Re: [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script
As for the original question, it should be possible to return a label to whatever target on UCM in the ICM script. Am I missing something non trivial? On Apr 6, 2014 1:19 PM, Chrysostomos Christofi ch.christ...@logicom.net wrote: Hi If you have CUE you can achieve this task in the same way as into UCCX #Chrysostomos *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *CCIEing *Sent:* Sunday, April 6, 2014 9:28 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] UCCE Script: Transfer Call to IP Phone UCCE Script Dear Group Members, I would to ask about this feature ..After a time check (if the time after mid night) I have a UCCE system and I need to create a script that allow the call to be transferred to an IP Phone (Not Agent) just a number or PSTN number , the most important is This is not agent. As you may all know this is easy from the UCCX , but is that doable from UCCE ? Thanks ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] Disabling almost all mailboxes
Here is a simpler solution. Create a new partition on unity and move all the users you want disabled there. Don't add it to any search space and you are done On Mar 6, 2014 4:15 PM, Christian Holst c...@netdesign.dk wrote: If people are not able to leave voice messages (no forward) then MWI won't be turned on. If it's on already people should listen to their voicemails :) - which will turn it off In the odd case it's on by accident force it off. regards Christian Holst -Original Message- From: Isamar Maia [mailto:isa...@gmail.com] Sent: 6. marts 2014 22:39 To: Christian Holst Cc: ccie_voice@onlinestudylist.com; mauri...@imtech.com.br Subject: Re: [OSL | CCIE_Voice] Disabling almost all mailboxes Is it gonna disable the MWI led as well ? Isamar 2014-03-06 17:52 GMT-03:00 Christian Holst c...@netdesign.dk: If just disable, i'd remove the forward and leave the unity users If you delete the users all calls will end up in default greeting - can't imagine a customer would be happy about that one. Regards Christian Holst System Engineer UC CCIE Voice #41370 -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] On Behalf Of Isamar Maia Sent: 6. marts 2014 21:32 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Disabling almost all mailboxes Hi folks, What is the best ways to disable most of mailboxes on Unity, leaving just some of them unchanged ? Removing the users on Unity admin GUI ? -- Isamar Maia Cel. VIVO SSA: (55) 71-9940-2012 Cel. TIM SSA: (55) 71-9289-5128 Cel. Claro SSA: (55) 71-9146-8575 Fixo: (55) 71-4062-8688 Skype ID: isamar.maia A vida é muito curta para ser pequena (Benjamin Disraeli) ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc -- Isamar Maia Cel. VIVO SSA: (55) 71-9940-2012 Cel. TIM SSA: (55) 71-9289-5128 Cel. Claro SSA: (55) 71-9146-8575 Fixo: (55) 71-4062-8688 Skype ID: isamar.maia A vida é muito curta para ser pequena (Benjamin Disraeli) ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] CTI Application Server (TCP/2748) vs JTAPI Application Server (TCP/2789)
Jtapi/tapi interfaces are northbound interfaces which are on top of the CTI layer. Taking the example of UCCX, UCCX can sync with ucm and download jtapi libraries from ccm. Its built in jtapi client uses those libraries to communicate with CTI on the ucm server. The term rmjtapi refers to the local credentials used by its jtapi client to connect to CTI. Hope that helps On Nov 2, 2013 4:08 AM, Somphol Boonjing somp...@gmail.com wrote: Could anyone help explain or refer me to the documentation that help me understand the role of JTAPI Application Server (tcp/2789) a bit more? I am interested to learn about which application server use that particular port TCP/2789? (CUC / UCCX / CUE / CUPC) I know that both CUE and CUPC (Deskphone mode) and UCCX, all of them, talk to CTI Application Server at port TCP/2748, but does JTAPI Application Serer at TCP/2789 ever get used by any of those application server/client? Note: I find it very confusing when people use rmjtapi account name (in case of UCCX) or cuejtapi (in case of CUE) to talk to CTI Application Server (TCP/2748) which really is a CTI Application Server and is not JTAPI Application (TCP/2789). REF: http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.htmlhttp://www.cisco.com/en/US/customer/docs/voice_ip_comm/cucm/port/9_1_1/CUCM_BK_T2CA6EDE_00_tcp-port-usage-guide-91_chapter_01.html http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/8_0_1/portlist801.html http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/port/7_0/CCM_7.0PortList.pdf Cisco Unified Communications App Unified CM 2748 / TCP CTI application server Cisco Unified Communications App Unified CM 2749 / TCP TLS connection between CTI applications (JTAPI/TSP) and CTIManager Cisco Unified Communications App Unified CM 2789 / TCP JTAPI application server See Also: CUPC Port Usage - http://www.cisco.com/en/US/docs/voice_ip_comm/cupc/7_1/english/release/notes/cupc71.htmlhttp://www.google.com/url?q=http%3A%2F%2Fwww.cisco.com%2Fen%2FUS%2Fcustomer%2Fdocs%2Fvoice_ip_comm%2Fcupc%2F7_1%2Fenglish%2Frelease%2Fnotes%2Fcupc71.htmlsa=Dsntz=1usg=AFrqEzczjzDW2L35ak1yNjFTQ0kPD4lofA UCCX Port Usage - http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/crs/express_7_0/configuration/guide/uccx70prtuti.pdf CUE Integration Guide that suggests TCP/2748 is used (and there is no reference to TCP/2789 at all) - http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Gatekeeper back up
Check your retry interval on device - gatekeeper On Aug 10, 2013 6:42 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: when we shut Gatekeeper, it always take time to go back up. is there any command to speed it up ? K ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CtiGwy Application User
Ctigwy user is for desk phone control. If you don't need cupc to work in desk phone control mode, no need for this user. On Jul 25, 2013 12:51 AM, Devakanth Gangavarapu devakanth2...@gmail.com wrote: Hi Cisco Presence solution is not integrated with JTAPI / TAPI It either uses SCCP or SIP It does not need CtiGwy application user Cheers Dev On Thu, Jul 25, 2013 at 4:58 AM, Barrera, Hugo hugo.barr...@nexusis.comwrote: Hi, ** ** Do you really need to add the CtiGwy Application User for the Presence section? ** ** *Hugo * ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPC
Its most likely a firewall blocking rtp. Cannot be routes as the signaling is OK (as you have ring back) On Jul 24, 2013 9:20 PM, Alex Mendoza aa.mend...@icloud.com wrote: Must check your routes Try pinging the ip phone's address from CUPC PC. If it is unsuccessful do a tracert, to see which hop do not know how to reach the voice vlan. I think is easy to figure out what is going on. Best regards Alejandro Mendoza Sent from my iPhone On 24/07/2013, at 20:12, Dharambir kumar varma dharambi...@gmail.com wrote: Hi Team. i have one phone CUPC over internet...and one cisco 7941 phone internal.. both registered to call manager. when i call from cupc to 7941 or viceversa,,ring out happens and when call is connected, only dead air/ No audio.. where can i check... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] AAR and uccx
Ram, If i remember correctly, the catch is that the AAR mask on the CTI port has to point to the RP. so your aar mask on CTI port needs to be 4000 not 400x On Jul 10, 2013 5:39 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Hi Piyush, I assigned HQ device pool and location setting on CTI route point and CTI ports to HQ. Following is isdn debug output when calling from 4002 to 4000 ( during AAR) HQ(config)# Jul 10 22:15:08.149: ISDN Se0/0/0:23 Q931: TX - SETUP pd = 8 callref = 0x000F Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98397 Exclusive, Channel 23 Calling Party Number i = 0x1181, '85224044011' Plan:ISDN, Type:International Called Party Number i = 0x91, '85224044102' Plan:ISDN, Type:International Jul 10 22:15:08.201: ISDN Se0/0/0:23 Q931: RX - CALL_PROC pd = 8 callref = 0x800F Channel ID i = 0xA98397 Exclusive, Channel 23 HQ(config)# Jul 10 22:15:08.329: ISDN Se0/0/0:23 Q931: RX - ALERTING pd = 8 callref = 0x800F Progress Ind i = 0x8088 - In-band info or appropriate now available Jul 10 22:15:08.341: ISDN Se0/0/0:23 Q931: TX - DISCONNECT pd = 8 callref = 0x000F Cause i = 0x8290 - Normal call clearing Site C phone 2 Line 2 4102 ring and call disconnect in 2 sec. with below error message Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX - SETUP pd = 8 callref = 0x0048 Sending Complete Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA9838B Exclusive, Channel 11 Display i = 'SiteC Phone 2' Calling Party Number i = 0x1183, '+85224044002' Plan:ISDN, Type:International Called Party Number i = 0x91, '85224044010' Plan:ISDN, Type:International Jul 10 22:16:48.524: ISDN Se0/0/0:15 Q931: TX - RELEASE pd = 8 callref = 0x80D5 Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - STATUS pd = 8 callref = 0x00D5 Cause i = 0x80E202 - Message not compatible with call state or not implemented Call State i = 0x0C Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - RELEASE pd = 8 callref = 0x8047 Jul 10 22:16:48.780: ISDN Se0/0/0:15 Q931: RX - RELEASE_COMP pd = 8 callref = 0x00D5 Basically when calling from HQ PH1 to 4000 ( in AAR) it work without any issue. Thanks, Ramcharan Arya On Tue, Jul 9, 2013 at 10:37 PM, jainpiyush2...@ymail.com wrote: Hello Ram, You can assign Hq device pool and location setting to cti route point and cti ports.. And assign site c device pool and location to site c phones... Regards, Piyush Jain Sent from my android device. -Original Message- From: Ramcharan Arya ramcharan.a...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Tue, 09 Jul 2013 7:59 AM Subject: [OSL | CCIE_Voice] AAR and uccx Hi, I have a CTI Route point 4000 and two CTI port 410x and 410y SiteC Phone 1 and SC Phone 2 are in CSQ which is assign to application and associated with trigger. Due to RSVP when call exceed ip rsvp bandwidth call from uccx to site Phones should use AAR and to over PSTN. My doubts are . What should be local and device pool of CTI ports so it should work in AAR when PSTN caller make call to CTI route point number 4000. Can someone please advice about this. Thanks, Ramcharan Arya ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Call Redirect - uccx
Check your redirect CSS on route point and CTI port On Jul 4, 2013 2:27 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: i script Call redirect to extension 5001 when press 1, by hard code and variable, both of them did not work. Usually pretty simple and just work, but not this time. Each time I press 1 , just said Please try again . What is most likely I miss here in configuration? Script: --- - Accept - Menu, press 1 , call redirect 5001 - Select Resource CSQ - End tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Query
Yes. On Jul 1, 2013 9:45 PM, Kapuria, Aman aman.kapu...@team.telstra.com wrote: Hey Guys, ** ** Do we have access to the Help page within the CUCM in lab? Do they block it? Can you click on service parameters to get the description? ** ** *Aman Kapuria** * ** ** ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] rtp location start time
I second that. Excellent experience at RTP. Wasn't impressed with the proctors at SJC. On May 1, 2013 7:52 PM, STEPHEN FREEBERG sfreeberg...@gmail.com wrote: RTP exams begin at 7:15 am, You should arrive no later than 7:00 am. David Blair is the proctor and in my opinion is the best proctor I have encountered at Cisco. Steve On May 1, 2013, at 5:06 PM, Karen Johnson karen.johnson...@yahoo.ca wrote: hi, anyone can share what start time in RTP and proctors there? tks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cucm dial string to match null string
It's quite trivial in UCM. Just configure plar http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_0_2/ccmcfg/b03dn.html#wpxref85834 Sent from my iSnuff On Oct 19, 2012, at 6:21 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: Forget my message below. Actually, ! has to match a number, so that probably won't work. You can use a ? To match 0 or more occurrences of the previous digit. So a few ideas to try: ^[0-9] (not 0 through 9) or \+? (nothing, + up through +++) You might need some explicit matches to avoid matching no-null strings. Interested to hear what works. DQ Dan Quinlan, CCIE #36129 daqui...@cisco.com On Oct 19, 2012, at 6:40 PM, Dan Quinlan (daquinla) daqui...@cisco.com wrote: Off the cuff not sure if there is a null. You can create two patterns: [0-9]! and ! . The first one matches anything with at least on digit - have it do no manipulation. The Null string should match the second pattern. DQ Dan Quinlan, CCIE #36129 daqui...@cisco.com On Oct 19, 2012, at 12:09 PM, Krishna vinayak_...@yahoo.com wrote: hi guys, In cucm what character or pattern that identifies the null string? in cme i am aware the // matches null string but in cucm i am trying to find the pattern. I have to do calling party transformation mask based on accepting the null string. any help is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Lab Exam Question
Hugo, I would recommend you spend the first 15 minutes reading through the exam and identify questions that may influence others. At the end of exam how you achieve a task is irrelevant. It's only the end result that matters Sent from my iSnuff On Oct 1, 2012, at 7:41 PM, Barrera, Hugo hugo.barr...@nexusis.com wrote: Hi Guy’s, This question is for anyone who has already taken the lab whether pass or fail… As a first timer during the lab as you progress in your configurations do the questions/tasks piggy back each other or have you found scenario’s where you have to wipe out previous configs to get the next task completed? The reason I ask is because I find myself doing this on the practice labs (not often) and it concerned me how a proctor would grade tasks if they had to be wiped out for another task towards the end of the lab. Regards, Hugo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] I'm screwed right?
