName: 150.200.30.13, hostAddress: 150.200.30.13 4402 06/11 20:14:05.601 netw smtp 3 hostname: 150.200.30.13 found in good address cache 4402 06/11 20:14:05.603 netw smtp 1 10444 06/11 20:14:05.604 netw smtp 5 Initial connection message 10444 06/11 20:14:05.631 netw smtp 6 UNKNOWN: EHLO cuc 10444 06/11 20:14:05.632 netw smtp 5 250-cue 10444 06/11 20:14:05.665 netw smtp 6 EHLO : MAIL FROM:6002 @ cisco.lab 10444 06/11 20:14:05.675 netw smtp 5 554 5.1.8 Bad senders system address 10444 06/11 20:14:05.697 netw smtp 6 MAIL FROM:: QUIT 10444 06/11 20:14:05.698 netw smtp 5 221 closing channel Although everything looks like it is configured correctly on CUC the smtp address I'm reciving at CUE is @ cisco.lab instead of @ cuc.cisco.lab, so CUE is rejecting the message This looks like a limatition/problem of cuc smtp server to send the full domain name to CUE the only workaround i have found to make this work with this dns configuration is adding the following at CUE side: network location id 666 email domain cisco.lab name fake end location This way messages are accepted and working in both directions Any idea would be apreciated Thanks _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VPIM problem
Thanks the reboot made the trick regards From: pav.c...@gmail.com To: gorr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] VPIM problem Date: Fri, 11 Jun 2010 08:19:59 -0500 CC: ccie_voice@onlinestudylist.com I ve had the exact same problem couple of days back. You can fix it by changing the smtp domain name in ucon and rebooting it. Sent from my phone On Jun 11, 2010, at 7:49 AM, Angel Perez gorr...@hotmail.com wrote: Hi: I've the following DNS configuration: cue- cue.cisco.lab cuc- cuc.cisco.lab (ip address 150.200.30.13) Both cue and cuc have properly dns address and domain configured Cue config: network location id 440 email domain cue.cisco.lab name cue end location network location id 330 email domain cuc.cisco.lab name cuc end location network local location id 440 Cuc config: -smtp server addres cuc.cisco.lab -vpim location added for cue (440) -converstion manager reloaded -smtp server reloaded -remote users added (to dial by number to remote ext) -If I check cuc user I can see: hq2 @ cuc.cisco.lab (extension 6002) -Also I reloaded the box At this point I can send vpim messages from cue to cuc but when I try to send it in the opposite direction (cuc to cue) I get this error on cue: cue# show trace buffer tail Press CTRL-C to exit... 4402 06/11 20:14:05.584 netw smtp 2 4402 06/11 20:14:05.601 netw smtp 3 socket hostName: 150.200.30.13, hostAddress: 150.200.30.13 4402 06/11 20:14:05.601 netw smtp 3 hostname: 150.200.30.13 found in good address cache 4402 06/11 20:14:05.603 netw smtp 1 10444 06/11 20:14:05.604 netw smtp 5 Initial connection message 10444 06/11 20:14:05.631 netw smtp 6 UNKNOWN: EHLO cuc 10444 06/11 20:14:05.632 netw smtp 5 250-cue 10444 06/11 20:14:05.665 netw smtp 6 EHLO : MAIL FROM:6002 @ cisco.lab 10444 06/11 20:14:05.675 netw smtp 5 554 5.1.8 Bad senders system address 10444 06/11 20:14:05.697 netw smtp 6 MAIL FROM:: QUIT 10444 06/11 20:14:05.698 netw smtp 5 221 closing channel Although everything looks like it is configured correctly on CUC the smtp address I'm reciving at CUE is @ cisco.lab instead of @ cuc.cisco.lab, so CUE is rejecting the message This looks like a limatition/problem of cuc smtp server to send the full domain name to CUE the only workaround i have found to make this work with this dns configuration is adding the following at CUE side: network location id 666 email domain cisco.lab name fake end location This way messages are accepted and working in both directions Any idea would be apreciated Thanks Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPC and presence status
Hi: No, to view presence status of your contacts: Add ippm service at ucm Subscribe to desired phones From phone access ippm service and finally add contacts from there (you will see the option in the menu) Or better integrate with ad, search from upc and double click on the contac :) thx Date: Fri, 11 Jun 2010 10:34:31 -0500 From: pav.c...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUPC and presence status CUPC installed and working. It is not integrated into AD. I can view status between two CUPC users (i.e status of user1 in CUPC2 and vice versa If i create my own contacts (Local contacts) on CUPC, should i be able to view their presence status ? Subscribe CSS on SIP trunk has been set appropriately. -- - Pavan _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME CUCM call hold problems
Hi: You need software mtp from ios not from ucm, make sure that ios mtp are configured and registered, to be sure that mtp is working verify it with sh sccp or from ucm. Once you have ios mtp registered add a mrg and include all ucm software mtp and cnf, then do not include this mrg to any mrgl, this way you will be sure that this resources are not available for your trunk/phones. Also be sure that in the trunk/phones mrgl the ios mtp rosource is above other ucm software resources. Then place a call, press hold and verify with sh sccp con For more information check: CUCM 7 SRND page 5-11 (H.323 Trunks with Media Termination Points ) http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15970.html hth Date: Thu, 10 Jun 2010 19:17:48 +1200 From: dberlin...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME CUCM call hold problems Hello all After completing lab 2 of volume 2 Gatekeeper section I found the following behaviour when testing call hold between phones registered to CUCM and CME respectively: By saying successful I mean the ability to place call on hold and resume Calls from CUCM phones bound for CME phones placed on hold by either CUCM or CME phones are successful Calls from CME phones bound for CUCM phones placed on hold by CUCM phones are not successful. The problem manifestates as not allowing me to resume the call. Same scenario but pushing hold from a CME phone is successful. With this scenario in mind the following was done: MTP required checkbox in trunk is checked and added to MRL of trunk's device pool and the trunk page itself, software MTPs and Hardware IOS xcoders While testing with these Media Resources configured show perf query class counters were not incrementing at all when I pushed hold on the CUCM phone - I was expecting to see MTP usage once pushing the hold.button - Am I right to expect it to happen? show sccp connections did not show anything either as I thought that the xcoder was being used instead. In addition, wait for TCS on trunk were unchecked and outbound faststart was also configured as last resort to see if any difference could be seen in behaviour. rebooting servers did not help either. Anyone experienced this? Cheers _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
Hi: Wich vmware version do you have installed? I'm working with esxi and I've never seen this Date: Thu, 10 Jun 2010 09:28:32 +0200 From: findko...@gmail.com To: wolfsru...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue I would expect this - it's not a rocket science. Possibly another issue thanks to VMware - I was wondering if anyone seen this also. thanks, kobel On Thu, Jun 10, 2010 at 4:52 AM, wolfsrudel wolfsru...@gmail.com wrote: i've tested this today and works fine, all call are first delivered to the first agent. On Wed, Jun 9, 2010 at 5:32 PM, wolfsrudel wolfsru...@gmail.com wrote: easiest would be routing by skill (most skilled). if one of the agents has a higher weight (on that skill, not the weight attribute) then any call should always be delivered to the same agent always, no matter what. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue
Same here... Some weird things that can cause issues similar to yours: did you check that ccx was in-service state? did you check dbreplication at ucm? - utils dbreplication repair all did you try to resynch jtapi? did you reset axl/cti services at ucm? did you installed ccx from cisco win 2003 or standard win 2003? Try it again with fresh virtual machines images and let us know It's true that sometimes i've problems with ccx image (service down, integration issues) but I've never seen this problem... hth Date: Thu, 10 Jun 2010 11:08:42 +0200 Subject: Re: [OSL | CCIE_Voice] UCCX CSQ hunting order issue From: findko...@gmail.com To: gorr...@hotmail.com CC: wolfsru...@gmail.com; ccie_voice@onlinestudylist.com VMWare ESXi 4.0.0 UCCX 7.0(1)_Build168 On Thu, Jun 10, 2010 at 10:33 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Wich vmware version do you have installed? I'm working with esxi and I've never seen this _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Better Voice Lab Locations
The better one is the one in wich you pass :) From: engnasse...@hotmail.com To: lakpr...@gmail.com; cci...@gmail.com Date: Thu, 10 Jun 2010 15:33:48 +0300 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations Yes sure, Now Dubai is available Regards, Mouhammad Date: Thu, 10 Jun 2010 18:01:22 +0530 Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations From: lakpr...@gmail.com To: cci...@gmail.com CC: amccar...@cciequest.com; ccie_voice@onlinestudylist.com; engnasse...@hotmail.com Guys Dubai also have one right ??? On Thu, Jun 10, 2010 at 5:50 PM, ccie voice cci...@gmail.com wrote: @Amp So you choose a lab location based on lunch? On Thu, Jun 10, 2010 at 1:14 PM, Amp amccar...@cciequest.com wrote: I live here in the RTP area but have decided to take the lab in San Jose. Here are my reasons: 1. Later Start Time 2. Longer Lunch 3. Better Weather 4. Just have a gut feeling about SJC Amp Quoting Jeff Garvas j...@cia.net: I heard that the West coast facility starts later, so someone east of that location would gain the time zone benefits as well as the late start. RTP supposedly starts first thing in the morning bright and early. 2010/6/9 Mouhammad Nasser engnasse...@hotmail.com Hi, I think it is better to take one that is closest to one's timezone! this will eliminate the factor of travel sickness, and one may go to exam awake enough! Regards, -- Hotmail: Trusted email with powerful SPAM protection. Sign up now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Ravindra Lakpriya +94 773 532 094 Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] FRTS and MLP over a Serial with Sub-Interfaces
Hi: Your mixing traffic shapping with link fragmentation and interleaving The first one (frame relay traffic shapping or cb traffic shapping) limits the amount of bandwith that you send to a link for example to avoid service provider policing to this traffic. The second one is a link efficiency tool (both frf 12 or mlp) permit bigger packects to be fragmented and be interleaved with other small (probably rtp) packets, to avoid the impact of serilization delay, both of them works with frame relay traffic shapping Date: Thu, 10 Jun 2010 09:35:56 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] FRTS and MLP over a Serial with Sub-Interfaces Quick question. In the lab, if the HQ site is setup with two sub-interfaces that connect to BR1 and BR2 (i.e. meaning, they're both running off the same interface), how would you configure MLP for one site and FRF.12 for another site? According to my understanding, MLP will require that frame-relay traffic-shaping is enabled on the serial interface. However, this would botch up your FRF.12 configuration on the other sub-interface. QoS is a weak area for me so I might be missing something obvious in this question. However, it came up so I thought I would ask. Thanks -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] BACD prompts
Hi all: Anybody knows if .wav files recorded with CUE prompt manager (aka TUI) are valid for bacd tcl scripts? BACD prompts are .au files but it think that .wav are also valid, anybody can clarify this? Thanks _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] clock summer-time
Hi: In real live thats depend on the timezone, for US time zones (PDT, EDT, ...) is not necessary becouse the default has the correct date, but for example at Europe summer time start at different week depending on the zone so you should manually configure. In the lab I suppose that you should ask proctor hth From: siddas...@gmail.com To: ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.com Date: Wed, 9 Jun 2010 16:36:00 +0100 Subject: Re: [OSL | CCIE_Voice] clock summer-time I have never done start/stop and it use to work fine. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Matthew Berry Sent: 09 June 2010 16:04 To: OSL Group Subject: [OSL | CCIE_Voice] clock summer-time Is it necessary to define a start/stop for the clock summer-time recurring command? I have been entering this as a general practice for all my exercises. However, I'm not sure if it's required to enter a start/stop time. Comments? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule
Hi: I tested it some time ago an it didn't works... so I needed to use voice translation... I think that other people had problems with this also Give it a try a let us know hth Date: Tue, 8 Jun 2010 20:36:49 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Setting number plan indicator on the dial peer without a translation rule Reading the Implementing Cisco Voice Gateways and Gatekeepers student guide, page 290. They cite another way to set numbering plan on a dial peer. Here is their example: dial-peer voice 100 pots numbering-type national destination-pattern 91408... prefix 1408 port 1/0:23 Has anyone tried this before? This might be a way to avoid (if needed) setting the type via a translation-rule/profile. Thoughts? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN-WAN Router Connectivity Question