You can try setting all the service params to default from the gui. -Pavan On Sep 22, 2012, at 15:14, chase mergenthal cm3_...@hotmail.com wrote: I didn't see there there was a carriage return when I copied and pasted to enable IPCC parameter... -Chase Mergenthal -- If winners never quit and quitters never win, then who coined the phrase, Quit while you’re still ahead.? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] sip trunk
I think cor can offer a good solution for this. Assign a separate cor group to each dialpeer say cust1 and cust2 in both inbound and outbound directions -Pavan On Sep 13, 2012, at 17:45, John John john_ccie2...@yahoo.com wrote: Dear All, We have two sipt trunk for 2 comapany:- company A - DID range 332211XX Company B - DID range 33XX and each company has own PBX,Company A has AVAYA and company B has cisco Call manager. and they have sip trunk to my gateway where is the E1 is connected. in my gateway there is two dial peer : dial-peer voice 1 voip description ## Customer-A ## answer-address 332211.. destination-pattern 332211.. modem passthrough nse codec g711ulaw session protocol sipv2 session target ipv4:192.168.10.10 dial-peer voice 2 voip description ## Customer-B ## answer-address 33.. destination-pattern 33.. modem passthrough nse codec g711ulaw session protocol sipv2 session target ipv4:192.168.20.10 So in my case if customer A send me the ANI as 3322 which is for customer B then the call will hit dial-peer 2 and the call will go outside with 3322 as the ANI number so how we can block Site A calls if they didn't use there own DID. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot
UCM 8.6.2 / Unity Connection 8.6.2 Is there a way to prevent users from dialing the Voicemail pilot directly ? I tried the following but it doesn't work (1) Place the Unity Route Pattern in a Partition (lets say Voicemail) (2) Configure the CSS on the Voicemail Pilot to include the Voicemail Partition (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it It seems to always use the Line/Device CSS to call into the Voicemail System when the Voicemail button is pressed on the phone (instead of using the CSS configured in Voicemail Pilot / Profile assigned to the device). Forwarded calls get routed to Voicemail as the CF CSS is set to use With Configured CSS and the CSS for Forwarding has the Voicemail CSS -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot
I am using a SIP based integration not the legacy SK so no hunt pilot. Just a route pattern to the SIP Trunk -Pavan On Tue, Sep 4, 2012 at 10:24 AM, William Affeldt william.affe...@yahoo.com wrote: Check the hunt pilot. What partition is it in? Sent from my iPhone On Sep 4, 2012, at 8:03 AM, Pavan K pav.c...@gmail.com wrote: UCM 8.6.2 / Unity Connection 8.6.2 Is there a way to prevent users from dialing the Voicemail pilot directly ? I tried the following but it doesn't work (1) Place the Unity Route Pattern in a Partition (lets say Voicemail) (2) Configure the CSS on the Voicemail Pilot to include the Voicemail Partition (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it It seems to always use the Line/Device CSS to call into the Voicemail System when the Voicemail button is pressed on the phone (instead of using the CSS configured in Voicemail Pilot / Profile assigned to the device). Forwarded calls get routed to Voicemail as the CF CSS is set to use With Configured CSS and the CSS for Forwarding has the Voicemail CSS -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Is it possible to prevent users from dialing VM Pilot
Stuart, I want all my users to be able to access voicemail using the voicemail button. I want to prevent users from picking up the phone and dialing the voicemail pilot from keypad. Stuart Geoghegan stuartmgeoghe...@yahoo.co.uk wrote: Pavan, Why don't you remove the enterprise device voicemail service parameter, re-add it (with enterprise not checked) subscribe it to only the users that you want to provide it to, Cheers Stuart Sent from my iPad On 4 Sep 2012, at 16:26, Pavan K pav.c...@gmail.com wrote: I am using a SIP based integration not the legacy SK so no hunt pilot. Just a route pattern to the SIP Trunk -Pavan On Tue, Sep 4, 2012 at 10:24 AM, William Affeldt william.affe...@yahoo.com wrote: Check the hunt pilot. What partition is it in? Sent from my iPhone On Sep 4, 2012, at 8:03 AM, Pavan K pav.c...@gmail.com wrote: UCM 8.6.2 / Unity Connection 8.6.2 Is there a way to prevent users from dialing the Voicemail pilot directly ? I tried the following but it doesn't work (1) Place the Unity Route Pattern in a Partition (lets say Voicemail) (2) Configure the CSS on the Voicemail Pilot to include the Voicemail Partition (3) Device/Line CSS on the Phone does not have the Voicemail pilot in it It seems to always use the Line/Device CSS to call into the Voicemail System when the Voicemail button is pressed on the phone (instead of using the CSS configured in Voicemail Pilot / Profile assigned to the device). Forwarded calls get routed to Voicemail as the CF CSS is set to use With Configured CSS and the CSS for Forwarding has the Voicemail CSS -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 79, Issue 8
Ram, If its based on 9.0 train of UCM i doubt v4.0 would come out before June 2013. I would expect ISR G2 and MPLS to be included in the new version with RT endpoints. Since Cisco seems to be adding CCIE tracks left and right, I would also like to see a CCIE in Contact Center Technologies added but thats just being too wishful ;) Again this is not based on any insider info and is pure unadulterated speculation -Pavan On Tue, Sep 4, 2012 at 1:25 PM, Ramcharan Arya ramcharan.a...@gmail.com wrote: Hi, I checked https://www.ipexpert.com/Cisco/CCIE/Voice/Bootcamps their website also advertising CCIE Voice version 4.0. When I click on one of the link its showing version 3.0. My guess is version 4.0 is coming soon because ipexpert start updating their website. Regards, Ramcharan Arya CCIE # 28926 (RS) On Mon, Sep 3, 2012 at 7:55 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: CCIE Blueprint version change (Dan Quinlan (daquinla)) 2. Voicemail SRST (Justin Barksdale) 3. UCCX (Bill Lake) 4. Re: Voicemail SRST (Dan Quinlan (daquinla)) -- Message: 1 Date: Tue, 4 Sep 2012 00:12:54 + From: Dan Quinlan (daquinla) daqui...@cisco.com To: Bikramjit Singh biksinghc...@gmail.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE Blueprint version change Message-ID: 7c6fde39-79f8-48b9-a096-caed1d166...@cisco.com Content-Type: text/plain; charset=us-ascii I have no inside info on this, but given that it wasn't announced at Cisco LIVE! this year, I'd expect that the next revision, if it happens in the next 12 months, will be 2900 and 3900 series gateways with PVDM3's, 9.x trains of software, and will include video endpoints of some fashion (8900 or 9900 series phones, perhaps the Jabber for Windows Client, maybe even a VCS-registered endpoint such as an EX unit or the Jabber Video client). As for timing, Cisco LIVE! next year seems logical to me. Again, this is all a guess and not based on any knowledge or fact. DQ d...@cisco.com Sent from my iPhone On Sep 3, 2012, at 1:51 PM, Bikramjit Singh biksinghc...@gmail.com wrote: Hi Folks, Is there anyone who has a logical prediction on when is the blueprint about to change? Also, what kind of changes are we expecting; both from software and hardware (voice gateways, pvdm, etc..) perspective. Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- Message: 2 Date: Mon, 3 Sep 2012 20:15:49 -0400 From: Justin Barksdale jus...@barksdale.net To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Voicemail SRST Message-ID: b824a61d-8404-461f-86c5-1dd6b6298...@barksdale.net Content-Type: text/plain; charset=us-ascii Steve, Voicemail while in SRST can be accomplished using the redirecting number in order to reach the required mailbox. Alternate extension are not required. Justin Barksdale CCIE #29866 Sent from my iPhone 4. On Sep 3, 2012, at 8:00 PM, ccie_voice-requ...@onlinestudylist.com wrote: Voicemail SRST -- Message: 3 Date: Mon, 3 Sep 2012 19:41:48 -0500 From: Bill Lake whl...@gmail.com To: CCIE Study ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] UCCX Message-ID: cadpb93on1m2ybbrofyktjw4zp-t4afi+h_fk7-j4kfaggjo...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Trying to install UCCX on laptop to practice scripting but while I installed Server 2003 and did the registry update, it says Installation of Cisco Unified Contact Center Express cannot be performed on the current version of MCS OS Service Release. Please upgrade the OS image version to 2003.1.5a or higher and try again Does anyone know how to overcome this issue. I tried the following [image: Inline image 1] Bill -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20120903/8c541fc4/attachment-0001.html -- next part -- A non-text attachment
Re: [OSL | CCIE_Voice] [cisco-voip] UCCX 9 Application User Password recovery ?