Hi: For pstn and wan connectivity you dont need any connection to sw. But if you are planning to use pstn gw as a remote gk, ntp, etc you can use one fast eth to connect to hq sw at servers vlan. Then at the other fast eth port on your pstn gw you can plug pstn phone directly, configure this port as trunk and add the voice vlan manually to the phone from phone menu hth Date: Wed, 9 Jun 2010 16:46:24 +0100 From: clare.turnbullal...@googlemail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN-WAN Router Connectivity Question Hi All, I am in the process of building my home lab and have a newbie question regarding the PSTN / WAN router. Can someone please confirm what the PSTN / WAN router’s Fast 0/0 interface connects to? I have checked some old config’s from the previous lab’s and it seems that it used to connect to the 6500, so does it now connect to the HQ’s 3750 for it's OSPF broadcast to work? If this is the case, do you also connect the 7960 PSTN phone to HQ’s 3750 and and hardcode to see the PSTN's CME. Thanks Clare _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] PSTN-WAN Router Connectivity Question
Hi, I forgot to say that you will need a cross cable to plug your pstn phone to the second fas eth port of your pstn gw cheers Date: Wed, 9 Jun 2010 19:12:55 +0100 Subject: Re: [OSL | CCIE_Voice] PSTN-WAN Router Connectivity Question From: clare.turnbullal...@googlemail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Thanks Angel, Thats exactly what I was after. Clare On Wed, Jun 9, 2010 at 6:37 PM, Angel Perez gorr...@hotmail.com wrote: Hi: For pstn and wan connectivity you dont need any connection to sw. But if you are planning to use pstn gw as a remote gk, ntp, etc you can use one fast eth to connect to hq sw at servers vlan. Then at the other fast eth port on your pstn gw you can plug pstn phone directly, configure this port as trunk and add the voice vlan manually to the phone from phone menu hth Date: Wed, 9 Jun 2010 16:46:24 +0100 From: clare.turnbullal...@googlemail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] PSTN-WAN Router Connectivity Question Hi All, I am in the process of building my home lab and have a newbie question regarding the PSTN / WAN router. Can someone please confirm what the PSTN / WAN router’s Fast 0/0 interface connects to? I have checked some old config’s from the previous lab’s and it seems that it used to connect to the 6500, so does it now connect to the HQ’s 3750 for it's OSPF broadcast to work? If this is the case, do you also connect the 7960 PSTN phone to HQ’s 3750 and and hardcode to see the PSTN's CME. Thanks Clare Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi cv: I didn't have time yesterday,sorry I'll try asap thx Date: Mon, 7 Jun 2010 23:07:57 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com; ar...@ipexpert.com CC: ccie_voice@onlinestudylist.com OK feedback time I tried doing everything which Amy and Angel suggested and still it did not work! I have this under my telephony now: telephony-service sdspfarm units 2 sdspfarm tag 1 sbconf no privacy conference hardware srst mode auto-provision all srst ephone template 1 srst dn template 1 srst dn line-mode dual-octo max-ephones 5 max-dn 10 preference 5 ip source-address 10.10.201.1 port 2000 max-conferences 8 gain -6 moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route 10.10.110.2 10.10.201.1 transfer-system full-consult secondary-dialtone 9 create cnf-files version-stamp 7960 Jun 07 2010 16:28:17 ephone-1[0] Mac:0017.9402.CF34 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:11 privacy:0 IP:192.168.10.15 31806 7961 keepalive 17 max_line 6 button 1: dn 5 number 4001 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw ephone-3[2] Mac:0018.195A.B173 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:11 privacy:0 IP:192.168.10.13 31052 7961 keepalive 14 max_line 6 button 1: dn 8 number 4003 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared privacy button is enabled Preferred Codec: g711ulaw the dial-peer hunt 3 has been configured. Privacy button under ephones configured. Now when I press the shared line button to Barge in..I sometimes get a glimpse of Cbarge Newcall Softkey and then within a fraction of a second it disappears with the new call softkey template and I get a dial tone. Router has been rebooted as well. did you give it a go Angel? cv On Mon, Jun 7, 2010 at 10:37 AM, Angel Perez gorr...@hotmail.com wrote: Hi, you are partially right: When your phones registered back to ucm the e-phone stay there but dial-peer generated by cme srst doesn't point to anything (you can varify this with show dial-peer voice summ). But it's true that sometimes (it think that it is a bug) the ephone dial-peer still points to the dn number so once that your phones register back to ucm you can have problems with incomming calls With mgcp you don't have this problem becouse the gw is totally under ucm control so an incoming call won't match dial-peers , with h323 gw is different and ephone dial-peer will be matched, but you can easly change this behaviour like this: telephony service ... srst mode auto all max-dn 10 preference 5 ... exit dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 dial-peer voice 1000 voip description toUCM destination-pattern 1... preference 1 ... This way when a call enter the gw it will match the preference first instead longest match and calls will still work in srst mode and out srst mode. Try it and play a littel bit with it, you will find that is not difficult ps: also there is another bug with cme srst auto all, if you modify the auto generated ephones or ephones dn (let say name or label) sometimes when you go back to ucm and then to srts again the ephone won't take a dn, it will register to srst but the phone display won't show a line, in this case reload the router and everithing will start working as expected (this have been discused on the list) As you see cme srst is not pretty stable but with this two trick you can make it works easly if you have littel experience with it hth Date: Mon, 7 Jun 2010 10:16:47 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From
Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
: arqp=0x49E870B4,crv=0x8059, answerCall=1 Jun 7 22:07:34.192: //E801AC8F8482/E80248B78484/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:35.268: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1# = Here is debug for Failed call from 4001 to 2001 HQ-R1# Jun 7 22:07:44.888: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:44.892: ////GK/gk_rassrv_arq: arqp=0x49EB9F60,crv=0x5B, answerCall=0 Jun 7 22:07:44.892: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/gk_dns_query: No Name servers Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_get_addrinfo: (12001) Matched tech-prefix 1 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_get_addrinfo: (12001) Matched zone prefix 2 and remainder 001 Jun 7 22:07:44.892: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49FFA4B8 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2 Jun 7 22:07:44.89 HQ-R1#2: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone and z_invianamep=GK Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49FFA4B8 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=2 Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone and z_outvianamep=GK Jun 7 22:07:44.892: //EE6A58F08490/EE6A58F08492/GK/rassrv_arq_select_viazone: Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am that viazone. Continue normal ARQ processing Jun 7 22:07:44.892: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 22:07:44.912: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:44.912: ////GK/gk_rassrv_arq: arqp=0x49E8C538,crv=0x22, answerCall=1 Jun 7 22:07:44.912: //EE6A58F08490/EE6A58F08492/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 22:07:44.928: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 22:07:44.936: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1# HQ-R1#do u all On Mon, Jun 7, 2010 at 6:19 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Can you paste the following: sh gatek gw Also deb gatek main 10 for a succes and a failed call thanks Date: Mon, 7 Jun 2010 05:50:23 -0400 Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call From: daniyal.vo...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Yeah I configured IP-IPGW but that doesn't matter and i took it out invia outvia it didn't help me as well still have issue with 4 Digit call site to site i am assuming H323 tcp connection time out problem could be but not sure coz i also change timer settings but it didn't help me as well any other idea ??? Thx Dani On Mon, Jun 7, 2010 at 4:44 AM, Angel Perez gorr...@hotmail.com wrote: Hi: The outvia and invia comands are for ip to ip gw and the show gateke calls doesn't show an ip2ip gw call... Date: Sun, 6 Jun 2010 19:48:27 -0400 From: daniyal.vo...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call I checked codec is G729 and here is config gatekeeper zone local GK cisco.com 142.1.64.254 invia GK outvia GK zone prefix GK 2* zone prefix GK 3* zone prefix GK 4* no shutdown === HQ-R1(config)#do sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 26-62701 3 16(Kbps) Endpt(s): Alias E.164Addr src EP: CUCME 4001 CallSignalAddr Port RASSignalAddr Port 142.102.66.254 1720 142.102.66.254 52357 Endpt(s): Alias E.164Addr dst EP: GK_Trunk_112001 CallSignalAddr Port RASSignalAddr Port 172.25.105.101 1720 172.25.105.101 32957 == HQ-R1(config)#do debug gatekeeper main 10 HQ-R1(config)# Jun 7 01:18:04.070: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1(config)# Jun 7 01:18:05.866: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.866: ////GK/gk_rassrv_arq: arqp
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi: Once the ephones have registered to srst add privacy button under ephone. Let us know, hth Date: Sun, 6 Jun 2010 21:46:26 +0100 From: cci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Hi, Sorry for the incomplete email earlier. Has anyone tried cBarge in SRST mode? I have my conference bridge registered to telephony service when it goes into SRST. I have used priority 3 as the telephony-service address. When my phones go in SRST mode, I can see they have shared line and when I am making a call on this shared line, I can see both phones ringing. However, if i receive call at one phone and try to Barge in using another all I get is a dialtone..no Cbarge softkey or anything even thou Cbarge key is configured under ephone template. Also privacy is off at telephony service. I even went into ephones and turned the privacy off but it is still not working. For some reason its not bringing the remote-in-use softkey template. I am getting a new call softkey template. R2#sh ephone ephone-1[0] Mac:0017.9402.CF34 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:11 privacy:0 IP:192.168.10.22 33399 7961 keepalive 0 max_line 6 button 1: dn 5 number 4001 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw ephone-2[1] Mac:0030.94C3.EE93 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 11/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:192.168.10.17 52129 Telecaster 7960 keepalive 3 max_line 6 button 1: dn 7 number 4002 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw Adios _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
Hi: The outvia and invia comands are for ip to ip gw and the show gateke calls doesn't show an ip2ip gw call... Date: Sun, 6 Jun 2010 19:48:27 -0400 From: daniyal.vo...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call I checked codec is G729 and here is config gatekeeper zone local GK cisco.com 142.1.64.254 invia GK outvia GK zone prefix GK 2* zone prefix GK 3* zone prefix GK 4* no shutdown === HQ-R1(config)#do sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 26-62701 3 16(Kbps) Endpt(s): Alias E.164Addr src EP: CUCME 4001 CallSignalAddr Port RASSignalAddr Port 142.102.66.254 1720 142.102.66.254 52357 Endpt(s): Alias E.164Addr dst EP: GK_Trunk_112001 CallSignalAddr Port RASSignalAddr Port 172.25.105.101 1720 172.25.105.101 32957 == HQ-R1(config)#do debug gatekeeper main 10 HQ-R1(config)# Jun 7 01:18:04.070: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1(config)# Jun 7 01:18:05.866: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.866: ////GK/gk_rassrv_arq: arqp=0x49EB9F38,crv=0x47, answerCall=0 Jun 7 01:18:05.866: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/gk_dns_query: No Name servers Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_get_addrinfo: (12001) Matched tech-prefix 1 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_get_addrinfo: (12001) Matched zone prefix 2 and remainder 001 Jun 7 01:18:05.866: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49FFA4B8 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2 Jun 7 01:18:05.86 HQ-R1(config)#6: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone and z_invianamep=GK Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49FFA4B8 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=2 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone and z_outvianamep=GK Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am that viazone. Continue normal ARQ processing Jun 7 01:18:05.866: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 01:18:05.886: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.886: ////GK/gk_rassrv_arq: arqp=0x49E86800,crv=0x19, answerCall=1 Jun 7 01:18:05.886: //5AF60F72841C/5AF60F72841E/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 01:18:05.902: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.910: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1(config)# == HQ-R1(config)#do debug ras H.323 RAS Messages debugging is on HQ-R1(config)# Jun 7 01:18:27.978: RecvUDP_IPSockData successfully rcvd message of length 193 from 142.102.66.254:52357 Jun 7 01:18:27.978: ARQ (seq# 4045) rcvdparse_arq_nonstd: ARQ Nonstd decode succeeded, remlen = 1141186144 Jun 7 01:18:27.978: IPSOCK_RAS_sendto: msg length 66 from 142.1.64.254:1719 to 142.102.66.254: 52357 Jun 7 01:18:27.978: RASLib::RASSendACF: ACF (seq# 4045) sent to 142.102.66.254 Jun 7 01:18:27.994: h323chan_chn_process_read_socket Jun 7 01:18:27.994: h323chan_chn_process_read_socket: fd=0 of type LISTENING has data Jun 7 01:18:27.998: h323chan_chn_process_read_socket Jun 7 01:18:27.998: h323chan_chn_process_read_socket: fd=3 of type ACCEPTED has data Jun 7 01:18:27.998: h323chan_chn_process_read_socket: h323chan accepted/connected fd=3 h323chan_dgram_send:Sent UDP msg. Bytes sent: 136 to 142.1.64.254:1719 fd=2 Jun 7 01:18:28.002: RASLib::GW_RASSendARQ: ARQ (seq# 3950) sent to 142.1.64.254 Jun 7 01:18:28.006: RecvUDP_IPSockData successf HQ-R1(config)#ully rcvd message of length 136 from 142.102.64.254:53515 Jun 7 01:18:28.006: ARQ (seq# 3950) rcvdparse_arq_nonstd: ARQ Nonstd decode succeeded, remlen = 1141186144 Jun 7 01:18:28.006: IPSOCK_RAS_sendto: msg length 36 from 142.1.64.254:1719 to 142.102.64.254: 53515 Jun 7 01:18:28.006:
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Mmm... not 100% sure know but I think that once you add the privacy buttom command it should stay there even if the phone registers back to ucm (I'm assuming that you are using srst mode auto all) Anyway I'll try to lab it during the day if I've time and update thanks Date: Mon, 7 Jun 2010 10:00:49 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Thanks Angel. I am at work but will give it a go today. I am just wondering that everytime when phones will go in SRST..do I have to go into ephones and add the button? this does not look practical. I will give it a go anyways. vc On Mon, Jun 7, 2010 at 9:39 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Once the ephones have registered to srst add privacy button under ephone. Let us know, hth Date: Sun, 6 Jun 2010 21:46:26 +0100 From: cci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Hi, Sorry for the incomplete email earlier. Has anyone tried cBarge in SRST mode? I have my conference bridge registered to telephony service when it goes into SRST. I have used priority 3 as the telephony-service address. When my phones go in SRST mode, I can see they have shared line and when I am making a call on this shared line, I can see both phones ringing. However, if i receive call at one phone and try to Barge in using another all I get is a dialtone..no Cbarge softkey or anything even thou Cbarge key is configured under ephone template. Also privacy is off at telephony service. I even went into ephones and turned the privacy off but it is still not working. For some reason its not bringing the remote-in-use softkey template. I am getting a new call softkey template. R2#sh ephone ephone-1[0] Mac:0017.9402.CF34 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:11 privacy:0 IP:192.168.10.22 33399 7961 keepalive 0 max_line 6 button 1: dn 5 number 4001 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw ephone-2[1] Mac:0030.94C3.EE93 TCP socket:[3] activeLine:0 REGISTERED in SCCP ver 11/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:8 privacy:0 IP:192.168.10.17 52129 Telecaster 7960 keepalive 3 max_line 6 button 1: dn 7 number 4002 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE shared Preferred Codec: g711ulaw Adios Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again)
Hi, you are partially right: When your phones registered back to ucm the e-phone stay there but dial-peer generated by cme srst doesn't point to anything (you can varify this with show dial-peer voice summ). But it's true that sometimes (it think that it is a bug) the ephone dial-peer still points to the dn number so once that your phones register back to ucm you can have problems with incomming calls With mgcp you don't have this problem becouse the gw is totally under ucm control so an incoming call won't match dial-peers , with h323 gw is different and ephone dial-peer will be matched, but you can easly change this behaviour like this: telephony service ... srst mode auto all max-dn 10 preference 5 ... exit dial-peer hunt ? 0-7 Dial-peer hunting choices, listed in hunting order within each choice: 0 - Longest match in phone number, explicit preference, random selection. 1 - Longest match in phone number, explicit preference, least recent use. 2 - Explicit preference, longest match in phone number, random selection. 3 - Explicit preference, longest match in phone number, least recent use. 4 - Least recent use, longest match in phone number, explicit preference. 5 - Least recent use, explicit preference, longest match in phone number. 6 - Random selection. 7 - Least recent use. dial-peer hunt 3 dial-peer voice 1000 voip description toUCM destination-pattern 1... preference 1 ... This way when a call enter the gw it will match the preference first instead longest match and calls will still work in srst mode and out srst mode. Try it and play a littel bit with it, you will find that is not difficult ps: also there is another bug with cme srst auto all, if you modify the auto generated ephones or ephones dn (let say name or label) sometimes when you go back to ucm and then to srts again the ephone won't take a dn, it will register to srst but the phone display won't show a line, in this case reload the router and everithing will start working as expected (this have been discused on the list) As you see cme srst is not pretty stable but with this two trick you can make it works easly if you have littel experience with it hth Date: Mon, 7 Jun 2010 10:16:47 +0100 Subject: Re: [OSL | CCIE_Voice] CBarge in SRST ~(Again) From: cci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com No Angel.. I am using srst mode auto-prov none.. I was thinking that if I use all then that will create ephone permenantly so after link comes back up (or primary ccm) then phones will register with the CUCM while ephones will still stay there...I have not tested this but this one colleague of mine had inbound calls issue after phones registred back to CUCM as the call was going to the ehpones instead of taking the voip dial-peer. Have you tested this? vc On Mon, Jun 7, 2010 at 10:00 AM, ccie voice cci...@gmail.com wrote: Thanks Angel. I am at work but will give it a go today. I am just wondering that everytime when phones will go in SRST..do I have to go into ephones and add the button? this does not look practical. I will give it a go anyways. vc On Mon, Jun 7, 2010 at 9:39 AM, Angel Perez gorr...@hotmail.com wrote: Hi: Once the ephones have registered to srst add privacy button under ephone. Let us know, hth Date: Sun, 6 Jun 2010 21:46:26 +0100 From: cci...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CBarge in SRST ~(Again) Hi, Sorry for the incomplete email earlier. Has anyone tried cBarge in SRST mode? I have my conference bridge registered to telephony service when it goes into SRST. I have used priority 3 as the telephony-service address. When my phones go in SRST mode, I can see they have shared line and when I am making a call on this shared line, I can see both phones ringing. However, if i receive call at one phone and try to Barge in using another all I get is a dialtone..no Cbarge softkey or anything even thou Cbarge key is configured under ephone template. Also privacy is off at telephony service. I even went into ephones and turned the privacy off but it is still not working. For some reason its not bringing the remote-in-use softkey template. I am getting a new call softkey template. R2#sh ephone ephone-1[0] Mac:0017.9402.CF34 TCP socket:[1] activeLine:0 REGISTERED in SCCP ver 17/9 mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0 caps:11 privacy:0 IP:192.168.10.22 33399 7961 keepalive 0 max_line 6 button 1: dn 5 number 4001 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE button 2: dn 6 number 4021 CM Fallback CH1 IDLE CH2 IDLE CH3 IDLE CH4 IDLE CH5 IDLE CH6 IDLE CH7 IDLE CH8 IDLE
Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
Hi: Can you paste the following: sh gatek gw Also deb gatek main 10 for a succes and a failed call thanks Date: Mon, 7 Jun 2010 05:50:23 -0400 Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call From: daniyal.vo...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Yeah I configured IP-IPGW but that doesn't matter and i took it out invia outvia it didn't help me as well still have issue with 4 Digit call site to site i am assuming H323 tcp connection time out problem could be but not sure coz i also change timer settings but it didn't help me as well any other idea ??? Thx Dani On Mon, Jun 7, 2010 at 4:44 AM, Angel Perez gorr...@hotmail.com wrote: Hi: The outvia and invia comands are for ip to ip gw and the show gateke calls doesn't show an ip2ip gw call... Date: Sun, 6 Jun 2010 19:48:27 -0400 From: daniyal.vo...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call I checked codec is G729 and here is config gatekeeper zone local GK cisco.com 142.1.64.254 invia GK outvia GK zone prefix GK 2* zone prefix GK 3* zone prefix GK 4* no shutdown === HQ-R1(config)#do sh gatekeeper call Total number of active calls = 1. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 26-62701 3 16(Kbps) Endpt(s): Alias E.164Addr src EP: CUCME 4001 CallSignalAddr Port RASSignalAddr Port 142.102.66.254 1720 142.102.66.254 52357 Endpt(s): Alias E.164Addr dst EP: GK_Trunk_112001 CallSignalAddr Port RASSignalAddr Port 172.25.105.101 1720 172.25.105.101 32957 == HQ-R1(config)#do debug gatekeeper main 10 HQ-R1(config)# Jun 7 01:18:04.070: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1(config)# Jun 7 01:18:05.866: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.866: ////GK/gk_rassrv_arq: arqp=0x49EB9F38,crv=0x47, answerCall=0 Jun 7 01:18:05.866: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/gk_dns_query: No Name servers Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_get_addrinfo: (12001) Matched tech-prefix 1 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_get_addrinfo: (12001) Matched zone prefix 2 and remainder 001 Jun 7 01:18:05.866: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x49FFA4B8 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: matched zone is GK, and z_invianamelen=2 Jun 7 01:18:05.86 HQ-R1(config)#6: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone and z_invianamep=GK Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x49FFA4B8 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: matched zone is GK, and z_outvianamelen=2 Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone and z_outvianamep=GK Jun 7 01:18:05.866: //5AF60F72841C/5AF60F72841E/GK/rassrv_arq_select_viazone: Received ARQ for a zone (GK) that has an outviazone (GK) specified, but I am that viazone. Continue normal ARQ processing Jun 7 01:18:05.866: ////GK/gk_rassrv_get_ingress_network: ARQ non-std ingress network = 1 Jun 7 01:18:05.886: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.886: ////GK/gk_rassrv_arq: arqp=0x49E86800,crv=0x19, answerCall=1 Jun 7 01:18:05.886: //5AF60F72841C/5AF60F72841E/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC Jun 7 01:18:05.902: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup Jun 7 01:18:05.910: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup HQ-R1(config)# == HQ-R1(config)#do debug ras H.323 RAS Messages debugging is on HQ-R1(config)# Jun 7 01:18:27.978: RecvUDP_IPSockData successfully rcvd message of length 193 from 142.102.66.254:52357 Jun 7 01:18:27.978: ARQ (seq# 4045) rcvdparse_arq_nonstd: ARQ Nonstd decode succeeded, remlen = 1141186144 Jun 7 01:18:27.978: IPSOCK_RAS_sendto: msg length 66 from 142.1.64.254:1719 to 142.102.66.254: 52357 Jun 7 01:18:27.978: RASLib::RASSendACF: ACF (seq# 4045) sent to 142.102.66.254 Jun 7 01:18:27.994: h323chan_chn_process_read_socket Jun 7 01:18:27.994: h323chan_chn_process_read_socket: fd=0 of type LISTENING has data Jun 7 01:18:27.998
Re: [OSL | CCIE_Voice] VOL2 LAB2 weird cme to ucm call over gk problem
Hi its a bug, it have been said several times
CSCsl74701
http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15752.html
Date: Mon, 7 Jun 2010 12:16:16 +0100
From: kevin.hobson2...@ntlworld.com
To: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] VOL2 LAB2 weird cme to ucm call over gk problem
Hi All,
I have a really werid issue with calls from UCME to UCM.
The issue is that if i call from BR2 to HQ and do a show gatek call it shows
the bandwidth being usesd as 128k. See below:
gk-cube#sh gatek call
Total number of active calls = 1.
GATEKEEPER CALL INFO
LocalCallIDAge(secs) BW
82-43591 21 16(Kbps)
Endpt(s): Alias E.164Addr
src EP: BR2-RTR 3001
CallSignalAddr Port RASSignalAddr Port
10.10.110.3 1720 10.10.110.3 58865
Endpt(s): Alias E.164Addr
dst EP: gk-trunk_21#5001
CallSignalAddr Port RASSignalAddr Port
10.10.210.111720 10.10.210.1132786
If i debug h225 asn1 i see that the CME is requesting 16k but the UCM is asking
for 128k. See below:
CME
value RasMessage ::= admissionRequest :
{
requestSeqNum 9277
callType pointToPoint : NULL
callModel direct : NULL
endpointIdentifier {49887C840001}
destinationInfo
{
dialedDigits : 1#5002
}
srcInfo
{
dialedDigits : 3001,
h323-ID : {BR2-RTR}
}
bandWidth 160
UCM
value RasMessage ::= admissionRequest :
{
requestSeqNum 1393
callType pointToPoint : NULL
endpointIdentifier {4857A5C80003}
destinationInfo
{
dialedDigits : 5002
}
srcInfo
{
dialedDigits : 3001
}
srcCallSignalAddress ipAddress :
{
ip ''H
port 20946
}
bandWidth 1280
When the call is connected i get no codec sent on the hq phone and g729 on the
BR2 phone.
If i then enable BRQ on the UCM services when the phone rings it requests 128k
again:
gk-cube#sh gatek call
Total number of active calls = 1.
GATEKEEPER CALL INFO
LocalCallIDAge(secs) BW
85-238615 128(Kbps)
Endpt(s): Alias E.164Addr
src EP: BR2-RTR 3001
CallSignalAddr Port RASSignalAddr Port
10.10.110.3 1720 10.10.110.3 58865
Endpt(s): Alias E.164Addr
dst EP: gk-trunk_21#5002
CallSignalAddr Port RASSignalAddr Port
10.10.210.111720 10.10.210.1132786
But when it connects this goes down to 16k:
gk-cube#sh gatek call
Total number of active calls = 1.
GATEKEEPER CALL INFO
LocalCallIDAge(secs) BW
86-24672 8 16(Kbps)
Endpt(s): Alias E.164Addr
src EP: BR2-RTR 3001
CallSignalAddr Port RASSignalAddr Port
10.10.110.3 1720 10.10.110.3 58865
Endpt(s): Alias E.164Addr
dst EP: gk-trunk_21#5001
CallSignalAddr Port RASSignalAddr Port
10.10.210.111720 10.10.210.1132786
The phones also show g729 on both of them for the codec in use.
The region is g729 and the dp is assigned this region.
A ucm call the other way only requests 16k.