Thank you guys. I was able to recover using Ryan's suggestion after resetting the password. -Pavan On Mon, Aug 27, 2012 at 7:41 AM, Ryan LaFountain (rlafount) rlafo...@cisco.com wrote: Hi Pavan, You can use the Application User that was set during install to get into the GUI at any time in UCCX 9.0(1). If you have forgotten that password, you can reset it using the CLI commands found in the CLI Guide (the same as if you forgot the Application User in CCM). This user doesn't require authentication to CUCM, so you can use it to reset the AXL Provider user and designate more CCM End Users as UCCX Admins. HTH. Thank you, Ryan LaFountain Unified Contact Center Cisco Services Direct: +1 919 392 9898 Email: rlafo...@cisco.com Hours: M F 9:00am 5:00pm On 8/26/12 11:07 PM, Pavan K pav.c...@gmail.com wrote: I have a test UCCX 9 server that was configured and operational. The UCM cluster that was integrated with the UCCX was re-installed from scratch. Is there a way to recover the application user password / switch the server to the default post install state Basically trying to figure out a way to get into the appadmin gui. -- - Pavan ___ cisco-voip mailing list cisco-v...@puck.nether.net https://puck.nether.net/mailman/listinfo/cisco-voip -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UCCX 9 Application User Password recovery ?
I have a test UCCX 9 server that was configured and operational. The UCM cluster that was integrated with the UCCX was re-installed from scratch. Is there a way to recover the application user password / switch the server to the default post install state Basically trying to figure out a way to get into the appadmin gui. -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CME Option 150
I would put it as the sip cme source interface. -Pavan On Aug 23, 2012, at 11:38, Randall Crumm rrcr...@yahoo.com wrote: Hello, If not specified, which interface is it better to configure as the option 150 IP address in the DHCP pool for CME phones? Thanks, Randall ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode
I saw this problem before. Some pointed it out to be a cupc bug. I could never get it to consistently work but the following seem to have helped - make sure cupc machine has access to a dns server and can resole the cups seever name and sip proxy domain and restart cupc client - turn off auth mode in sip proxy service params on cup and restart cup server -Pavan On Aug 9, 2012, at 13:37, Jason Murray murr...@usa.com wrote: Nevermind seems to be a bug with the software. I choose Deskphone Mode and nothing happens, close CUPC and reopen and it comes up with Deskphone mode. Happens in reverse too, so if its already in Deskphone mode and I chose Disable nothing happens, close CUPC and reopen and it comes up disabled again. Strange. Nothing to do with the profile - Original Message - From: Jason Murray Sent: 08/09/12 01:06 PM To: Ramy Abdelrahim, ke...@kevinspicer.co.uk Subject: Re: [OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode I have a question that falls along these same lines. For the user that is assigned Deskphone mode, lets say they are at Site B. You would think the logical thing to do is to assign that user to the SB CTI Gateway profile, correct. When I do that I can never get deskphone mode to take, I end up having to assign it to the HQ CTI Gateway profile and it works just fine. Does this happen for anyone else? Just curious if its a config issue or something on why deskphone mode doesnt work when assigned to another profile other than HQs. Thanks Jason - Original Message - From: Ramy Abdelrahim Sent: 08/09/12 12:26 PM To: ke...@kevinspicer.co.uk Subject: Re: [OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode That's right. I forgot that and I assigned a CTI Gateway profile to the softphone user. Thanks Kevin. Regards, Ramy Date: Thu, 9 Aug 2012 18:20:18 +0100 Subject: Re: [OSL | CCIE_Voice] CUPS - Softphone vs Deskphone mode From: ke...@kevinspicer.co.uk To: ramyoth...@hotmail.com CC: ccie_voice@onlinestudylist.com This is because you have assigned a CTI profile to the CUPC user in Cups. The deskphone control settings are not relevant to CUPC users. On 9 Aug 2012 18:09, Ramy Abdelrahim ramyoth...@hotmail.com wrote: Dear All, In Workbook vol2, Lab 4 question 1.2, it's requested to configure CUPC as softphone which I did but still I can go to deskphone mode on the CUPC. Please note that the user is not assigned deskphone capabilities in presence. I configured another CUPC user as Deskphone mode then I tried to switch to softphone and off course it didn't work as expected because there was no UPCXX added in the Device list The question now is: - is it normal for a CUPC user configured as softphone (as requested in the question) to be able to switch and work in Deskphone mode as well? Regards, Ramy ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] service url in the cucm for the phoneview
Take a look at this. There is no service URL. http://blog.ipexpert.com/2010/11/17/setting-up-phone-view/ Sent from my iSnuff On Aug 1, 2012, at 9:29 PM, Krishna vinayak_...@yahoo.com wrote: hi folks, can any one help me out with the service url for phoneview for ip phones in cucm thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] dead air when calling CUE vmail gui doesn't work
Can you ping it from your test pc? I had a similar problem when cue was in SRST and nothing I tried worked Sent from my iSnuff On Jul 4, 2012, at 10:21 PM, Krishna vinayak_...@yahoo.com wrote: hi folks, I couldn't able to understand why the CUE giving me the dead air though after the configuration is absolutely correct with the right codecs. when i pressed the vmail button on the phone, it connects to the vmail number but i cannot hear anything. And, also i couldn't access web gui for the cue even after providing all the right info such as ip http server, ip http path, ip http auth local.. the web browser sits there forever with no output... does anyone experienced the same problem as i am??? your advice on this matter is much appreciated. thank you krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] CME SIP Phone -- Unity Connection DTMF issue
CME SIP Phone is in remote site with cRTP enabled between the CME HQ sites so RFC2833 / rtp-nte is un-usable. The only other option on CME SIP Phones is sip-notify. The available options on the dial-peer are sip-notify sip-kpml. The only option on Unity Connection is sip-kpml (RFC2833 cannot be used). Everything i have read / tried seems to indicate that sip-notify / sip-kpml do not work (even if using an XCODER). http://www.cisco.com/en/US/docs/voice_ip_comm/connection/7x/integration/cucme_sip/guide/cucintcucmesip030.html#wp1094879 Did anybody get sip-kpml / sip-notify to work with Unity connection from CME ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ?
With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] no trace found but calls get routed br1-Hq
Krishna, When dialing from station to station and both stations are registered to UCM, the call does not normally traverse through the PSTN (no AAR case). The signaling media flows over voip directly which is why you dont see any gateway / q931 debugs being active. However for a Voip flow to maintain proper quality, CAC/RSVP is used to ensure sufficient bandwidth being used which is why you see the RSVP debug active. Media flows from endpoint to endpoint directly through the RSVP agents which is what you see in sh sccp connections Signaling flows from endpoint to UCM direct. Remember the gateway is not in the signaling path which is why you do not see anything on the gw. On Mon, Jun 11, 2012 at 11:42 AM, Krishna vinayak_...@yahoo.com wrote: Hi folks, I couldn't understand the call flow between HQ and BR1 which are provisioned/registed in the cucm. here is the detail structure: HQ-phone1 -5002 css-hq-international pt-pt-internal BR1-phone1-1002 css-br1-ld pt-pt-internal Both phones are residing in the partition pt-internal, and br1 is a mgcp site and whereas the hq is the h323 site. when i call 1002 from 5002 or vice versa the call works fine, but when i enable deb isdn q931 or deb voip dialp, i dont see anything. Whereas when i enable RSVP based CAC, i can see the traces with the show sccp connections. could any one help me out how the calls are working in between these two. is it because the phones are registered to cucm, but logically in a different device pool and therefore it routes directly on cucm your help is much appreciated. Thank you. Krishna. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] [Resolved] UCCX - Do CTI Ports in Partition work ?
Thanks Gurpreet, Dan Krishna. This is now fixed. As Dan mentioned the CSS of caller matters. == When a CTI Route point, redirects the call to the CTI port, the CSS of the device that calls the Route point is used to search for the CTI Port. == From: Pavan K pav.c...@gmail.com Date: Mon, Jun 11, 2012 at 1:33 PM To: ccie_voice@onlinestudylist.com With UCCX, did anybody get calls to work when the CTI ports are in a partition ? If so what CSS did you have to configure ? I have created a RoutePoint in the NULL Partition and CTI ports in a UCCX partition Added a CSS for the RoutePoint that includes the UCCX partition (on both the line device) but the call doesn't connect. If i take the CTI ports out of the partition, everything works perfectly. TIA -- - Pavan -- From: Gurpreet Singh Kukreja tycoononway1...@gmail.com Date: Mon, Jun 11, 2012 at 3:27 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com Hi Pavan, We've seen this behavior with UCCX. Logically, the calls should work w/ or w/o partition applied on the CTI Port Group, keeping in mind the CSS applied on the CTI Route Point. Few things to keep in mind: 1) Always apply the changes on these Triggers/ Port Groups from the CCX and never from the CM. 2) If you apply the correct CSS on the Trigger which includes the partition of the Port group, the calls should work. 3) Even after applying the changes if the calls do not work, it could be very possible that the changes you're making from the CCX are not getting updated on the CM. In this case, first run the Data Resync from the CCX and make sure there are no exceptions in the output. Then, restart the CTI Manager on all CM servers and then restart the CCX Engine. - Gurpreet ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- From: Dan Quinlan (daquinla) daqui...@cisco.com Date: Mon, Jun 11, 2012 at 3:34 PM To: Pavan K pav.c...@gmail.com Cc: ccie_voice@onlinestudylist.com I would think that the inbound caller (ip phone or gw) would need the CSS to access the CTI ports. DQ d...@cisco.com Sent from my iPhone -- From: Krishna vinayak_...@yahoo.com Date: Mon, Jun 11, 2012 at 3:53 PM To: Pavan K pav.c...@gmail.com pavan, I worked on uccx lab and it worked fine for me. All that you need to remember one point always, what does the CTI Route point has to see. in this case the CTI route point has to see the phones partition in order to handover the calls to the phone agents. Check that internal dns are listed in your css to make this work. thank you krishna. From: Pavan K pav.c...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Monday, June 11, 2012 1:33 PM Subject: [OSL | CCIE_Voice] UCCX - Do CTI Ports in Partition work ? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cme sccp phone cannot watch the sip phone
I havent used 7960 so cant comment on icon. Sip phones off hook status will be known to cme only when the invite comes in to cme which would be after a digit press (if using kpml) or a pattern match ( if using vr dialplan) -Pavan On Jun 9, 2012, at 10:55 PM, Krishna vinayak_...@yahoo.com wrote: hi folks, i have a blf-sppedd-dial enabled for both sip and sccp phones on the cme site, i can view the status of the sccp phone on sip phone when the sccp phones goes off hook, whereas i cannot view the same from sccp phone for sip phone..is this expected behavior?? and also i am attaching the image that shows the status on the sccp phone which looks so weird compared to normal blf icon... please have a look at the attached image for reference. any advice on this matter is much appreciated. thank you Krishna. photo (2).JPG ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
[OSL | CCIE_Voice] Directed Call Park Range + BLF Question
Ok, If i create a DPark number (say 5100) then i have no problem assigning it to a BLF DPark Softkey. However if i create a range of DPark numbers (say 510[0-3] or 510[0123] ) then the numbers do not show up in the drop down while assigning them to a BLF DPark Softkey. Also there is no box to enter a freeform number. Any trick to get this to work ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Built-in-Bridge
Correct. Barge however will not work. Only cbarge/conference will. Also Be aware software CFB is g711 only. Sent from my iSnuff On Apr 29, 2012, at 4:49 AM, Ken Wyan kew...@gmail.com wrote: If we disable built-in-bridge of a phone , it uses conference resources available through MRGL for ad-hoc conferences Barge/cBarge . These external Conference resources may be hardware CFB or CUCM ipvoicemediastreamingapp software CFB resources. Is this correct? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Changing NTP Timezone on UCM Pub for OS Admin ?