All help appreciated,
On 7 June 2010 11:46, ccie_voice-requ...@onlinestudylist.com wrote:
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Today's Topics:
1. Setting up Voicemail to send Email - CME/CUE 7.0 (Ashar Siddiqui)
2. Re: Setting up Voicemail to send Email - CME/CUE 7.0 (Angel Perez)
3. Re: Setting up Voicemail to send Email - CME/CUE 7.0
(kerboute kerboute)
--
Message: 1
Date: Mon, 7 Jun 2010 11:28:09 +0100
From: Ashar Siddiqui siddas...@gmail.com
To: ccie_voice@onlinestudylist.com
Subject: [OSL | CCIE_Voice] Setting up Voicemail to send Email -
CME/CUE 7.0
Message-ID: 018701cb062c$221091b0$6631b5...@com
Content-Type: text/plain; charset=us-ascii
Hello all,
I am setting up Email notification for one of my
Re: [OSL | CCIE_Voice] Shared lines in CME SRST
Hi: Do you have privacy on at any of the phones before going to srst? Also sometimes you have to reload the gw with cme srst to make it works properly hth From: 1.matt.h...@gmail.com Date: Sat, 5 Jun 2010 22:55:23 -0500 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Shared lines in CME SRST Problem I'm having is as follows: CME SRST Two phones have a shared line 2010 I have srst dn mode set to octo, when registering, one or both of the phones always come up as remote in use and stay that way, no matter how many times I get them to unregister and reregister. Anyone else seen this before? Thanks Matt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [SUSPECTED SPAM] RE: Lab and Language settings
Your are right, NDA affects those candidates who have attempted the lab, anyway, please for these people under NDA don't answer any question regarding the lab http://www.cisco.com/web/learning/downloads/guest/learning/c644/ccmigration_09186a00803641d2.pdf http://www.cisco.com/web/learning/le3/ccie/exam/violation_rules.html Thanks Subject: [SUSPECTED SPAM] RE: [OSL | CCIE_Voice] Lab and Language settings From: r.ochi...@mfient.com To: gorr...@hotmail.com; jon1...@hotmail.com CC: siddas...@gmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 22:33:43 +0300 It isn’t true that I cannot use the word lab….i can ask what the temperature is like in the lab, is the proctor in the lab, what is the lab topology like without necessarily breaking the NDA. You can ask anything, It’s upon me the person restricted by NDA to tell you that I cannot answer that as I’ll be breaking NDA I think NDA would apply to those who’ve attempted or passed the lab. Others have not agreed to any NDA From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez Sent: Friday, June 04, 2010 9:37 PM To: jon1...@hotmail.com; siddas...@gmail.com; osl osl Subject: Re: [OSL | CCIE_Voice] Lab and Language settings Don't worry, just think that if you include the word lab in you question you would be breaking NDA :( From: jon1...@hotmail.com To: gorr...@hotmail.com; siddas...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab and Language settings Date: Sat, 5 Jun 2010 03:19:11 +0900 Thanks, and sorry didn’t really mean to ask contents, more of a rough info. question as in blueprints don’t say it, so was pretty much curious. Thanks for the heads up From: Angel Perez Sent: Saturday, June 05, 2010 2:25 AM To: siddas...@gmail.com ; jon1...@hotmail.com ; osl osl Subject: RE: [OSL | CCIE_Voice] Lab and Language settings Hi Jon, you can't ask anything about exam contents, sorry From: siddas...@gmail.com To: jon1...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 15:39:27 +0100 Subject: Re: [OSL | CCIE_Voice] Lab and Language settings No, I don’t think so.. As a rule of thumb just select US (English) where ever needed. Ash From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: 04 June 2010 15:36 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab and Language settings Hi Was just curious, does the lab get involved at all with locale settings or how to upload files for the languages to be available? I didn’t see much mention of it in the blueprint, so was curious for those who have attempted, any mention of it? Thanks Jon Hotmail: Powerful Free email with security by Microsoft. Get it now. Hotmail: Trusted email with powerful SPAM protection. Sign up now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...
Hi: Sometimes you have to reload the gw to make presence works hth Date: Sat, 5 Jun 2010 12:18:43 +0200 From: findko...@gmail.com To: salman.shaik...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ... and maybe sip-ua presence enable will help? On Sat, Jun 5, 2010 at 12:16 PM, kobel findko...@gmail.com wrote: try create cnf-files restart the phones. On Sat, Jun 5, 2010 at 4:21 AM, Shadow of Voice salman.shaik...@gmail.com wrote: Hi Guys I have issue when configure presence in CME I allow subscribe and allow watch globally still can't see caller list on missed call does any one know where i am wrong and why my CME presence caller-list is not working ! presence presence call-list allow subscribe ! ephone-dn 2 octo-line number 4002 no-reg primary description +6524044002 name SiteC-Ph2 allow watch call-forward busy 4220 call-forward noan 4220 timeout 20 ! ! ephone 1 device-security-mode none mac-address 001A.A1C8.0H8F ephone-template 1 blf-speed-dial 1 4002 label SiteC-Ph2 type 7961 button 1:1 3:3 4:5 ! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VLAN interfaces down
Hi:
If the vlan.dat file is deleted you will get this result
Make sure that the vlan exists and also that it is active:
vlan 130
create
name data
status active
vlan 240
create
name voice
status active
hth
Date: Thu, 3 Jun 2010 19:08:22 -0400
From: amccar...@cciequest.com
To: ciscovoiceg...@gmail.com
CC: ccie_voice@onlinestudylist.com
Subject: Re: [OSL | CCIE_Voice] VLAN interfaces down
Hey Bro,
I ran into an issue similar to that before but mine was because there
was no phone connected to the port.
Antonio
Quoting Matthew Berry ciscovoiceg...@gmail.com:
I see this issue from time to time. The VLAN interfaces on my BR1-RTR
show a state of up, but line protocol is down.
I made sure that there are ports with the vlans configured. I reload
the router. I also made sure the vlans were in existence.
At this point, my BR1-RTR is useless until I get this working. Any ideas?
interface Vlan130
ip address 10.10.101.1 255.255.255.0
!
interface Vlan240
ip address 10.10.201.1 255.255.255.0
...
interface FastEthernet1/1
switchport trunk native vlan 130
switchport mode trunk
switchport voice vlan 240
BR1-RTR#show vlan-switch br
VLAN Name Status Ports
-
---
...
130 DATA active Fa1/1, Fa1/15
240 PHONES active Fa1/1, Fa1/2, Fa1/3, Fa1/4
Fa1/5, Fa1/6, Fa1/7, Fa1/8
Fa1/9, Fa1/10, Fa1/11, Fa1/12
Fa1/13, Fa1/14, Fa1/15
BR1-RTR#show ip int bri
Interface IP-Address OK? Method Status
Protocol
...
Vlan130 10.10.101.1 YES manual up
down
Vlan240 10.10.201.1 YES NVRAM up
down
--
*Matthew Berry*
/A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written/
*_Vitals:_*
*GVoice: *+1.612.424.5044
*Gmail*: ciscovoiceg...@gmail.com
*Skype*: ciscovoiceguru
*Twitter*: ciscovoiceguru
*_Cert Stats:_*
Cisco Cert Journey Began: Jan 1, 2009
1st Lab Attempt: Aug 16, 2010
___
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com
_
Hotmail: Powerful Free email with security by Microsoft.
https://signup.live.com/signup.aspx?id=60969___
For more information regarding industry leading CCIE Lab training, please visit
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Re: [OSL | CCIE_Voice] Lab and Language settings
Hi Jon, you can't ask anything about exam contents, sorry From: siddas...@gmail.com To: jon1...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 15:39:27 +0100 Subject: Re: [OSL | CCIE_Voice] Lab and Language settings No, I don’t think so.. As a rule of thumb just select US (English) where ever needed. Ash From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: 04 June 2010 15:36 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab and Language settings Hi Was just curious, does the lab get involved at all with locale settings or how to upload files for the languages to be available? I didn’t see much mention of it in the blueprint, so was curious for those who have attempted, any mention of it? Thanks Jon _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lab and Language settings
Don't worry, just think that if you include the word lab in you question you would be breaking NDA :( From: jon1...@hotmail.com To: gorr...@hotmail.com; siddas...@gmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Lab and Language settings Date: Sat, 5 Jun 2010 03:19:11 +0900 Thanks, and sorry didn’t really mean to ask contents, more of a rough info. question as in blueprints don’t say it, so was pretty much curious. Thanks for the heads up From: Angel Perez Sent: Saturday, June 05, 2010 2:25 AM To: siddas...@gmail.com ; jon1...@hotmail.com ; osl osl Subject: RE: [OSL | CCIE_Voice] Lab and Language settings Hi Jon, you can't ask anything about exam contents, sorry From: siddas...@gmail.com To: jon1...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 4 Jun 2010 15:39:27 +0100 Subject: Re: [OSL | CCIE_Voice] Lab and Language settings No, I don’t think so.. As a rule of thumb just select US (English) where ever needed. Ash From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: 04 June 2010 15:36 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Lab and Language settings Hi Was just curious, does the lab get involved at all with locale settings or how to upload files for the languages to be available? I didn’t see much mention of it in the blueprint, so was curious for those who have attempted, any mention of it? Thanks Jon Hotmail: Powerful Free email with security by Microsoft. Get it now. _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts
Hi Amy: I'm working on my own gear, other people has experience similar behaviour http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15814.html I can't post my configs (wr erase yesterday :( ) but I will try to recreate the issue today and post Regars Date: Wed, 2 Jun 2010 23:52:39 -0400 Subject: Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts From: ar...@ipexpert.com To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Angel, I have not experienced this behavior. Can you post the configuration of the router hosting MVA? Are you using Proctor Labs vRack Sessions or a home lab? Thank you, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Angel Perez gorr...@hotmail.com Date: Wed, 2 Jun 2010 17:21:42 + To: osl osl ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] MVA works but I don't hear prompts Hi: When I call mva number from pstn, the rd number is matched so I enter de pin 12345 # then 1 # for call and finally the number I want to call 911 # The problem I have is that between the prompts there is a silence of 5 - 7 sec, sometimes the prompt doesn't sounds, but if I press the correct order of digits: 12345 #1 #911 # the call proceeds If the prompt doesn't sounds and I still waiting the call disconects... It sounds like a problem with vm ware, but I'm not sure Anybody has seen this before??? Thanks Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] show vlan-s brief
Thanks all From: wormh...@sch.hu To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] show vlan-s brief Date: Wed, 2 Jun 2010 20:55:27 +0200 In case of ESW the suggested method is trunk mode. http://www.cisco.com/en/US/docs/ios/lanswitch/configuration/guide/lsw_hwic_ethsw_ic_ps6441_TSD_Products_Configuration_Guide_Chapter.html#wp1051730 - Original Message - From: Peter Farkas To: Angel Perez Cc: osl osl Sent: Wednesday, June 02, 2010 8:36 PM Subject: Re: [OSL | CCIE_Voice] show vlan-s brief I also cheked mine. Further interesting these different output of the same config. BR2-RTR#sh vlan-switch VLAN Name StatusPorts - --- 1default activeFa0/1/2, Fa0/1/3 200 DATA active 400 PHONES active 1002 fddi-default act/unsup 1003 token-ring-default act/unsup 1004 fddinet-default act/unsup 1005 trnet-defaultact/unsup VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2 - -- - -- -- -- -- 1enet 11 1500 - - ---1002 1003 200 enet 100200 1500 - - ---0 0 400 enet 100400 1500 - - ---0 0 1002 fddi 101002 1500 - - ---1 1003 1003 tr101003 1500 1005 0 --srb 1 1002 1004 fdnet 101004 1500 - - 1ibm -0 0 1005 trnet 101005 1500 - - 1ibm -0 0 BR2-RTR#sh run int Fas 0/1/0 Building configuration... Current configuration : 119 bytes ! interface FastEthernet0/1/0 switchport trunk native vlan 200 switchport mode trunk switchport voice vlan 400 end BR2-RTR#sh vlan-switch id 400 VLAN Name StatusPorts - --- 400 PHONES activeFa0/1/0, Fa0/1/1 VLAN Type SAID MTU Parent RingNo BridgeNo Stp BrdgMode Trans1 Trans2 - -- - -- -- -- -- 400 enet 100400 1500 - - ---0 0 - Original Message - From: Angel Perez To: osl osl Sent: Wednesday, June 02, 2010 7:53 PM Subject: [OSL | CCIE_Voice] show vlan-s brief Hi: When I configure the swich port of my hwic-esw with the old method: interface range fas 0/3/0 - 3 swicht mode trunk swicht trunk encap dot1q native vlan 200 swicht voice 300 I get the following result: sh vlan-s bri VLAN Name StatusPorts - --- 1default activeFa0/3/1, Fa0/3/3 300 voiceactiveFa0/3/1, Fa0/3/3 200 data active Everything works as expected, it is just a problem in the show comand, but I wonder if the proctor wants to check the vlans with this command he/she could think that it is wrong... With the new method: switch mode acc switch acc vlan 200 swith voice vlan 300 I get this output, that looks better: sh vlan-s bri VLAN Name StatusPorts - --- 1default active 300 voiceactiveFa0/3/1, Fa0/3/2, Fa0/3/3 200 data activeFa0/3/1, Fa0/3/2, Fa0/3/3 What do you think about it? Thanks Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Presence in 7.0 and 7.1
Hi Daniyal: Check question 10 and answer 11 from Ben at may 17 https://learningnetwork.cisco.com/message/68646#68649 Date: Thu, 3 Jun 2010 10:06:03 -0400 From: daniyal.vo...@gmail.com To: earl.ho...@pcmall.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME Presence in 7.0 and 7.1 Thanks Earl, Then i would say it's rumor info, but one quick question about the CME presence does any one have idea about the CME presence config how to configure coz in CME 7.0 we just need to allow watch but in CME 7.1 you have to enable presence and configured under ephone blf and blf list should be enable globally ... pls correct me if i am wrong .. anyways just want to confirm how to configure CME presence in CME 7.0 Thanks in advance. Daniyal On Thu, Jun 3, 2010 at 9:54 AM, Hough, Earl earl.ho...@pcmall.com wrote: From last we heard, everything is still at a 7.0 release. Nothing has been announced regarding a refresh of application versions to 7.1 Additionally, any major change like that is suppose to be announced with a 6 month lead-time before such changes go into affect. Search the archives here in the past couple of weeks and someone posted the definitive answer from Ben Ng regarding the specific versions of IOS, CUCM, CUC, UCCX, and CUPS. Earl Hough, CCIE #16508 (RS, Security) From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccievoice ccievoice Sent: Thursday, June 03, 2010 8:20 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CCIE Voice Lab v3.0 Equipment and Software List Hi, As per current Lab IOS software and lab equipment list shown below, dose any have idea in current lab cisco is using CME 7.0 or 7.1 also presence 7.1 as i heard i think they upgraded from CME 7.0 to 7.1 but i couldn't found even cisco didn't announced Pls let me know if any have any idea about the upgrading lab IOS Lab Equipment: 1. Cisco MCS-7845 Media Convergence Servers 2. Cisco 3825 Series Integrated Services Routers (ISR) 3. Cisco 2821 Series Integrated Services Routers (ISR) 4. ISR Modules and Interface Cards+ VWIC2-1MFT-T1/E1 - PVDM2 - HWIC-4ESW-POE - NME-CUE 5. Cisco Catalyst 3750 Series Switches 6. IP Phones and Soft Clients Software Versions Any major software release which has been generally available for six months is eligible for testing in the CCIE Voice Lab Exam. oCisco Unified Communications Manager 7.0 oCisco Unified Communications Manager Express 7.0 oCisco Unified Contact Center Express 7.0 oCisco Unified Presence 7.0 oCisco Unity Connection 7.0 oAll routers use IOS version 12.4T Train. oCisco Catalyst 3750 Series Switches uses 12.2 Main Train Network Interfaces 1. Fast Ethernet 2. Frame Relay Telephony Interfaces 1. T1 2. E1 Thanks Daniyal _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts [solved]
Hi, I can confirm that it's a vmware issue, First: I launched only pub and sub vm machines Then I relabed everything as yesterday triying to reproduce the same issue, prompts were working as expected but with 7 sec of silent between them, the call didn't disconnect ... Second: Then I started ccx, uc, cups and ad vm machines, I launched ccx integration plus cup integration at the same time, also I made some searches at ucm and uc all at the same time Then I tried mva and magic call were disconecting... Finally I stoped ccx, cups and uc and mva started working again Maybe there are a lot of vm machines for a single server (dual-core 8gb ram) Thanks Date: Thu, 3 Jun 2010 09:01:23 -0400 Subject: Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts From: ar...@ipexpert.com To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com If you are able to reproduce it, let me know. I am interested in troubleshooting our way outta this one. :-) Thank you, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http://www.ipexpert.com/ From: Angel Perez gorr...@hotmail.com Date: Thu, 3 Jun 2010 08:33:34 + To: Amy Ryan ar...@ipexpert.com, osl osl ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] MVA works but I don't hear prompts Hi Amy: I'm working on my own gear, other people has experience similar behaviour http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg15814.html I can't post my configs (wr erase yesterday :( ) but I will try to recreate the issue today and post Regars Date: Wed, 2 Jun 2010 23:52:39 -0400 Subject: Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts From: ar...@ipexpert.com To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Angel, I have not experienced this behavior. Can you post the configuration of the router hosting MVA? Are you using Proctor Labs vRack Sessions or a home lab? Thank you, Amy --- Amy Ryan – CCIE #24677 (Voice) Technical Instructor - IPexpert, Inc. Mailto: ar...@ipexpert.com http://ipexpert.com/ Telephone: +1.810.326.1444 Live Assistance, Please visit: www.ipexpert.com/chat http://www.ipexpert.com/chat eFax: +1.810.454.0130 IPexpert is a premier provider of Self-Study Workbooks, Video on Demand, Audio Tools, Online Hardware Rental and Classroom Training for the Cisco CCIE (RS, Voice, Security Service Provider) certification(s) with training locations throughout the United States, Europe, South Asia and Australia. Be sure to visit our online communities at www.ipexpert.com/communities http://www.ipexpert.com/communities and our public website at www.ipexpert.com http:// http:/// www.ipexpert.com/ From: Angel Perez gorr...@hotmail.com http://hotmail.com/ Date: Wed, 2 Jun 2010 17:21:42 + To: osl osl ccie_voice@onlinestudylist.com http://onlinestudylist.com/ Subject: [OSL | CCIE_Voice] MVA works but I don't hear prompts Hi: When I call mva number from pstn, the rd number is matched so I enter de pin 12345 # then 1 # for call and finally the number I want to call 911 # The problem I have is that between the prompts there is a silence of 5 - 7 sec, sometimes the prompt doesn't sounds, but if I press the correct order of digits: 12345 #1 #911 # the call proceeds If the prompt doesn't sounds and I still waiting the call disconects... It sounds like a problem with vm ware, but I'm not sure Anybody has seen this before??? Thanks Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Hotmail: Free, trusted and rich email service. Get it now. https://signup.live.com/signup.aspx?id=60969 _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MVA works but I don't hear prompts
Hi: When I call mva number from pstn, the rd number is matched so I enter de pin 12345 # then 1 # for call and finally the number I want to call 911 # The problem I have is that between the prompts there is a silence of 5 - 7 sec, sometimes the prompt doesn't sounds, but if I press the correct order of digits: 12345 #1 #911 # the call proceeds If the prompt doesn't sounds and I still waiting the call disconects... It sounds like a problem with vm ware, but I'm not sure Anybody has seen this before??? Thanks _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] show vlan-s brief
Hi: When I configure the swich port of my hwic-esw with the old method: interface range fas 0/3/0 - 3 swicht mode trunk swicht trunk encap dot1q native vlan 200 swicht voice 300 I get the following result: sh vlan-s bri VLAN Name StatusPorts - --- 1default activeFa0/3/1, Fa0/3/3 300 voiceactiveFa0/3/1, Fa0/3/3 200 data active Everything works as expected, it is just a problem in the show comand, but I wonder if the proctor wants to check the vlans with this command he/she could think that it is wrong... With the new method: switch mode acc switch acc vlan 200 swith voice vlan 300 I get this output, that looks better: sh vlan-s bri VLAN Name StatusPorts - --- 1default active 300 voiceactiveFa0/3/1, Fa0/3/2, Fa0/3/3 200 data activeFa0/3/1, Fa0/3/2, Fa0/3/3 What do you think about it? Thanks _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] ip rsvp bandwith
Thanks From: r.ochi...@mfient.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] ip rsvp bandwith Date: Tue, 1 Jun 2010 08:18:55 +0300 From SRND Configuration Recommendation Because the initial reservation will be larger than the actual packet flow, over-provisioning the RSVP and LLQ bandwidth is required to ensure that the desired number of calls can complete. When provisioning the RSVP bandwidth value for N calls, Cisco recommends that the Nth value be the worst-case bandwidth to ensure that the Nth call gets admitted.3-65 Cisco Unified Communications SRND (Based on Cisco Unified Communications Manager 7.x) OL-16394-05 Chapter 3 Network Infrastructure WAN Infrastructure For example: • To provision four G.729 streams: (3 ∗ 24) + 40 = 112 kbps • To provision four G.711 streams: (3 ∗ 80) + 96 = 336 kbps • To provision four 384 kbps video streams (G.729 audio) (3 ∗ (384 - 8) + 384) ∗ 1.07 = 1618 kbps • To provision four 384 kbps video streams (G.711 audio) (3 ∗ (384 - 64) + 384) ∗ 1.07 = 1438 kbps From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Angel Perez Sent: Monday, May 31, 2010 9:16 PM To: osl osl Subject: [OSL | CCIE_Voice] ip rsvp bandwith Hi all: For two g729 calls, how much band would you set at ip rsvp bandwith These are the two options: 1: ip rsvp bandwith 64 (40 + 24) or 2: ip rsvp bandwitn 80 (40 + 40) The first one looks find becouse once the call is completed the rsvp bandwith is reduced to 24 and a second call would be possible, but what happens if one call is ringing and in this moment a second call arrives... then the second call will be rejected due insufficient bandwith This is way I would use the second option, and also if two calls are stablished and a third call arrives rsvp will reject the third call (expected) 24+24 + 40 = 88 ; 88 80 What do you think? Regards Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] ip rsvp bandwith
Hi all: For two g729 calls, how much band would you set at ip rsvp bandwith These are the two options: 1: ip rsvp bandwith 64 (40 + 24) or 2: ip rsvp bandwitn 80 (40 + 40) The first one looks find becouse once the call is completed the rsvp bandwith is reduced to 24 and a second call would be possible, but what happens if one call is ringing and in this moment a second call arrives... then the second call will be rejected due insufficient bandwith This is way I would use the second option, and also if two calls are stablished and a third call arrives rsvp will reject the third call (expected) 24+24 + 40 = 88 ; 88 80 What do you think? Regards _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] SIP TRUNK
Hi: I have a sip trunk to my pstn router I'm trying to check the codec that the call is using but I can't this info at ucm traces or pstn gw debugs. I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't this info Any suggestion? _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] SIP TRUNK
Thanks, is it possible to check the call type also? Regards Subject: Re: [OSL | CCIE_Voice] SIP TRUNK From: ghopk...@wolf-rock.co.uk Date: Sat, 29 May 2010 20:04:36 +0100 CC: ccie_voice@onlinestudylist.com To: gorr...@hotmail.com Yes you should pick it up in the invite and OK messages thus m=audio 47100 RTP/AVP 8 0 18 98 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:98 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=nortpproxy:yes Regards Graham Hopkins On 29 May 2010, at 19:08, Brian Valentine wrote: You should try debug ccsip messages on the PSTN or CUBE router. It will show you the codec negotiation. On May 29, 2010 1:55 PM, Angel Perez gorr...@hotmail.com wrote: Hi: I have a sip trunk to my pstn router I'm trying to check the codec that the call is using but I can't this info at ucm traces or pstn gw debugs. I have try sip stack traces at ucm and also deb ccsip all at pstn, but I can't this info Any suggestion? Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Background Image Issue
Hi: Your problem is here: flash:/Desktops/320x196x4/ flash file should look like this: flash:Desktops/320x196x4/ hth Date: Thu, 27 May 2010 19:46:06 -0400 From: salman.shaik...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME Background Image Issue Hi can any have any idea why my image is not showing. here is my config and debug ... CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/T-VOICE-7961.PNG URL=TFTP:Desktops/320x196x4/VOICE1-7961.PNG/ /CiscoIPPhoneImageList ! ! SC-R3#dir Directory of flash:/Desktops/320x196x4/ 53 -rw- 165 May 27 2010 22:33:34 +00:00 List.xml 54 -rw- 148026 May 27 2010 22:34:14 +00:00 VOICE1-7961.PNG 55 -rw- 10855 May 27 2010 22:34:36 +00:00 T-VOICE-7961.PNG 128034816 bytes total (44347392 bytes free) ! ! tftp-server flash:Desktops/320x196x4/T-VOICE-7961.PNG tftp-server flash:Desktops/320x196x4/VOICE1-7961.PNG tftp-server flash:Desktops/320x196x4/List.xml ! ! SC-R3(config)#do debug tftp events *May 27 22:49:38.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:42.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:46.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:50.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:54.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:58.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# when i press settings User Preferences Background Image it shows me requesting selections but didn't see any image and then after a min try it shows selection Unavailable Thanks _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA issue
Hi: Take a look to DNA on UCM, maybe you can have some information there hth Date: Thu, 27 May 2010 17:48:59 -0700 From: lme...@signal.ca To: bga...@gmail.com CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA issue Checked that.. I got TAC involved.. They do not know where it is coming from. Work around is to have a partial match for 4 digits.. This also appends the 7 but at least it matches on 4 and not complete….. Leslie From: Bo Gao [mailto:bga...@gmail.com] Sent: Thursday, May 27, 2010 4:29 PM To: Leslie Meade Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] MVA issue Can you check Mobile Voice Access Number field in the Service Parameter to see if there are any prefix? Bo On Thu, May 27, 2010 at 10:21 AM, Leslie Meade lme...@signal.ca wrote: I know that general support is not the best option here. But I will ask.. I just noticed that my MVA is not working. Users can log into the system and attempt to dial, but the then get dead air Debugs show that some where I am appending an extra 7 to the remote destination profile, but I do not understand where. I am not using any transformation patterns, the gateway is not adding any digits.. The debug from vxml app on the gateway is showing correct numbers, debug ccapi is also showing correct, it is something on the Callmanager that is doing this. How can I track down what is adding the 7 ? 05/25/2010 20:09:57.870 CCM|SPROC :: stripAndPrependDigits- The number 777 is prepended with prefix 7, updated number=82284339|CLID::StandAloneClusterNID::CCM7-01LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|SPROC getCtrlPid - callingNum=, inputCtrlPid=(1,100,175,1)|CLID::StandAloneClusterNID::x.x.x.xLVL::DetailedMASK::0800 05/25/2010 20:09:57.870 CCM|DbMobility: getMatchedRemDest starts: cnumber = |CLID::StandAloneClusterNID:: x.x.x.x LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|DbMobility: getMatchedRemDest: full match case|CLID::StandAloneClusterNID:: x.x.x.x LVL::DetailedMASK::ff 05/25/2010 20:09:57.870 CCM|DbMobility: can't find remdest in map|CLID::StandAloneClusterNID::CCM7-01LVL::ErrorMASK::ff 05/25/2010 20:09:57.871 CCM|H225D::restart0_RSVPRegisterRes, CI=24083271, branch=0|CLID::StandAloneClusterNID:: x.x.x.x CT::1,100,152,1.1IP::10.1.1.5DEV::LVL::DetailedMASK::0800 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME Background Image Issue