Thanks Baktha. That did it. -Pavan On Tue, Apr 10, 2012 at 10:56 AM, Baktha Muralidharan muralic...@gmail.comwrote: show timezone list set timezone but I don't think you need to worry about, as far as times displayed on the phones. DTGs will take care of that. thanks, /Baktha Message: 3 Date: Tue, 10 Apr 2012 09:39:42 -0500 From: Pavan K pav.c...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Changing NTP Timezone on UCM Pub for OS Admin ? Message-ID: CAJDPBuVfgGzp5HNtYMrcUXfZYJ53jZU7negpE2q--ztE=y0...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 When looking at utils ntp status on UCM pub, i see the Timezone as CST. Is there a way to change the Timezone from Amer/Chicago to a diff Zone. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] [Hyderabad India Dialplan]
Incorrect. Hyderabad is a city :) There are a bunch of issues with TRAI regulations in INDIA. Cisco IT has a case study on cisco.com -Pavan On Mar 13, 2012, at 2:37 PM, Kumar Vishal (kvishal) kvis...@cisco.com wrote: Hi Mike, Hyderabad is a state and not a country. Better use India Dial Plan. The total length of all phone numbers (STD code and the phone number) in India is constant at 10 digits, so you should be good. Things to keep in mind – 1 Check with Telco if they accept E164 number. Once you are sure, only then implement + dialing else stick with the leading 0 concept. 2 Check if you need to implement Logical Partitioning(LP) as India does not allow toll bypass. Let me know if you need help here. Thanks Kumar From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of arun thomas Sent: Wednesday, March 14, 2012 12:52 AM To: michael.se...@compucom.com Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] [Hyderabad India Dialplan] http://en.wikipedia.org/wiki/Telephone_numbers_in_India On Mon, Mar 12, 2012 at 9:51 PM, michael.se...@compucom.com wrote: I’ve had a new site come up in Hyderabad India. Wondering if someone could share information regarding the dial-plan used there. Any information would be appreciated. Thank you, --ms Michael Sears ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Question on Join Across Lines
Its basically join softkey. On Sep 21, 2011, at 10:30, Mann Chaddha mann.chad...@gmail.com wrote: Hi Guys Does anyone know what construct JAL uses while bridging 2 calls on different Line Buttons? I ask as I need to plan India specific dial plan which shall restrict bridging of VoIP Calls to Local PSTN Calls. I went through Geolocations but so far am not too convinced with its usability as a well constructed dial plan shall never zero in on 2 IP Endpoints which are not allowed to converse with each other in the first place. Do advise. Thanks Mann ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Disable G.722?
Michael, Regions never restrict based on codec, they restrict based on bw req per call. Since g711 g722 both consume same bw and g722 has a higher preference, phones will negotiate to g722 if the enterprise param is not changed. On Jun 10, 2011, at 16:00, Randall Crumm rrcr...@yahoo.com wrote: If the phones can negotiate g722 they will Randall From: Michael Luo hout...@gmail.com To: ccie_voice@onlinestudylist.com Sent: Friday, June 10, 2011 1:03 PM Subject: [OSL | CCIE_Voice] Disable G.722? I heard the rumor that even if you specify G.711 as intra-region codec, you'll have to disable G.722 in Enterprise Parameters and Service Parameters. Otherwise, G.722 will be chosen. I tested it in the lab. That was not the case. If I explicitly specified G.711 as intra-region codec, the phones will always use G.711. Do we still need to disable G.722? Thanks! ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Problems invoking ios transcoders
Karim, It is ccm that invokes the xcoder (not ipcc) when it sees a cap mismatch between cti port and the phone. Check the ccm sdl logs and see if you see an allocatemtpreq and see what happened to it. If you do not see such a req then for some reason ccm thinks there is no cap mismatch. -pavan On May 23, 2011, at 0:53, Karim Hany karimhan...@gmail.com wrote: Hi, G.729r8 is configured under the transcoder profile, this is i'm 100% sure about, and the transcoders are registered with CUCM. I have HQ-IOS-MRGL which has HQ-IOS-MRG and this includes hq-xcoder and the same configuration for BR1 and both are registered correctly with CUCM. I've tried every possible combination. i.e I assigned br1-xcoder to HQ-MRG and vice versa for BR1, but i know that it is IPCC which should be invoking the transcoder resources. and the CTI ports are in the internal-pt/HQ-DP. This is why i'm going nuts and doesn't have any explanation to what's going on. Is there any missing configuration or setup i might be missing? Thanks Karim An On Mon, May 23, 2011 at 1:10 AM, Bartosz Sokolowski ibartosz.sokolow...@gmail.com wrote: Hi, If your MRG/MRGL config is correct (CTI Ports have access to xcoder resource) then it must work. Check if you have g729r8 codec in your dsp profile on IOS. By default this codec is missing. -- Regards, Bartosz 2011/5/22 Karim Hany karimhan...@gmail.com Hi All, I have problem with invoking transcoder. I registered two IOS transcoders one for HQ BR1 both are registered to CCM. The goal is BR1 should be able to call ICD pilot (CRS uses codec G711 only) number on HQ using codec G729. But it always fail to invoke transcoder at HQ and call fails and I get a fast busy tone. Call from HQ to ICD is ok because no transcoder is needed. However, when i call from BR1 to HQ via PSTN, the call gets through and I can hear the welcoming prompt. I've also tried to associate BR1 xcoder with HQ IOS MRG/MRGL which is assigned to HQ DP as if it's a local resource but still not working. Any idea on what could be the problem or misconfiguration. Appreciate your advice and inputs. Thanks Regards, Karim ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Volume 2 Lab 7, Task 4.4
Did you change settings in the default system transfer restrictions table? By default all dns are blocked On Nov 24, 2010, at 5:39 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi, I'm currently facing the problem in which UC response with You cannot be transfer to this number... during the opening greeting. Steps done according to the PG, which check the option Allow Transfer to Number not associated with user or call handler, but still no luck. Some test applied was, tried calling VM by pressing the message button, press # while asking for login PIN. During Opening Greeting, dial the user DN with no mailbox on UC. Did i miss out anything? Shingei ___ 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
Re: [OSL | CCIE_Voice] Presence CTIGW AppUser
Assign it cti enabled and cti control of all devices. No need to associate any device to it. Also for your end user you will need cti enabled -pavan On Nov 15, 2010, at 10:07 AM, Mann Chaddha mann.chad...@gmail.com wrote: Hi Guys I somehow am not able to recall the Roles for the CTIGW User for Presence Desktop Control. Don't even have a SERVER Infront of me. This is the UCM Appuser. What are the roles we need to assign to it wht devices are associated to this? Kindly respond at your earliest convenience as I have my Lab in 12 hrs. Thanks Mann ___ 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
Re: [OSL | CCIE_Voice] retaking date
If you select a date and go to the next page it will throw an error if you are not allowed On Nov 10, 2010, at 12:19 AM, rsmail...@solcon.nl rsmail...@solcon.nl wrote: Well, that's why i am wonering. it let's me take it on 22 and 24 the other date's are full. the 22 is for sure not after 30 day's 24 is the 30th day I am waiting for the cisco support team to respond to my request. regards Ron It won't let you schedule any earlier than you're allowed to. Try to schedule it online, if it let's u...you're all set Sent from my phone, apologies for any typos. On Nov 9, 2010, at 3:04 AM, Shady Hasan shady@gmail.com wrote: I have a similar issue. To be 100% sure, please register to Cisco Certification and Communities Online support https://ciscocert.secure.force.com/english/MainPage; Ask your case and you will have official reply from Cisco within 2 business days. Regards, Shady. On Tue, Nov 9, 2010 at 8:34 AM, rsmail...@solcon.nl rsmail...@solcon.nl wrote: hi, i am wondering how the 30 day retake period counts. the 30 days start the day after taking the exam. but is it that you can take a new exam on day 30 or is it also the day after day 30 ? i am wondering, because otherwise i have to wait till next year before retaking the exam. example: 25 october exam date 24 november retake (is the 30st day) regards for the info ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] BLF call list - Presence group issue
I assume your subscribe css are all set correctly and you restarted ccm service ? On Nov 10, 2010, at 4:43 PM, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi All!! Working with lab 13 and blf call list I’ve found and issue that is driving me crazy!! I have two Presence groups Standard Presence Group , Employees Presence group. The group Relationship: Use System Default (Dissallow Subscriptions) hqph1, hqpph2 (Employees) hqph3,br1ph1(Standard) The current behavior I have: From directory hqph1 and hqpph2 can’t see presence status of anybody: for hqph3,br1ph1, that’s ok, but hqph1 can’t see hqph2 and vice versa, and they are in the same presence group And hqph3, br1ph1 can see the presence status of all, also hqph1 and hqph2, although they are in different presence group and subscriptions are disallowed What am I doing wrong? Does anyone understand what it’s happening? Thank you very much!! FRancesc ___ 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
Re: [OSL | CCIE_Voice] delayed Multicast MOH streaming - Any ideas ?