Hi again: In my case is working like this: br2#sh flash: -#- --length-- -date/time-- path 86 0 Apr 15 2010 17:14:52 Desktops 87 0 Apr 15 2010 17:14:56 Desktops/320x196x4 88 647 Apr 15 2010 17:15:00 Desktops/320x196x4/logo.png 89 239 Apr 15 2010 17:15:18 Desktops/320x196x4/logoTH.png 90 150 Apr 15 2010 17:15:44 Desktops/320x196x4/List.xml br2#more Desktops/320x196x4/List.xml CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/logoTH.png URL=TFTP:Desktops/320x196x4/logo.png/ /CiscoIPPhoneImageList tftp-server flash:Desktops/320x196x4/List.xml tftp-server flash:Desktops/320x196x4/logo.png tftp-server flash:Desktops/320x196x4/logoTH.png Then create cnf files and reset just in case From: gorr...@hotmail.com To: salman.shaik...@gmail.com; ccie_voice@onlinestudylist.com Date: Fri, 28 May 2010 07:53:04 + Subject: Re: [OSL | CCIE_Voice] CME Background Image Issue Hi: Your problem is here: flash:/Desktops/320x196x4/ flash file should look like this: flash:Desktops/320x196x4/ hth Date: Thu, 27 May 2010 19:46:06 -0400 From: salman.shaik...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME Background Image Issue Hi can any have any idea why my image is not showing. here is my config and debug ... CiscoIPPhoneImageList ImageItem Image=TFTP:Desktops/320x196x4/T-VOICE-7961.PNG URL=TFTP:Desktops/320x196x4/VOICE1-7961.PNG/ /CiscoIPPhoneImageList ! ! SC-R3#dir Directory of flash:/Desktops/320x196x4/ 53 -rw- 165 May 27 2010 22:33:34 +00:00 List.xml 54 -rw- 148026 May 27 2010 22:34:14 +00:00 VOICE1-7961.PNG 55 -rw- 10855 May 27 2010 22:34:36 +00:00 T-VOICE-7961.PNG 128034816 bytes total (44347392 bytes free) ! ! tftp-server flash:Desktops/320x196x4/T-VOICE-7961.PNG tftp-server flash:Desktops/320x196x4/VOICE1-7961.PNG tftp-server flash:Desktops/320x196x4/List.xml ! ! SC-R3(config)#do debug tftp events *May 27 22:49:38.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:42.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:46.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:50.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:54.068: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# *May 27 22:49:58.064: TFTP: Looking for Desktops/320x196x4/List.xml SC-R3(config)# when i press settings User Preferences Background Image it shows me requesting selections but didn't see any image and then after a min try it shows selection Unavailable Thanks Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] H323 Gateway - Called Party Number Type: Unknown
Hi: Do you have any called party transformation in the gw called party transformation calling search space? hth Date: Fri, 28 May 2010 11:35:42 -0500 From: tamnhu...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] H323 Gateway - Called Party Number Type: Unknown Hi all, Not sure if someone already posted the issue below or not, but I could not find one on OSL, so I post it here. The problem I have is the H323 gateway outbound called party number Type always show Unknown, even though I set it to National in the UCM. However, my BR1 MGCP gateway shows correct Type: National. Here is the call flow: HQ phone -- dialling 16178632683 -- TP 9.1617XXX [Called Party Num Type = Nation] -- RP \+1[2-9]xx[2-9]xx -- rg-local-gw It doesn't make any different when I tried to set the Type at the RP or TP. Also, the Calling Party Num Type is Unknown as well, even though, the 5XXX Calling Party Xform Pattern set to National Any suggestions would be apppricated. Thanks, Tam May 28 16:36:24.854: ISDN Se0/2/0:23 Q931: TX - SETUP pd = 8 callref = 0x0090 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98383 Exclusive, Channel 3 Display i = 'HQ-PHN1' Calling Party Number i = 0x0081, '+12123945001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '16178632683' Plan:Unknown, Type:Unknown May 28 16:36:24.878: ISDN Se0/2/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x8090 Channel ID i = 0xA98383 Exclusive, Channel 3 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question
Hi Matthew: Did you set calling name and epnm at line from the RDP? (Not from the phone) Date: Tue, 25 May 2010 20:11:49 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 4 Section 3.1 - Mobile Connect Question Fellow nerds, I am battling a single number reach (i.e. Mobile Connect) question on Lab 4. Question 3.1 says the call should appear to BR1 Phone 2 as if it is actually coming from HQ Phone 2 directly (Calling Name and Number). When I call in from the PSTN phone to BR1 Phone 2, the display on BR1 Phone 2 shows 5002. The calling number is represented just fine. However, I cannot get the calling nmae to be presented on the display. I have tinkered around with the partial/complete match and significant digits parameters under the mobility section of the Call Manager service parameters but nothing has changed. Any ideas? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail
Hi: You can try this: Add a cue aa with a dummy ext for example 2999, then at this aa just transfer to 2904 vm, then at 2905 phone add a speed dial to 2999 with a label like 2904 VM, this way when 2905 wants to transfer to 2904 vm the user should press transfer + speed dial + transfer This should work, let us know From: siddas...@gmail.com To: ccie_voice@onlinestudylist.com Date: Wed, 26 May 2010 10:19:32 +0100 Subject: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail Hello all, One of my customer is interested in direct transfer of an incoming call to voicemail after it has been picked up by someone else in the pickup group. For e.g. If a call comes in and ring x2904 but he is not available, person at x2905 picks up the call but the calling party wants to leave a VM for x2904. How the person at x2905 can direct transfer the call to x2904 voicemail. One way is to transfer the call back to x2904 which will ring and ring for 10s and then go to voicemail. This is not what they want. They want the ability to transfer the call directly to voicemail of Original called party. ephone-dn 1 octo-line number 2904 pickup-group 1 label Tim Flynn (2904) name Tim Flynn call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! ephone-dn 2 octo-line number 2905 pickup-group 1 label Steve Zander (2905) name Steve Zander call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! Ash _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail
For all these extension wouldn't be scalable... I think that this behaviour could be changed system wyde but I can't remember how From: siddas...@gmail.com To: r.ochi...@mfient.com Date: Wed, 26 May 2010 12:03:46 +0100 CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail So you mean in CUE, I just have to assign alternate extension for every user starting with 6 like 62904, 62905, 62906 ... If x2905 transfer the call to 62904, would it go straight to VM for 2904 or will it first ring for 10s and then go to voicemail? Do I have to create ephone-dn for all of these? (remember customer has 100+ users and dn) Thanks for your help Ash From: Rogers Ochieng [mailto:r.ochi...@mfient.com] Sent: 26 May 2010 10:57 To: 'Ashar Siddiqui' Subject: RE: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail I’m thinking secondary number in CUE for the user say 62904 and you route that to CUE so 6 can be your assumed prefix for diverting calls to CUE for other subscriber From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Wednesday, May 26, 2010 12:20 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Direct transfer to Original called party Voicemail Hello all, One of my customer is interested in direct transfer of an incoming call to voicemail after it has been picked up by someone else in the pickup group. For e.g. If a call comes in and ring x2904 but he is not available, person at x2905 picks up the call but the calling party wants to leave a VM for x2904. How the person at x2905 can direct transfer the call to x2904 voicemail. One way is to transfer the call back to x2904 which will ring and ring for 10s and then go to voicemail. This is not what they want. They want the ability to transfer the call directly to voicemail of Original called party. ephone-dn 1 octo-line number 2904 pickup-group 1 label Tim Flynn (2904) name Tim Flynn call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! ephone-dn 2 octo-line number 2905 pickup-group 1 label Steve Zander (2905) name Steve Zander call-forward busy 8005 call-forward noan 8005 timeout 10 corlist incoming User-international ! ! Ash _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Attendat console
Hi: Yesterday I was trying to create an AC pilot point, but the pilot didn't get registered, I created a user with cti rights and with phones controlled, then I configured this user as the ac user... I tried to reset the ac service but with no luck Any other suggestion? Thanks _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Attendat console
Thanks to everybody, user ac 12345, cti call park monit, cti allow control of all devi and reseting ac service made the trick regards Date: Tue, 25 May 2010 06:29:16 -0700 Subject: Re: [OSL | CCIE_Voice] Attendat console From: cristobalpri...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com have you added the ac application user to the Standard CTI Allow Call Park Monitoring group? 2010/5/25 Angel Perez gorr...@hotmail.com Hi: Yesterday I was trying to create an AC pilot point, but the pilot didn't get registered, I created a user with cti rights and with phones controlled, then I configured this user as the ac user... I tried to reset the ac service but with no luck Any other suggestion? Thanks Hotmail: Powerful Free email with security by Microsoft. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME - direct incoming call from PSTN
Hi: You can use incoming called number 7771234 at dial-peer range 7771000-7771005 and 7771235 at dial-peer range 7771006-77710010 hth Date: Mon, 24 May 2010 16:59:08 +1000 From: vip...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CME - direct incoming call from PSTN Hi Guys, Is there any way to direct specific incoming call from PSTN to a specific dial-peer range number on CME using COR? - 3 CMEs, only 1 cme connected to PSTN. for example, public phone number 7771234 that originate from PSTN only allow to ring dial-peer range 7771000-7771005 on CME-A then public phone number 7771235, only allow to ring dial-peer range 7771006-7771010 on CME-C. please advice Thanks in advance ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Device Mobility
Hi: Dev Mob is working as expected (the phone is taken romain sensitive setting from the roaming device pool) but when I click on View Current Device Mobility Settings link the roaming settings are not shown at the pop up windows... but the phones has the correct settings... The same thing happend to me in the past but I'm not sure how did I manage to solve it Any clues? thx _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CME-CUE VOICEVIEW
Hi all: I've the following problem with voiceview and CME: I've sucsesfully configure the voiceview service for phones, I access the service (no pin asked) but once I see the menu options (1 inbox, 2 Send Messages, 3 etc) I can't select any of the options neither logout with Logout button, I've follow these steps but with no luck http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/7vview.html Here is my config per the above guide: CME: telephony-service url services http://CUE-hostname/voiceview/common/login.do url authentication http://cme-ip-address/CCMCIP/authenticate.asp authentication credential Admin cisco CUE: site name local phone-authentication Admin cisco end site cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Am I missing something? Regards _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW
Hi: Wich is the correct one? This one?: url authentication http://cue/voiceview/authentication/authenticate.do thx Date: Thu, 20 May 2010 18:01:55 +0530 Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW From: voip.ccieci...@gmail.com To: gorr...@hotmail.com You have wrong authentication URL On Thu, May 20, 2010 at 4:51 PM, Angel Perez gorr...@hotmail.com wrote: Hi all: I've the following problem with voiceview and CME: I've sucsesfully configure the voiceview service for phones, I access the service (no pin asked) but once I see the menu options (1 inbox, 2 Send Messages, 3 etc) I can't select any of the options neither logout with Logout button, I've follow these steps but with no luck http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/7vview.html Here is my config per the above guide: CME: telephony-service url services http://CUE-hostname/voiceview/common/login.do url authentication http://cme-ip-address/CCMCIP/authenticate.asp authentication credential Admin cisco CUE: site name local phone-authentication Admin cisco end site cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Am I missing something? Regards Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW [SOLVED]
Hi: The problem was a combination of to things: 1: At cue I had login pinless configured, no that I've deleted this command from cue's config, I can press voiceview service button and the pin is prompted. Then the menus are working properly. 2: Once that menus are working, when I try to listen a mail (open a rtp session) I get an error about athentication, the solution is to change the athentication url (thanks cisco voip) to: telephony-service url authentication http://142.102.66.253/voiceview/authentication/authenticate.do Very important reset the phones at this point!!! You can verify with: cue# sh voiceview sessions Mailbox RTP User ID Phone MAC Address 1001Yes user10017.E066. ! This user is listen a message 1002No user20017.E066. If you can remember the whole sintax try to remember this one: cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do cue# show phone-authentication configuration Authentication service URL: http://CUE-hostname/voiceview/authentication/authenticate.do Authentication Fallback Server URL: You will get the urls without waisting time with doc cd A usefull link: http://www.ccievoicestudy.com/Cisco/VoIP/Enabling_CUE_VoiceView_Express_for_CME/ But I've a last question: Is it possible to make this work with login pinless enable on cue? Thanks for the comments Date: Thu, 20 May 2010 19:03:19 +0530 Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Yeah http://cue/voiceview/authentication/authenticate.do is correct On Thu, May 20, 2010 at 6:51 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Wich is the correct one? This one?: url authentication http://cue/voiceview/authentication/authenticate.do thx Date: Thu, 20 May 2010 18:01:55 +0530 Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW From: voip.ccieci...@gmail.com To: gorr...@hotmail.com You have wrong authentication URL On Thu, May 20, 2010 at 4:51 PM, Angel Perez gorr...@hotmail.com wrote: Hi all: I've the following problem with voiceview and CME: I've sucsesfully configure the voiceview service for phones, I access the service (no pin asked) but once I see the menu options (1 inbox, 2 Send Messages, 3 etc) I can't select any of the options neither logout with Logout button, I've follow these steps but with no luck http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/7vview.html Here is my config per the above guide: CME: telephony-service url services http://CUE-hostname/voiceview/common/login.do url authentication http://cme-ip-address/CCMCIP/authenticate.asp authentication credential Admin cisco CUE: site name local phone-authentication Admin cisco end site cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Am I missing something? Regards Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW [SOLVED]
I tried to make this working with login pinless using a combination of: telephony-service authentication credential user2 1234 authentication credential 1002 1234 and cue site name local phone-authentication username user2 password 1234 ! also phone-authentication username 1002 password 1234! 1002 is the extension end site with no luck, anybody has been able to make this work with login pinless enable on cue? Thanks From: gorr...@hotmail.com To: voip.ccieci...@gmail.com Date: Thu, 20 May 2010 14:24:52 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW [SOLVED] Hi: The problem was a combination of to things: 1: At cue I had login pinless configured, no that I've deleted this command from cue's config, I can press voiceview service button and the pin is prompted. Then the menus are working properly. 2: Once that menus are working, when I try to listen a mail (open a rtp session) I get an error about athentication, the solution is to change the athentication url (thanks cisco voip) to: telephony-service url authentication http://142.102.66.253/voiceview/authentication/authenticate.do Very important reset the phones at this point!!! You can verify with: cue# sh voiceview sessions Mailbox RTP User ID Phone MAC Address 1001Yes user10017.E066. ! This user is listen a message 1002No user20017.E066. If you can remember the whole sintax try to remember this one: cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do cue# show phone-authentication configuration Authentication service URL: http://CUE-hostname/voiceview/authentication/authenticate.do Authentication Fallback Server URL: You will get the urls without waisting time with doc cd A usefull link: http://www.ccievoicestudy.com/Cisco/VoIP/Enabling_CUE_VoiceView_Express_for_CME/ But I've a last question: Is it possible to make this work with login pinless enable on cue? Thanks for the comments Date: Thu, 20 May 2010 19:03:19 +0530 Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW From: voip.ccieci...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Yeah http://cue/voiceview/authentication/authenticate.do is correct On Thu, May 20, 2010 at 6:51 PM, Angel Perez gorr...@hotmail.com wrote: Hi: Wich is the correct one? This one?: url authentication http://cue/voiceview/authentication/authenticate.do thx Date: Thu, 20 May 2010 18:01:55 +0530 Subject: Re: [OSL | CCIE_Voice] CME-CUE VOICEVIEW From: voip.ccieci...@gmail.com To: gorr...@hotmail.com You have wrong authentication URL On Thu, May 20, 2010 at 4:51 PM, Angel Perez gorr...@hotmail.com wrote: Hi all: I've the following problem with voiceview and CME: I've sucsesfully configure the voiceview service for phones, I access the service (no pin asked) but once I see the menu options (1 inbox, 2 Send Messages, 3 etc) I can't select any of the options neither logout with Logout button, I've follow these steps but with no luck http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/rel3_2/administrator/voicemail/7vview.html Here is my config per the above guide: CME: telephony-service url services http://CUE-hostname/voiceview/common/login.do url authentication http://cme-ip-address/CCMCIP/authenticate.asp authentication credential Admin cisco CUE: site name local phone-authentication Admin cisco end site cue# show voiceview configuration Phone service URL: http://CUE-hostname/voiceview/common/login.do Enabled: Yes Idle Timeout (minutes): 5 Am I missing something? Regards Hotmail: Free, trusted and rich email service. Get it now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. Hotmail: Free, trusted and rich email service. Get it now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cannot dial from MVA
Hi, check this topic: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16572.html hth From: wormh...@sch.hu To: ccie_voice@onlinestudylist.com Date: Tue, 18 May 2010 20:24:30 +0200 Subject: [OSL | CCIE_Voice] cannot dial from MVA Gents, I have an issue with MVA. MVA collects PIN and I press 1 to dial but it does not proceed with any call instead the well known prompt sounds: The call cannot be completed... Even if the called number is local and placed in the None partition. This prompt suggests CSS issue however as Vik advised before I created a totally new CSS just for RDP but it does not solve the problem. Service Parameters: Complete Match and RDP+Line CSS. I have read near all the thread regarding MVA here, but the issue remains. I attached the vxml debug. Any suggestion? _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2 Question 2.2 - CAC Locations
Hi, the bandwith assigned to a location affects incoming and outgoing call to/from this location, this is way is only valid for a hub and spoke topology (all the calls go throw the hub). If your topology is not hub and spoke you should use rsvp wich match one to one locations (you specify reservation from one location to another one) hth From: martybeut...@hotmail.com To: ciscovoiceg...@gmail.com; ccie_voice@onlinestudylist.com Date: Tue, 18 May 2010 22:59:50 -0500 Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 2 Question 2.2 - CAC Locations Hey Matthew, Locations based CAC is suitable for a Hub-and-Spoke topology. The recommendation would be to use the Hub-none location (with unlimited bandwidth) at the Hub site (HQ). You would only create new locations for the Spoke sites, and specify the appropriate bandwidth for the site. Your assignment of 96K to the BR1 location would be correct. Hope that helps Date: Tue, 18 May 2010 22:27:55 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Vol 2 Lab 2 Question 2.2 - CAC Locations Still trying to understand how Hub_none plays into the mixed when locations-based CAC is operational. The question asks for a maximum of four calls between devices registered at the HQ and BR1 sites. That seems to imply setting 96 kbps on LOC-BR1 and LOC-HQ. Using a location for HQ would require changing the Hub_none references to LOC-HQ. Am I correct in my understanding? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Vol 2 Lab 2 Question 2.2 - CAC Locations
Hi Matthew: If you set br1 device pool and hub-none loc at gw settings and this device pool has location br1, in this case, dp general configuration will overwrite gw specific configuration, hub-none location is an exception to the general rule. Check the first 3 paragraphs of this post: http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16570.html hth Date: Wed, 19 May 2010 05:46:15 -0500 From: ciscovoiceg...@gmail.com To: gorr...@hotmail.com CC: martybeut...@hotmail.com; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Vol 2 Lab 2 Question 2.2 - CAC Locations Good explanations, Angel and Marty. In the Proctor Guide, I noticed that the CUCM GUI configuration for the gateways did not specify custom locations. For example, the MGCP gateway for BR1 was configured for Hub_none. My only explanation is that the wording implied phones not gateways - Between devices at HQ and BR1. However, technically, a registered gateway would be a device. The downside to setting locations-based CAC on the gateway would be the limitation it'd impose on inbound PSTN calls to phones in HQ (Hub_none). Thoughts? -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE CLI
Hi, start with this :) http://pushkarbhatkoti.wordpress.com/category/cue-voicemail-vpim-networking-cue-to-unity-in-10-minutes/ http://www.brainbump.net/2009/04/easy-approach-for-configuring-and-setting-up-cisco-unity-express/#more-503 http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/administrator/AA_and_VM/guide/vmadmin_book.html Date: Wed, 19 May 2010 08:58:20 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUE CLI All - Are there any good documents out there on how to configure CUE using CLI? I'd like to use this approach to reduce time, but I haven't been able to find a good resource for reference. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
Hi: Add the following: dspfarm profile 1 trans shut codec g729r8 no shut By default g729r8 is not configured Let us know Date: Tue, 18 May 2010 19:21:24 +0300 Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM From: waelag...@gmail.com To: gorr...@hotmail.com Hi Angel, Below is the output BR2#sh sdspfarm units mtp-1 Device:XCODE_2 TCP socket:[3] REGISTERED in SCCP ver 17/10 actual_stream:0 max_stream 2 IP:10.10.202.1 51447 MTP Dixieland keepalive 271 Supported codec: G711Ulaw G711Alaw G729 G729a G729ab conf-2 Device:conference TCP socket:[1] REGISTERED in SCCP ver 17/10 actual_stream:16 max_stream 16 IP:10.10.202.1 12196 Conference Dixieland keepalive 271 Supported codec: G711Ulaw G711Alaw G729 G729a G729b G729ab max-mtps:2, max-streams:0, alloc-streams:0, act-streams:0 BR2#sh dspfarm all Dspfarm Profile Configuration Profile ID = 1, Service = TRANSCODING, Resource ID = 1 Profile Description : Profile Service Mode : Non Secure Profile Admin State : UP Profile Operation State : ACTIVE Application : SCCP Status : ASSOCIATED Resource Provider : FLEX_DSPRM Status : UP Number of Resource Configured : 1 Number of Resource Available : 1 Codec Configuration Codec : g711ulaw, Maximum Packetization Period : 30 Codec : g711alaw, Maximum Packetization Period : 30 Codec : g729ar8, Maximum Packetization Period : 60 Codec : g729abr8, Maximum Packetization Period : 60 Codec : g729r8, Maximum Packetization Period : 60 Dspfarm Profile Configuration _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE Voice Schedule
Yes it's possible I made it From: naoufal.kerbo...@cbi.ma To: akashapa...@yahoo.com Date: Mon, 17 May 2010 23:55:49 + CC: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE Voice Schedule 90 days before the exam date if you Will pay by wire transfer, you CAN schedule and pay by Visa card, it's better Envoyé de mon iPhone Le 17 mai 2010 à 22:10, akash patel akashapa...@yahoo.com a écrit : I am planning to take lab in couple months. I called Cisco Support and they told me that you can't schedule your exam less than 90 days ago. Does anyone know if there is a workaround and schedule the lab whenever the seats are available. Thank you ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
Hi: Make sure that you have all the neccesary commands under telephony-service max-dn max-phone source add Of course you also need sdspfarm related commands you may already have hth Date: Tue, 18 May 2010 19:42:29 +0300 Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM From: waelag...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Hi It is already added, G729r8, however i did it to reset, but same issue as below: //66/9FB18C6DADAC/SIP/Error/sipSPI_ipip_copy_channelInfo_to_sdp: filter mis-match, failing call May 18 16:59:58.874: //-1//SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!! May 18 16:59:58.874: //-1//SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!! May 18 16:59:58.874: //66/9FB18C6DADAC/SIP/Error/sipSPIAddSDPMediaPayload: Call Origination Failed: None of the selected codec from CLI is supported by SIP May 18 16:59:58.874: //66/9FB18C6DADAC/SIP/Error/sipSPIOutgoingCallSDP: Error with codec types on media line : 1 May 18 16:59:58.874: //66/9FB18C6DADAC/SIP/Error/sipSPICreateOutboundSDP: Error in creating an SDP for the outbound call - Check for supported codecs May 18 16:59:58.874: //66/9FB18C6DADAC/SIP/Error/preprocessSetup: Error during outbound SDP creation May 18 16:59:58.874: //-1//SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!! May 18 16:59:58.874: //-1//SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!! May 18 16:59:58.874: //-1//SIP/Error/ccsip_spi_process_ccapi_event: CCAPI Event Preprocessor Failure _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
Hi add this under telephony service max-dn 1 max-phone 1 ip source add 10.10.202.1 hth Date: Tue, 18 May 2010 19:52:42 +0300 From: waelag...@gmail.com To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM All config seems fine: voice register global mode cme source-address 10.10.202.1 port 5060 max-dn 2 max-pool 2 _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM
Ummm, did you add sip bind all source interface ??? Date: Tue, 18 May 2010 20:01:06 +0300 Subject: Re: [OSL | CCIE_Voice] CUBE Issue - Vol2 Lab 8 - BR2 Calls to CUCM From: waelag...@gmail.com To: gorr...@hotmail.com CC: ccie_voice@onlinestudylist.com Done, but still the same :( BR2(config)#telephony-service BR2(config-telephony)#max-dn 25 BR2(config-telephony)#max-ephones 33 BR2(config-telephony)#ip source-address 10.10.202.1 Updating CNF files CNF files updating complete BR2(config-telephony)# BR2(config-telephony)#^Z BR2# BR2# BR2#term mon BR2#sh deb BR2#sh debugging CCSIP SPI: SIP error debug tracing is enabled (filter is OFF) BR2# SIP: (69) Attribute mid, level 1 instance 1 not found. May 18 17:18:18.416: //69/2F9E1026ADB5/SIP/Error/sipSPI_ipip_copy_sdp_to_channelInfo: failed to update call entry May 18 17:18:19.568: //70/2F9E1026ADB5/SIP/Error/sipSPI_ipip_copy_channelInfo_to_sdp: filter mis-match, failing call May 18 17:18:19.568: //-1//SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!! May 18 17:18:19.568: //-1//SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!! May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPIAddSDPMediaPayload: Call Origination Failed: None of the selected codec from CLI is supported by SIP May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPIOutgoingCallSDP: Error with codec types on media line : 1 May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/sipSPICreateOutboundSDP: Error in creating an SDP for the outbound call - Check for supported codecs May 18 17:18:19.572: //70/2F9E1026ADB5/SIP/Error/preprocessSetup: Error during outbound SDP creation May 18 17:18:19.572: //-1//SIP/Error/sipSPIGetContentQSIG: No Inbound Container Created !!! May 18 17:18:19.572: //-1//SIP/Error/sipSPIGetContentQ931: No Inbound Container Created !!! May 18 17:18:19.572: //-1//SIP/Error/ccsip_spi_process_ccapi_event: CCAPI Event Preprocessor Failure BR2# _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM DB Replication Issues
Hi Matthew: The following procedure will show you if replication is broken, the key is search is both databases, pub and sub: (I assume that subs is the primary, pub is the secondary and also that you have enable voice media streaming app service in ucm) Go to UCM MEDIA RESOURCES ANNUNCIATOR Select ANN_3 and change its name to something different for example ANN_SUBS, save and reset Go to pub CLI, also to sub CLI and use this command on bouth: run sql select name from device You will see all devices into your database, something like this: name = MTP_2 CFB_2 ANN_2 ANN_SUBS MOH_3 SEP001906DC4E1D SEP0017E032F90D If the ANN name in sub database isn't the new name you have just set, this means that you have a db replication problem 100% sure If db replicatino is broken you will also notice that the ANN wich you changed the name will be unregistered or rejected in UCM GUI If both names are the same your replication is working hth Date: Sun, 16 May 2010 21:07:07 -0500 From: ciscovoiceg...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] CUCM DB Replication Issues Can someone explain this to me? Below is my output from show dbreplcation status after I issued a utils dbreplication repair all and reboot the Pub then Sub. I always though a status of 3 meant replication was broken, but the explanation below says that no errors or mismatches were found. This seems off to me. == Please use file view activelog cm/trace/dbl/sdi/ReplicationStatus.2010_05_17_02_02_43.out command to see the output admin:file view activelog cm/trace/dbl/sdi/ReplicationStatus.2010_05_17_02_02_43.out SERVER ID STATESTATUS QUEUE CONNECTION CHANGED --- g_ucmpub_ccm7_0_1_11000_22 Active Local 0 g_ucmsub_ccm7_0_1_11000_23 Active Connected 0 May 17 01:56:20 - No Errors or Mismatches found. Replication status is good on all available servers. -- Matthew Berry A+, CCENT, CCNA, CCNA Voice, CCVP, CCIE Voice Written Vitals: GVoice: +1.612.424.5044 Gmail: ciscovoiceg...@gmail.com Skype: ciscovoiceguru Twitter: ciscovoiceguru Cert Stats: Cisco Cert Journey Began: Jan 1, 2009 1st Lab Attempt: Aug 16, 2010 _ Hotmail: Free, trusted and rich email service. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Physical Components for CCIE Voice Lab