I did reboot everything. I kind of isolated the issue to frame relay links. Thats the only thing that makes sense at this point. Any special config for mcast over fr? I tried nbma-mode and it didnt make any difference On Nov 10, 2010, at 4:21 PM, Prashant Patel prashantpatel...@gmail.com wrote: I have seen a delay like 30 seconds when I am using g729 stream to the branches but like Miron said 5 mins is a long time and I would have rebooted everything by then :) On Wed, Nov 10, 2010 at 5:09 PM, Miron Kobelski findko...@gmail.com wrote: forget it, I missed the part where you write in works in HQ... no other idea. what's SK? On Wed, Nov 10, 2010 at 23:08, Miron Kobelski findko...@gmail.com wrote: Wow, you must be very persistent to wait 5 minutes to make such observation :) This issue is strange indeed. I'd try different Audio Source / wave file + maybe check if you have looping enabled (MoH repeating, I don't remember the exact setting name/location - either on the audio file config or audio source or moh server...). regards kobel ___ 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
Re: [OSL | CCIE_Voice] delayed Multicast MOH streaming - Any ideas ?
G711 On Nov 10, 2010, at 3:07 PM, ccieid1ot ccieid...@gmail.com wrote: What's the region set to for the SB to MOH? On Tue, Nov 9, 2010 at 6:01 PM, Pavan K pav.c...@gmail.com wrote: Multicast MOH from CCM. No Spoofing. For HQ site, MMOH stream perfectly For Branch sites, on the Branch router i can see MMOH packets coming in with the debug ip mpacket command but the phone doesnt play MOH until about 5 mins later. In other words, after pressing Hold, Phone starts streaming music after being on hold for 5 mins. Phones are SK registered to CCM. CCM counters look good (show mcast stream active) pim is configured for sparse-dense-mode on all serial subifs, vlans loopbacks. IOS is 12.4 (20T2) -- - Pavan ___ 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
Re: [OSL | CCIE_Voice] BLF call list - Presence group issue
To rule out css, can you put the problem phones in the null partition and see if you still have the issue On Nov 10, 2010, at 5:15 PM, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi!! Yes all the phones have the same subscribe css with phones partition. And also I have restarted the ccm service, in fact the whole cluster. Francesc De: Prashant Patel [mailto:prashantpatel...@gmail.com] Enviado el: miércoles, 10 de noviembre de 2010 23:50 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] BLF call list - Presence group issue Hi Francesc, On the phones check the subscribe css. It should be the one with the phone partitions in it. HTH Prashant On Wed, Nov 10, 2010 at 5:43 PM, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi All!! Working with lab 13 and blf call list I’ve found and issue that is driving me crazy!! I have two Presence groups Standard Presence Group , Employees Presence group. The group Relationship: Use System Default (Dissallow Subscriptions) hqph1, hqpph2 (Employees) hqph3,br1ph1(Standard) The current behavior I have: From directory hqph1 and hqpph2 can’t see presence status of anybody: for hqph3,br1ph1, that’s ok, but hqph1 can’t see hqph2 and vice versa, and they are in the same presence group And hqph3, br1ph1 can see the presence status of all, also hqph1 and hqph2, although they are in different presence group and subscriptions are disallowed What am I doing wrong? Does anyone understand what it’s happening? Thank you very much!! FRancesc ___ 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
[OSL | CCIE_Voice] Unity connection Broadcast message MWI
Is there a way to get MWI for broadcast messages (sent using Broadcast admin) The message is in the mailbox but no MWI for these messages. MWI works normally in other cases. -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Missed call redial + dialing
Q1 if you want to try it out, it should display as 4 digits but call list should have full e164. Q2 You can put a TP in HQ phone 1 CSS to exapnd calling number to e164. But really there are tons of possibilities ways of accomplishing this (gateway translations, transformations) depends on what you want. On Tue, Nov 9, 2010 at 12:19 PM, Shrini linuxbos...@gmail.com wrote: Hi Mike, Ans_to_Q1 : I am not following any specific question, just testing and wanted to understand how it should display. Ans_to_Q2 : If I manipulate at gateway level all other calls are affected and all calls ANI will be displayed as +14082011001 including where I want ANI to be 2011001, so setting at GW does not work. -Shrini On 11/9/2010 10:10 AM, Mike Nipp (mnipp) wrote: Shini, Q: When I redial a missed call +19193012001 (on HQ phone 1001) Br1 Phone 2001 is ringing and is displaying number as from 1001. Is this as per lab requirement or does it need to display +14082011001 ? A: What is the question you to display? Q: Since redial of missed call matches route pattern \+! directly goes to GW. Where should I set so that calling party should displayed as +14082011001. A: You can apply a calling party transformation at the egress gateway to manipulate the calling number to the PSTN. This will change 1001 to +14082011001. *From:* ccie_voice-boun...@onlinestudylist.com [ mailto:ccie_voice-boun...@onlinestudylist.comccie_voice-boun...@onlinestudylist.com] *On Behalf Of *Shrini *Sent:* Sunday, November 07, 2010 7:08 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Missed call redial + dialing Greetings Experts: When I redial a missed call +19193012001 (on HQ phone 1001) Br1 Phone 2001 is ringing and is displaying number as from 1001. Is this as per lab requirement or does it need to display +14082011001 ? Since redial of missed call matches route pattern \+! directly goes to GW. Where should I set so that calling party should displayed as +14082011001. TIA Shini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] delayed Multicast MOH streaming - Any ideas ?
Multicast MOH from CCM. No Spoofing. For HQ site, MMOH stream perfectly For Branch sites, on the Branch router i can see MMOH packets coming in with the debug ip mpacket command but the phone doesnt play MOH until about 5 mins later. In other words, after pressing Hold, Phone starts streaming music after being on hold for 5 mins. Phones are SK registered to CCM. CCM counters look good (show mcast stream active) pim is configured for sparse-dense-mode on all serial subifs, vlans loopbacks. IOS is 12.4 (20T2) -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] [Q] SRST DHCP in real world.
Lets say we are required to configure Branch1 to be an SRST site. Also during normal operation, Branch1 phones get DHCP from CCM in HQ. Now when BR1 is in SRST, If any of the phones reset, they will not be able to get DHCP and will be unable to register to SRST. Does this question imply an additional DHCP configuration in BR1 site ? If so, is there any way to make the HQ DHCP primary and BR1 DHCP as fallback when the HQ DHCP pool is in-accessible. How do people deal with this in real world (other than having local DHCP on BR1) ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [Q] SRST DHCP in real world.
Thanks for all the input folks. Yes i did notice that reseting a phone from CLI causes the phone to lose its IP. -Pavan On Fri, Nov 5, 2010 at 5:31 PM, George Goglidze gogli...@gmail.com wrote: nobody will keep an IP if there is no DHCP present. Neither Phone, nor PC (Win/Linux/Mac). Regards, On Fri, Nov 5, 2010 at 3:13 PM, cciefo...@hotmail.com wrote: If I am not mistaken they would keep their previous address, the only it would try to get a new ip if it was a new phone or there was a factory reset or the network settings were changed, other than that they function normally so it is not an issue. -Original Message- From: Pavan K pav.c...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Fri, 5 Nov 2010 09:24:57 To: osl oslccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] [Q] SRST DHCP in real world. ___ 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 -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] [Q] SRST DHCP in real world.