Hi: 4 gateways: 2801 for hq, 2801 for br1, 2811 for br2 (you need a NM cue or 2801 with AIM CUE), 2811 for pstn + wic cards + t1/e1 cards + pvdms + cue + hwic-4esw + 3750 + 5-6 ip phones + cables Plus a quad core with 8/16 gb mem hth From: amuno...@hotmail.com To: ccie_voice@onlinestudylist.com Date: Sat, 15 May 2010 14:28:49 -0500 Subject: [OSL | CCIE_Voice] Physical Components for CCIE Voice Lab Hello, I am recently passed the CCIE Voice Written, then I am so excited for going on with the CCIE Voice Lab. The question that I have for yours, what physical components such as router should I buy for preparing for the Lab??? I have thought in buying the following: · router C2901-CME-SRST/K9, included Unity Express base release 8.0 · Two ip phones 7942G · Server for virtualization of CUCM (Pub + Sub), UC, UCCX, UPS, WinXP for IP Communicator. What could you suggest me for preparing for the Lab??when I feel that I am ready, I will take a bootcamp with IPExpert. I would appreciate your help and experience in this case, I want to start well since the beginning. Best regards, Alexis Munoz CCNP, CCVP, PMP _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] HW Conf Bridge Problem on Lab7 Vol1
Hi: Try to set unlimet bandwith from/to this location if this works check the following¨ http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg16570.html hth Date: Sun, 16 May 2010 00:20:08 +0100 From: naoufal.kerbo...@cbi.ma To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] HW Conf Bridge Problem on Lab7 Vol1 Hi, I'm working on lab 7 Vol 1 question 7.3 and ran into an issue with BR1. BR1 Conf bridge is registred to CUCM (No problem) and add it to MRGL to BR1 DP. scenario: HQ-Phone2 BR1Phone2 CONF-TO PSTN If I try to join the conference call from HQ Phone dropped and get on BR1 phone screen Cannot complete conference and call to pstn still. My BR1 config: sccp local Vlan240 sccp ccm 10.10.210.10 identifier 2 version 5.0.1 sccp ccm 10.10.210.11 identifier 1 version 5.0.1 sccp ! sccp ccm group 1 bind interface Vlan240 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 3 register br1Conf ! dspfarm profile 3 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 3 associate application SCCP Any Ideas? Regards Naoufal ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] conference bridge and locations
Hi Roger: I guess that setting location at cnf bridge to the specific location will do the trick Let us know Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Thu, 13 May 2010 20:58:52 +0200 Subject: SV: [OSL | CCIE_Voice] conference bridge and locations Hi Angel, I had a similar experience as you describe it when I did lab 7 vol 2 today, never figured it out fully before I ran out of time tough. But I'll plan to do the same lab tomorrow morning, I'll try your settings out and let you know if that solved my problem. Best regards Roger Källberg Consultant Cygate AB Från: Angel Perez [gorr...@hotmail.com] Skickat: den 13 maj 2010 13:45 Till: osl osl Ämne: [OSL | CCIE_Voice] conference bridge and locations Hi all: In UCM there is a general rule that says that most specific settings overwrite general settings for example mrgl in a phone has preference over mrgl configured at the device pool of this phone There are some exceptions to this rule, for example a roaming device with device mob enable, in this case device pool settings (of the roaming device pool) have preference over phone settings. Another exception to this rule is hub-none location, in this case device pool location takes precedence over phone location if the phone is configured with hub-none loc (if the phone has another location different from hub-none the general rule is applied) With this in mind I was setting a hw conference between some devices all of them in the same dp br1 gw: br1 device pool, hub-none location br1 ph1: br1 device pool, hub-none location br1 ph2: br1 device pool, hub-none location br1 conference bridge: br1 device pool, hub-none location br1 device pool: br1 location, br1 region br1 location: Audio bandwith 48 br1 region: g711 with br1 and g729 with hq and br2 With this config the br1 device pool location (br1) should overwrite br1 conf bridge location (hub-none) But when I initiate a conferen between br1 ph1, br1 ph2 and a pstn call (ingressing from br1 gw), i get Cannot complete conference, then if I set audio bandwith at br1 location to unlimited conference is completing... Then when I initiate the same conference and I set br1 conference bridge to: br1 device pool, br1 location the conference complete The conclusion is that for UCM the br1 conference bridge is at hub-none location not at br1 location, but device pool location should overwrite conf bridge location following the logic explained above Anybody has an explantion for this behaviour??? Thanks Hotmail: Trusted email with powerful SPAM protection. Sign up now. _ Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] DISA keeping me up!!!!
Hi: I've seen this before with two different scenarios, both of them without a clear logic behind: Case 1: Check that the mva number is in a partition that the rdp css can see (if you set the service parameter to Inbound destination profile + line CSS you have to check this becouse the normal situation is to use inbound gw/trunk) . There is a strange behaviour with this sometimes you can call the mva number without a css that can do it, but later when you try to call the desired number you enter in a loop Case 2: Are you hiting the rdp number with partial match or total mach? I've heard from someone in the forum that there is bug with partial match, also I've experience weird issues with it (simial to what you describe), my advice is to set to total match, and apply the proper translation rule at h323 gw to accomplish this. As you can see there are some strange behaviour with mva, in my oppinion is a bug, but I haven't seen any Cisco doc about it Let us know how it goes Thanks Date: Fri, 14 May 2010 01:30:12 -0400 From: shurric...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] DISA keeping me up I just can't seem to be able to call anywhere while logged in to MVA.i can successfully authenticateit recognizes my remote destination but when i enter the number to call the message just loops and asks me to enter my pin again.anyone seen this before...i am using parameter Inbound destination profile + line CSS and confirmed that the CSS on the RDP can call the destination i want to callbut the message just keeps looping and it eventually disconnects .. any thoughts? _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] conference bridge and locations
Hi Roger: Glad to hear that it's working, there are a lot of small weird issues with ucm 7.0 regards From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Fri, 14 May 2010 13:55:23 +0200 Subject: SV: SV: [OSL | CCIE_Voice] conference bridge and locations Update. You were totaly right Angel, after I set the location on the BR2 cfb all worked as expected. I guess we have to add that to the funny feature list to remember. Roger Källberg Consultant Cygate AB Från: Angel Perez [gorr...@hotmail.com] Skickat: den 14 maj 2010 10:44 Till: Roger Källberg; osl osl Ämne: RE: SV: [OSL | CCIE_Voice] conference bridge and locations Hi Roger: I guess that setting location at cnf bridge to the specific location will do the trick Let us know Thanks From: roger.kallb...@cygate.se To: gorr...@hotmail.com; ccie_voice@onlinestudylist.com Date: Thu, 13 May 2010 20:58:52 +0200 Subject: SV: [OSL | CCIE_Voice] conference bridge and locations Hi Angel, I had a similar experience as you describe it when I did lab 7 vol 2 today, never figured it out fully before I ran out of time tough. But I'll plan to do the same lab tomorrow morning, I'll try your settings out and let you know if that solved my problem. Best regards Roger Källberg Consultant Cygate AB Från: Angel Perez [gorr...@hotmail.com] Skickat: den 13 maj 2010 13:45 Till: osl osl Ämne: [OSL | CCIE_Voice] conference bridge and locations Hi all: In UCM there is a general rule that says that most specific settings overwrite general settings for example mrgl in a phone has preference over mrgl configured at the device pool of this phone There are some exceptions to this rule, for example a roaming device with device mob enable, in this case device pool settings (of the roaming device pool) have preference over phone settings. Another exception to this rule is hub-none location, in this case device pool location takes precedence over phone location if the phone is configured with hub-none loc (if the phone has another location different from hub-none the general rule is applied) With this in mind I was setting a hw conference between some devices all of them in the same dp br1 gw: br1 device pool, hub-none location br1 ph1: br1 device pool, hub-none location br1 ph2: br1 device pool, hub-none location br1 conference bridge: br1 device pool, hub-none location br1 device pool: br1 location, br1 region br1 location: Audio bandwith 48 br1 region: g711 with br1 and g729 with hq and br2 With this config the br1 device pool location (br1) should overwrite br1 conf bridge location (hub-none) But when I initiate a conferen between br1 ph1, br1 ph2 and a pstn call (ingressing from br1 gw), i get Cannot complete conference, then if I set audio bandwith at br1 location to unlimited conference is completing... Then when I initiate the same conference and I set br1 conference bridge to: br1 device pool, br1 location the conference complete The conclusion is that for UCM the br1 conference bridge is at hub-none location not at br1 location, but device pool location should overwrite conf bridge location following the logic explained above Anybody has an explantion for this behaviour??? Thanks Hotmail: Trusted email with powerful SPAM protection. Sign up now. Your E-mail and More On-the-Go. Get Windows Live Hotmail Free. Sign up now. _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] conference bridge and locations
Hi all: In UCM there is a general rule that says that most specific settings overwrite general settings for example mrgl in a phone has preference over mrgl configured at the device pool of this phone There are some exceptions to this rule, for example a roaming device with device mob enable, in this case device pool settings (of the roaming device pool) have preference over phone settings. Another exception to this rule is hub-none location, in this case device pool location takes precedence over phone location if the phone is configured with hub-none loc (if the phone has another location different from hub-none the general rule is applied) With this in mind I was setting a hw conference between some devices all of them in the same dp br1 gw: br1 device pool, hub-none location br1 ph1: br1 device pool, hub-none location br1 ph2: br1 device pool, hub-none location br1 conference bridge: br1 device pool, hub-none location br1 device pool: br1 location, br1 region br1 location: Audio bandwith 48 br1 region: g711 with br1 and g729 with hq and br2 With this config the br1 device pool location (br1) should overwrite br1 conf bridge location (hub-none) But when I initiate a conferen between br1 ph1, br1 ph2 and a pstn call (ingressing from br1 gw), i get Cannot complete conference, then if I set audio bandwith at br1 location to unlimited conference is completing... Then when I initiate the same conference and I set br1 conference bridge to: br1 device pool, br1 location the conference complete The conclusion is that for UCM the br1 conference bridge is at hub-none location not at br1 location, but device pool location should overwrite conf bridge location following the logic explained above Anybody has an explantion for this behaviour??? Thanks _ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UCM strange behaviour
Hello: I've the following scenario: route pattern- route list (h323 gw, mgcp gw) 1: route pattern - h323 gw At route pattern level I've strip predot, so at route list I don't need to strip it, everything works as expected, (the call arrives at the gw with predot striped) 2: route pattern - mgcp gw (backup) When I use the second option (mgcp gw) of the route list, cucm isn't striping predot at route pattern level, so I've to strip it again at route list level. It's strange becouse although the ucm is not striping predot at route pattern I can see the called number at phone display TO where is the called number with predot striped... IMHO this is not the normal situation, correct me if I'm wrong Thanks in advance _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UCM strange behaviour
Hi again and good morning: Doing some more test the same happend with another route list First example route list was [br1-gw-h323 (striping at rp level) , hq-gw-mgcp (not striping at rp level)] New route list created is [br2-gw h323 (striping at rp level) , br1-gw-h323 (not striping at rp level)] Also I've tried to delete the first gw of the new route list but still not striping at rp level, but then when I copy the route list (only with one gw) with the exact config and applied to route pattern, ucm is striping the predot at rp... So it looks like it only happend with the second option of the route list, and it is not related with the gw itself This looks like a bug but can't find anyone related with this... Anybody have seen this behaviour before? From: ciscovoiceg...@gmail.com To: gorr...@hotmail.com Subject: Re: [OSL | CCIE_Voice] UCM strange behaviour Date: Wed, 12 May 2010 07:11:01 -0700 CC: ccie_voice@onlinestudylist.com From my understanding, route list trumps route pattern when digit manipulation is concerned. However, route pattern manipulations, even though trumped in the end, will affect what is displayed on the phone itself. I'm in line at Starbucks so I can't double-check my notes. Double-check for yourself. :) Matthew Berry **Sent from my iPhone** Skype/Twitter: ciscovoiceguru Google Voice: +1 612 424 5044 On May 12, 2010, at 6:53 AM, Angel Perez gorr...@hotmail.com wrote: Hello: I've the following scenario: route pattern- route list (h323 gw, mgcp gw) 1: route pattern - h323 gw At route pattern level I've strip predot, so at route list I don't need to strip it, everything works as expected, (the call arrives at the gw with predot striped) 2: route pattern - mgcp gw (backup) When I use the second option (mgcp gw) of the route list, cucm isn't striping predot at route pattern level, so I've to strip it again at route list level. It's strange becouse although the ucm is not striping predot at route pattern I can see the called number at phone display TO where is the called number with predot striped... IMHO this is not the normal situation, correct me if I'm wrong Thanks in advance Hotmail: Trusted email with Microsoft’s powerful SPAM protection. Sign up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