Agreed. My question was can it be done if DHCP for Branch1 was scoped from HQ. Looks like it cannot. -Pavan On Fri, Nov 5, 2010 at 6:07 PM, Shrini linuxbos...@gmail.com wrote: IOS DHCP on Br1 router is required. You will provide option 150 as CUCM IP address under DHCP pool. Also you are configuring SRST ref on CUCM which means Callmanger tell the devices if I fail get the config from router. Hope this clarifies. On 11/5/2010 7:24 AM, Pavan K wrote: Lets say we are required to configure Branch1 to be an SRST site. Also during normal operation, Branch1 phones get DHCP from CCM in HQ. Now when BR1 is in SRST, If any of the phones reset, they will not be able to get DHCP and will be unable to register to SRST. Does this question imply an additional DHCP configuration in BR1 site ? If so, is there any way to make the HQ DHCP primary and BR1 DHCP as fallback when the HQ DHCP pool is in-accessible. How do people deal with this in real world (other than having local DHCP on BR1) ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Lost of + and calling name after send to PSTN
You need to add a translation profile on outgoing side of pots dp. On Sep 1, 2010, at 6:23 PM, vcciev vcc...@gmail.com wrote: For the calling name, I verified with local outgoing call. But it is international call having problem. For the + sign, I already added back via a voice translation-profile at the POTS dial-peer. Still the same. ___ 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
Re: [OSL | CCIE_Voice] GateKeeper Problems
Well looks like either the css did not take effect(i.e. No reset issued after change) or no tp exists. Can you post the complete da signals. The little bit you posted doesnt help. On Aug 28, 2010, at 9:29 AM, DeShon Crayton dcrayto...@gmail.com wrote: I have included ccm and sdl traces. I see that no matches are found: Fqdn=pi=0si1 Cgpn=tn=0npi=0nd=3003pi=1si0 DialedNum=tn=0npi=0nd=1#5004pi=0si1 requestID=0 DigitAnalysisComplexity=0 04710| 2010/08/28 08:50:01.090| 002| SdlSig | DaRes | setup_da | Cdcc(2,100,171,6) | Da(2,100,164,1) | (0,0,0,0).0-(*:10.10.3.1) | [R:NP - HP: 0, NP: 0, LP: 0, VLP: 0, LZP: 0 DBP: 0]CI=48500372 Block NoPotentialMatchesExist OnNetrequestID =0 My trunk has a CSS that has access to a partition with a translation pattern that strips the 1#. On Fri, Aug 27, 2010 at 10:42 PM, Josmar Ramirez jrami...@ccsinet.com wrote: Of course I meant on the incoming digits on the ccm trunck - Original Message - From: Josmar Ramirez To: 'edot...@ams.net' edot...@ams.net; 'pav.c...@gmail.com' pav.c...@gmail.com; 'ccie_voice-boun...@onlinestudylist.com' ccie_voice-boun...@onlinestudylist.com; 'dcrayto...@comcast.net' dcrayto...@comcast.net Cc: 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com Sent: Fri Aug 27 22:39:20 2010 Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems On the callmanager make sure you set the significant digits to 4 so that it strips the 1# when it hits the trunk on ccm. Careful this might break any teho config that you might be sending to ccm. - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Pavan pav.c...@gmail.com; ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com; dcrayto...@comcast.net dcrayto...@comcast.net Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Fri Aug 27 21:55:40 2010 Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems Are you doing any digit translation to remove the prefix. That was tripping me up. Sent from my Verizon Wireless BlackBerry -Original Message- From: Pavan pav.c...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Fri, 27 Aug 2010 20:19:04 To: dcrayto...@comcast.netdcrayto...@comcast.net Cc: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems ___ 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 ccm0001.txt SDL002_100_98.txt ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] GateKeeper Problems
Ok, i missed the traces you attached at the very bottom. From the below line it is very clear that you either dont have a CSS assigned or did not reset after assigning CSS as the pss field is blank. and as warren pointed out the most important keyword one is looking for is the BLOCK and not the NoPotentialMatchesExist Digit Analysis: getDaRes - voiceMailCallingSearchSpace=[]|CLID::StandAloneClusterNID::172.16.30.201CT::0,0,0,0.0IP::10.10.3.1DEV::LVL::State TransitionMASK::0800 08/28/2010 08:50:01.090 CCM|Digit analysis: match(pi=2,fqcn=, cn=3003, plv=5, *pss=*, TodFilteredPss=, *dd=1#5004*,dac=0)|CLID::StandAloneClusterNID::172.16.30.201CT::0,0,0,0.0IP::10.10.3.1DEV::LVL::DetailedMASK::0800 Regards -Pavan On Sat, Aug 28, 2010 at 4:10 PM, Warren Heaviside (wheavisi) wheav...@cisco.com wrote: When interpreting SDI traces and you see NoPotentialMatchesExist it's a bit misleading. What it actually means is no more potential matches exist and the Digit Analysis process is complete and has made a match. Warren Warren Heavisidewheav...@cisco.com ENGINEER.CUSTOMER SUPPORT High Touch Technical Support Phone: +1 408 853 7995 Office Hour 9 am - 5 pm Pacific Monday - Friday For corporate legal information go to: http://www.cisco.com/web/about/doing_business/legal/cri/index.html -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Saturday, August 28, 2010 1:41 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 54, Issue 95 Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: GateKeeper Problems (ccieid1ot) 2. Re: GateKeeper Problems (Pavan) 3. Proctorlabs.com is not loading (Aug 28, 1600 hours EST) (David Lee) 4. Re: Proctorlabs.com is not loading (Aug 28, 1600 hours EST) (David Lee) 5. proctor lab down ??? (Erwan Erwan) -- Message: 1 Date: Sat, 28 Aug 2010 13:12:15 -0500 From: ccieid1ot ccieid...@gmail.com To: Josmar Ramirez jrami...@ccsinet.com Cc: ccie_voice@onlinestudylist.com, ccie_voice-boun...@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems Message-ID: aanlktimvnzvdwu+o_hcp0-j57hfgjgqssexs4gwoq...@mail.gmail.comaanlktimvnzvdwu%2bo_hcp0-j57hfgjgqssexs4gwoq...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Yes, digit manipulation or sig dig 4. On Fri, Aug 27, 2010 at 9:42 PM, Josmar Ramirez jrami...@ccsinet.comwrote: Of course I meant on the incoming digits on the ccm trunck - Original Message - From: Josmar Ramirez To: 'edot...@ams.net' edot...@ams.net; 'pav.c...@gmail.com' pav.c...@gmail.com; 'ccie_voice-boun...@onlinestudylist.com' ccie_voice-boun...@onlinestudylist.com; 'dcrayto...@comcast.net' dcrayto...@comcast.net Cc: 'ccie_voice@onlinestudylist.com' ccie_voice@onlinestudylist.com Sent: Fri Aug 27 22:39:20 2010 Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems On the callmanager make sure you set the significant digits to 4 so that it strips the 1# when it hits the trunk on ccm. Careful this might break any teho config that you might be sending to ccm. - Original Message - From: ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com To: Pavan pav.c...@gmail.com; ccie_voice-boun...@onlinestudylist.com ccie_voice-boun...@onlinestudylist.com; dcrayto...@comcast.net dcrayto...@comcast.net Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Fri Aug 27 21:55:40 2010 Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems Are you doing any digit translation to remove the prefix. That was tripping me up. Sent from my Verizon Wireless BlackBerry -Original Message- From: Pavan pav.c...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Fri, 27 Aug 2010 20:19:04 To: dcrayto...@comcast.netdcrayto...@comcast.net Cc: ccie_voice@onlinestudylist.comccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] GateKeeper Problems ___ 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
Re: [OSL | CCIE_Voice] GateKeeper Problems
Can you post ccm sdi trace. That should point to the problem in a jiffy On Aug 27, 2010, at 7:21 PM, dcrayto...@comcast.net wrote: Hello guys, I am struggling with a gatekeeper scenario. Basically, the lab calls for HQ, BR1, and BR2 to use a GK to route internal calls via 4 digits. Of course BR2 is a CUCME. I can succesfully place calls from the CUCM registered phones, but not from CUCME to CUCM registered phones. I am sure that CUCM is giving the incorrect number dialed message, but I cannot pinpoint why. My config is as follows: HQ Gatekeeper voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 interface Loopback0 ip address 10.10.1.1 255.255.255.255 h323-gateway voip interface h323-gateway voip bind srcaddr 10.10.1.1 gatekeeper zone local PL cisco.com zone prefix PL 1... gw-priority 10 gk-trunk_2 zone prefix PL 1... gw-priority 9 gk-trunk_1 zone prefix PL 1... gw-priority 0 SiteC zone prefix PL 3* gw-priority 10 SiteC zone prefix PL 3* gw-priority 0 gk-trunk_2 gk-trunk_1 zone prefix PL 5... gw-priority 10 gk-trunk_2 zone prefix PL 5... gw-priority 9 gk-trunk_1 zone prefix PL 5... gw-priority 0 SiteC no shutdown BR2 CUCME voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol cisco h323 h225 listen-port 1820 no call service stop sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 interface Loopback0 ip address 10.10.3.1 255.255.255.255 h323-gateway voip interface h323-gateway voip id PL ipaddr 10.10.1.1 1719 h323-gateway voip h323-id SiteC h323-gateway voip tech-prefix 3 h323-gateway voip bind srcaddr 10.10.3.1 dial-peer voice 1000 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric codec g711ulaw ip qos dscp cs3 signaling no vad Sucessful call from HQ/BR1 to BR2 site From ext 1004 to 3001 via gatekeeper *Aug 27 21:18:17.167: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Aug 27 21:18:17.167: ////GK/gk_rassrv_arq: arqp=0x487106CC,crv=0x1, answerCall=0 *Aug 27 21:18:17.167: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/gk_dns_query: No Name servers *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_get_addrinfo: (3001) Matched tech-prefix 3 *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_get_addrinfo: (3001) unresolved zone prefix, using source zone PL *Aug 27 21:18:17.167: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_arq_select_viazone: about to check the source side, src_zonep=0x4877E344 *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_arq_select_viazone: matched zone is PL, and z_invianamelen=0 *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_arq_select_viazone: about to check the destination side, dst_zonep=0x4877E344 *Aug 27 21:18:17.167: //004AB3920100/004AB3920100/GK/rassrv_arq_select_viazone: matched zone is PL, and z_outvianamelen=0 *Aug 27 21:18:17.167: ////GK/gk_rassrv_get_ingress_network: returning default ingress network = 1 *Aug 27 21:18:17.199: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Aug 27 21:18:17.199: ////GK/gk_rassrv_arq: arqp=0x48856C9C,crv=0x1B, answerCall=1 *Aug 27 21:18:17.199: //004AB3920100/004AB3920100/GK/gk_rassrv_dep_arq: ARQ Didn't use GK_AAA_PROC *Aug 27 21:18:26.799: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Aug 27 21:18:26.807: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Aug 27 21:18:31.015: ////GK/gk_process: got a TIMER event Failed Call from BR2 to UCM From CUCME 1004 to HQ 5001 *Aug 27 21:44:12.055: ////GK/gk_process: QUEUE_EVENT (minor 0) wakeup *Aug 27 21:44:12.055: ////GK/gk_rassrv_arq: arqp=0x47EE1530,crv=0x1C, answerCall=0 *Aug 27 21:44:12.055: ////GK/gk_rassrv_sep_arq: ARQ Didn't use GK_AAA_PROC *Aug 27 21:44:12.055: //B6D0B3BB80B3/B6D14FE380B5/GK/gk_dns_query: No Name servers *Aug 27 21:44:12.055: //B6D0B3BB80B3/B6D14FE380B5/GK/rassrv_get_addrinfo: (1#5001) Matched tech-prefix 1# *Aug 27 21:44:12.055: //B6D0B3BB80B3/B6D14FE380B5/GK/rassrv_get_addrinfo: (1#5001)
Re: [OSL | CCIE_Voice] problem with outbound call
In such cases grabbing detailed SDL SDI traces would immensley help. Without them it is difficult to guess Sent from my phone On Aug 22, 2010, at 3:46 PM, CCIE Voice cc...@corb.net wrote: Tried with 1 sip phone and 1 sccp phone. Route pattern was set to route. Thanks for the ideas though. -- On Aug 22, 2010, at 14:13, bkvalent...@gmail.com bkvalent...@gmail.com wrote: Was the phone using SIP? - Reply message - From: CCIE Voice cc...@corb.net Date: Sun, Aug 22, 2010 3:49 pm Subject: [OSL | CCIE_Voice] problem with outbound call To: ccie_voice@onlinestudylist.com I have run into a strange problem that I can not figure out. Dialing digits on phone at BR2 (with what I can tell are correct CSS/partitions, gateway assignments) disconnect immediately after completing the dialing. e.g. Dialing 912123942123 call disconnects the moment that last digit is dialed. The call never hits the gateway. It is supposed to use MGCP gateway on BR1 router which appeared to be functional. I even converted this to h323 gateway and used a specific route pattern to force to that gateway...same problem. Reset, br1-rtr, reset gateway(s), reset CUCM (pub sub) all to no avail. I have run out of lab time and could not do debugs in rtmt to figure this out but was hoping someone else has experienced it and figured it out. tia...scd -- Steve Dickey ___ 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
Re: [OSL | CCIE_Voice] MGCP please add to first mail on Mgcp.
It means layer 3 is being backhauled to ccm bcoz you have an isdn bind-l3 ccm-manager on the interface. Consequently router may not have any/correct information about layer 3. Sent from my phone On Aug 17, 2010, at 4:23 AM, Pithog Oil pithog...@yahoo.com wrote: Oh i think my questtion was not properly framed, i should be asking, some one to help explain what that statment means. From: Graham Hopkins ghopk...@wolf-rock.co.uk To: Pithog Oil pithog...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Mon, August 16, 2010 12:37:24 PM Subject: Re: [OSL | CCIE_Voice] MGCP please add to first mail on Mgcp. Why do you want this to stop appearing? What do you think this is saying ? Graham On 16 Aug 2010, at 06:26, Pithog Oil pithog...@yahoo.com wrote: how do i make sure this prompt stops appearing when configuring MGCP? Will this prompt affect my configurations, what is the effect of this prompt on my lab. %Q.931 is backhauled to ccm manager 0X003 on DSL1 . layer 3 output may not apply. ___ 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
[OSL | CCIE_Voice] 3750 Sw - Queuing Threshold2 Threshold1. What would happen ?
Folks, When Configuring QOS Thresholds on the 3750 Switch, I noticed that the switch allows us to configure Threshold 2 to be a lower number than Threshold 1 What would happen in this scenario ? Would COS/DSCP levels assigned to Threshold 2 be dropped at 10% while those assigned to Threshold 1 be dropped at 20% or Something else ? HQ-SW(config)#mls qos srr-queue input thres HQ-SW(config)#mls qos srr-queue input threshold 1 20 10 HQ-SW(config)#do sh mls qos input-q Queue : 1 2 -- buffers :90 10 bandwidth : 90 10 priority : 10 0 threshold1: *20* 100 threshold2: *10* 100 -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing???
I have not played with it on cme but i know for sure it cannot be changed in ccm Sent from my phone On Aug 7, 2010, at 11:44 AM, Trying 2nd CCIE dukelon...@gmail.com wrote: Hi guys, I've noticed that when CUCM is up and cbarge is used, the words To conference appear on the phones. However, when the site goes to SRST, it shows To Barge instead of To Conference Can anyone help to make the display consistent in both CUCM and SRST mode? thanks and best regards On 8 August 2010 00:41, CCIE givemeccievoice2...@gmail.com wrote: The scenario specifically involves using auto-provision none. Have you tested and verified this? On Aug 7, 2010, at 9:35 AM, cisco voip voip.ccieci...@gmail.com wrote: That bug is for srst mode auto provision none.. for provision all, it should work The problem you are facing of having cbarge for split second is because you had single button cbarge when phones were registered to CUCM, disable that setting and make it normal cbarge, they will start to work in srst mode as well On Fri, Aug 6, 2010 at 5:05 PM, Ashar Siddiqui siddas...@gmail.com wrote: I am glad that the solution proposed by Cisco is exactly what I did months back after trying different solutions. Ash. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of CCIE Voice GMAIL Sent: 06 August 2010 03:13 To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing??? I thought I’d share this with everyone as this have been extremely frustrating for me. Apparently this is a known bug (well…recently known). CSCti11843 From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of MARSHALL, JODY C (ATTBCS) Sent: Wednesday, August 04, 2010 4:55 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cbarge and SRST (Again) (Again) and (Again) what am I missing??? I have read (several) post on this and have tested several different ways. None of which have I been able to make work. Can you please take a look and see if I am missing something. The first configuration is from auto-provision all. I had the phones registers then unregister bounced the router and register again. Cbarge does not work. I see the remote-in-use state for a second then disappears. I then registered the phones to CUCM and removed telephony-service reloaded the router and reconfigured telephony-service with auto-provision none with the second configuration posted. Cbarge does not work. 124-20.T5.bin telephony-service sdspfarm units 5 sdspfarm tag 2 sitebcfb conference hardware srst mode auto-provision all srst ephone template 1 srst ephone description SRST : Aug 03 2010 13:28:28 : Aug 03 2010 21:20:20 srst dn template 1 srst dn line-mode octo max-ephones 4 max-dn 30 preference 3 ip source-address 10.12.202.1 port 2000 system message CCIEVOICE time-zone 8 date-format dd-mm-yy voicemail 2220 max-conferences 8 gain -6 web admin system name administrator password ccievoice transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Aug 03 2010 21:20:26 ! R2#sho sccp SCCP Admin State: UP Gateway IP Address: 10.12.202.1, Port Number: 2000 IP Precedence: 5 User Masked Codec list: None Call Manager: 10.12.202.1, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 3 Trustpoint: N/A Call Manager: 10.12.200.21, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 2 Trustpoint: N/A Call Manager: 10.12.200.22, Port Number: 2000 Priority: N/A, Version: 6.0, Identifier: 1 Trustpoint: N/A Conferencing Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 10.12.202.1, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 8, Reported Max OOS Streams: 0 Supported Codec: g711ulaw, Maximum Packetization Period: 30 Supported Codec: g711alaw, Maximum Packetization Period: 30 Supported Codec: g729ar8, Maximum Packetization Period: 60 Supported Codec: g729abr8, Maximum Packetization Period: 60 Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: g729br8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 R2#sho ephone ephone-1[0] Mac:001A.6D57.021D 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:8 privacy:0
Re: [OSL | CCIE_Voice] VPIM issue ---- CUE to Unity
Take a look at detailed smtp traces on cuc. Please paste them and i can help. Sent from my phone On Aug 7, 2010, at 10:43 AM, Erwan Erwan e_er...@yahoo.com wrote: yes, i use prcotor lab mx record is on DNS --- On Sat, 8/7/10, Miron Kobelski findko...@gmail.com wrote: From: Miron Kobelski findko...@gmail.com Subject: Re: [OSL | CCIE_Voice] VPIM issue CUE to Unity To: Erwan Erwan e_er...@yahoo.com Cc: ccie_voice@onlinestudylist.com Date: Saturday, August 7, 2010, 3:00 PM do you have mx records on the dns server? regards -- Sent from my mobile device. On Aug 7, 2010 6:52 AM, Erwan Erwan e_er...@yahoo.com wrote: hi, I try to config VPIM by these steps : CUE --- Location ID : 34 Location Name : br2 Domain Name : cue.com Location ID : 212 Location Name : uc Domain Name : proctorlabs.com UC -- Networking , COnnection Location Display Name : cuc7-pub Host address : 10.10.210.13 SMTP Domain Name : proctorlabs.com Networking , VPIM Location Display name : br2 Dtmf Access ID : 34 Domain Name : cue.com IP address : 10.10.202.2,Allow Blind Addressing - DNS work ok for CCM, Unity and CUE (all pingable , CUE and UC) - Unity and CUE restart already - CUE and Unity voicemail work fine for local call (MWI and left message) However - call from CUE to UC , said the number u entered was not found, try differnt number - call from UC to CUE is OK , and left message to 34-3001 (but message is not there) And the debug in CUE (trace networking vpim all) , not generate anything, below -- BR2# sh trace buffer tail Press CTRL-C to exit... 4654 08/07 12:57:45.203 ACCN ENGN 0 Notifying Debug Task Aborted Can pls shed light on these, what went wrong? tks ___ 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
Re: [OSL | CCIE_Voice] VMWare server
I tried installing ccm 7 on vmware server 2 on top of ubuntu 10 (64 bit) couple of times and could never get install to complete successfully. On the other hand, I have used esxi and vmware workstation without any problems Sent from my phone On Jul 20, 2010, at 12:06 PM, akash patel akashapa...@yahoo.com wrote: I am planning to install CUCM Pub/Sub, UCCX, Unity Connection and Presence server on VMWare Server 2 on top of Ubantu. The reason to choose VMWare Ser 2 instead of ESXi because I was told that it works better with dynamics in order to simulate voice routers including FR and PSTN simulation. the server config I am looking in to is Intel Quad Processor 8 gig RAM two- 250G hard=drive, one for Pub and UCCX and other one for other servers Does any one has any suggestion, specifically to find out cheaper server with this or recommended hardware requirements? appreciate all feedback. thank you, ___ 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
Re: [OSL | CCIE_Voice] VMWare server
My intent was to use gns3. Sent from my phone On Jul 20, 2010, at 1:56 PM, Akash akashapa...@yahoo.com wrote: Thanks Pavan for sharing your experience. Were you using dynamics as well on the server? Do you know good deal for suffient hardware requirements? Akash Patel Presales Consultant On Jul 20, 2010, at 2:36 PM, Pavan pav.c...@gmail.com wrote: I tried installing ccm 7 on vmware server 2 on top of ubuntu 10 (64 bit) couple of times and could never get install to complete successfully. On the other hand, I have used esxi and vmware workstation without any problems Sent from my phone On Jul 20, 2010, at 12:06 PM, akash patel akashapa...@yahoo.com wrote: I am planning to install CUCM Pub/Sub, UCCX, Unity Connection and Presence server on VMWare Server 2 on top of Ubantu. The reason to choose VMWare Ser 2 instead of ESXi because I was told that it works better with dynamics in order to simulate voice routers including FR and PSTN simulation. the server config I am looking in to is Intel Quad Processor 8 gig RAM two- 250G hard=drive, one for Pub and UCCX and other one for other servers Does any one has any suggestion, specifically to find out cheaper server with this or recommended hardware requirements? appreciate all feedback. thank you, ___ 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
Re: [OSL | CCIE_Voice] VMWare server
Graham, Can you get me the following info. I have tons of issues with vmware server 2 Your kernel version Are you running 32 bit/64/pae? Vmware server version? Ccm version. I will try to replicate your setup. Thanks in advance for your help. Sent from my phone On Jul 20, 2010, at 2:18 PM, Graham Hopkins ghopk...@wolf-rock.co.uk wrote: I run two machines of the spec you mention with ubuntu. No problems with CUCM UC UCCX CUPS and a client XP machine using vmware server 2 except restore from snapshot is slow if you do too many at once and MVA tends to go very sluggish when servers are heavily loaded. Just generic hardware but I do spread the load over the two servers I do run dynamips but not usually at the same time Graham On 20 Jul 2010, at 19:56, Akash akashapa...@yahoo.com wrote: Thanks Pavan for sharing your experience. Were you using dynamics as well on the server? Do you know good deal for suffient hardware requirements? Akash Patel Presales Consultant On Jul 20, 2010, at 2:36 PM, Pavan pav.c...@gmail.com wrote: I tried installing ccm 7 on vmware server 2 on top of ubuntu 10 (64 bit) couple of times and could never get install to complete successfully. On the other hand, I have used esxi and vmware workstation without any problems Sent from my phone On Jul 20, 2010, at 12:06 PM, akash patel akashapa...@yahoo.com wrote: I am planning to install CUCM Pub/Sub, UCCX, Unity Connection and Presence server on VMWare Server 2 on top of Ubantu. The reason to choose VMWare Ser 2 instead of ESXi because I was told that it works better with dynamics in order to simulate voice routers including FR and PSTN simulation. the server config I am looking in to is Intel Quad Processor 8 gig RAM two- 250G hard=drive, one for Pub and UCCX and other one for other servers Does any one has any suggestion, specifically to find out cheaper server with this or recommended hardware requirements? appreciate all feedback. thank you, ___ 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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VMware server for home lab
You dont need rhel. You can install vmware esxi which install directly on the hardware. I am not familiar with t310, personally i use mcs 7845h2 servers for my lab with 6g ram and a nas for disk Sent from my phone On Jul 9, 2010, at 7:23, akash patel akashapa...@yahoo.com wrote: This might have been covered in various posts in the past, but I could not find particular answer from the archieve. I am planning to set up CUCM/Unity Connection/Presence and UCCX7 on VMWare using Dell T310. Can you please help me with the minimum spec that do I need to have on Dell T310 (or on equivelent server if that server is not compitable with VMWare)? As far operating system, do we have to buy REHL, or any alternative to it? I appreciate any help setting up the UCM cluster on VMWare? Thank you ___ 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
[OSL | CCIE_Voice] MLPP anyone ?
IS anybody out there using MLPP on either callmanager or CME in production. If you are using it, what criteria do you use for preemption. I understand DoD uses it but trying to see if anybody is using it on the Commercial / Enterprise front. Sent from my phone ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Why are my SIP Phones registering to the PUB
Thats really wierd from ccm perspective Can you check your dbreplication status, looks like the sip phones did not get replicated to sub. You can check it from rtmt- database summary Sent from my phone On Jun 30, 2010, at 5:55, Duncan Hamilton-Walker dun...@rosethorn.plus.com wrote: Update to my Problem; If i turn off the CCM service on the Publisher .. the SIP phones are rejected registration to the SUB.. and will not register at all. Anyone have any ideas please.. thanks From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Duncan Hamilton-Walker Sent: 30 June 2010 00:21 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Why are my SIP Phones registering to the PUB Dont understand only my SIP phones are registering to the PUB, when the SUB is set are the primary Server in the Cluster Cisco 7940 SEP000D65ECB991 BR1 PHN 2 Default SCCP Registered with 10.10.210.11 10.10.201.252 Cisco 7941G-GE SEP001AE2BCE80F HQ PHN 2 Default SIP Registered with 10.10.210.10 10.10.200.248 Cisco 7941 SEP001B0CDBB104 BR1 PHN 1 Default SIP Registered with 10.10.210.10 10.10.201.254 Cisco 7961G-GE SEP001B2AC6A44A HQ PHN 1 DefaultSCCP Registered with 10.10.210.11 10.10.200.251 Cisco IP Communicator SEP02004C4F4F50HQ CIPC DefaultSIP Registered with 10.10.210.10 10.10.210.250 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 9.0.830 / Virus Database: 271.1.1/2967 - Release Date: 06/29/10 07:35:00 ___ 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
Re: [OSL | CCIE_Voice] CUCM 7.0.1 , Subscriber does not show up in CM group configuration Window
You will need to start ccm service in sub for it to show up in cm group Sent from my phone On Jun 29, 2010, at 8:40, jeremy co jeremy.coo...@gmail.com wrote: Hi, I installed both Subscriber and publisher both with same version of 7.0.1 on Vmware. ( both added as CUCM not CUCMBE). Show network cluster shows both servers. I added SUB as a server and Cisco Unified CM ,but it will not show up in Cisco Unified CM group when I try to add it to the group. It might be a simple thing but because my background is in version4 , probably I missed it. Cheers, jeremy ___ 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
Re: [OSL | CCIE_Voice] Qos 3750 --- output queue
You are mistaken. Its not changing the threshold. The first line means that cos 5 will be assigned to q1 upto its third threshold. The 3750 command reference is a better book if you want to ubderstand the parameters Sent from my phone On Jun 29, 2010, at 18:52, Erwan Erwan e_er...@yahoo.com wrote: hi all, I am reading the 3750 QoS , and confuse with the threshold in output queue. When I run int f0/4 auto-Qos cisco-phone ! it generete following : mls qos srr-queue output cos-map queue 1 threshold 3 5 mls qos srr-queue output cos-map queue 2 threshold 3 3 6 7 mls qos srr-queue output cos-map queue 3 threshold 3 2 4 mls qos srr-queue output cos-map queue 4 threshold 2 1 mls qos srr-queue output cos-map queue 4 threshold 3 0 which is threshold 3, but in sRND they said , threshold 3 is 100% by default and it cannot be changed 1. Why it give us threshold 3 , if not supposed to be change ? 2. What we should you use, between threshold 1 and 2 , and 3 (as they have 3 threshold) ? Tks ___ 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
Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2
Daniel, Before you go check replication, check to see if cups is even requesting the correct config xml file. Replication could only be a problem when cups tries to register to ucm and ucm rejects the register request Sent from my phone On Jun 26, 2010, at 16:52, Roger Källberg roger.kallb...@cygate.se wrote: Hi Daniel, It's not always that you can trust the information given by the show perf query class Number of Replicates Created and State of Replication command. One easy thing that you can do to verify if you have a db repl problem is to put your phones, or any other device, in a pub only enviroment. If all works then you know that the sub didn't have the correct info. And in thet case you need to repair the db replication by utils debreplication stop ,1 on sub, then when promtpt returns on the sub put in the same command on pub). When the prompt returns on the pub use utils dbreplication repair all on the pub. This will take some time to complete. Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se Från: Daniel Berlinski [dberlin...@gmail.com] Skickat: den 26 juni 2010 23:44 Till: Roger Källberg Kopia: kobel; osl osl Ämne: Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Hi Kobel Owner was setup for the mobility section to work. It is in there. Hi Roger The way I know how to verify dbReplication is: admin:show perf query class Number of Replicates Created and State of Replication ==query class : - Perf class (Number of Replicates Created and State of Replication) has instances and values: ReplicateCount - Number of Replicates Created = 412 ReplicateCount - Replicate_State= 2 My reading of this is that is all good. Am I right? Well, I have rebooted this many times already so I think I will just upgrade the client and see what happens. Will update you all. Thnaks 2010/6/27 Roger Källberg roger.kallb...@cygate.se Try to verify if db replication is ok, if not, fix that. You might also want to restart the CTI Manager on both sub and pub. Brgds, Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Direkt: +46108787498 Växel: +46108787400 roger.kallb...@cygate.se Från: Daniel Berlinski [dberlin...@gmail.com] Skickat: den 26 juni 2010 23:18 Till: kobel Kopia: osl osl Ämne: Re: [OSL | CCIE_Voice] Presence Issues Softphone mode in CUPC Lab 5 Volume 2 Thanks for your replies. Primary extension is assigned to end user and that extension matches with the line number of CUPC. The users are assigned to the Standard CCM End Users, and CTI Enabled groups What is the version of CUPC you guys use? Thank you On Sun, Jun 27, 2010 at 9:03 AM, kobel findko...@gmail.com wrote: See if adding the end user to Standard CUCM users group in CUCM helps regards On Sat, Jun 26, 2010 at 10:30 PM, Daniel Berlinski dberlin...@gmail.com wrote: Hello all Out of ideas now after troubleshooting extensively a Presence problem. I'm finalizing lab 5 Vol 2 and I can't get CUPC to download its TFTP configuration file from CUCM and for that reason I do not even see the option for selecting softphone control Any help is appreciated. What I have and what I've done is the following: 1- Cretaed device named UPC+12alphanumeric characters, in my case UPCTERRELLEPRYO, associated its line to the enduser 2- End user configured with primary extension, associated with UPC phone device, CTI control of its devices and group association to CTI enabled group. 3- Still in CUCM, Capabilities Assignment was provided for the user. 5- In CUPS, Application-Cisco Unified Personal Comm - Settings I have provided the IP addresses for TFTP server primary and secondary Presence status is working fine and Deskphone control works fine as well. My problem here is that the CUPC SIP phone is not getting in Show Server Health a tftp file to download. It displays the IP addres of TFTP primary and seoondary but it does not display the UPCTERRELLEPRYO.CNF.XML file to download. To troubleshoot this I have done the following: 1- Went in DOS and did a tftp -i 10.10.210.10 get UPCTERRELLEPRYO.CNF.XML and files downloaded OK so there is no network issues here. Inside the file I saw references to TFTP server as IP addresses so no name resolution issues either. 2- Ran Wireshark and did not see any attempts from the client machine to register with CUCM via SIP so client is not even attempting to register. In fact nothing displays when I filter the capture by the CUCM ip addresses. 3- Listing my cupc users by clicking in CUPS, application, Cisco Unified personal comm, user settings I see my users listed there but under the column Client Type nothing displays 4- Created another UPC device
Re: [OSL | CCIE_Voice] Vol2 Lab1, 4.2 Supplementary services
You will need a hardware mtp (i.e dsp / ios sw ) not the one provided by ucm Sent from my phone On Jun 26, 2010, at 18:47, David Lee d16...@gmail.com wrote: Hello, I am at a lost. I got most of this section working. I can resume a call if the hold was initiated by an UCM phone or the CME SCCP phone. However, I cannot resume if the hold was initiated by the CME SIP phone. Any one have ideas what can be looked at to troubleshoot? (Software MTP is configured and active during the call. Codecs are also right.) Thanks, -Dave ___ 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
Re: [OSL | CCIE_Voice] service redundancy
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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] VPIM problem
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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] NTP on UCCX
When configuring NTP on the UCCX server, i see two approaches when asked to configure NTP. Which way would you go ? - Configure NTP on the Windows OS (Using windows registry hack) (More Involved) - Configure NTP on the UCCX app itself during Integration. (Faster Seems to work) (Disadvantage : windows time does not seem to sync up since only app is synced) -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP on UCCX
you would also need clock timezone and summertime (if asked) On Fri, Jun 11, 2010 at 10:22 AM, Dani Bug daniyal.vo...@gmail.com wrote: go with option 2 NTP UCCX also i just want to confirm in NTP on R1 R2 and R3 which commands we required lab perspective view ... ntp server x.x.x.x wht else command we required to configure Thx Dani On Fri, Jun 11, 2010 at 9:28 AM, Pavan K pav.c...@gmail.com wrote: When configuring NTP on the UCCX server, i see two approaches when asked to configure NTP. Which way would you go ? - Configure NTP on the Windows OS (Using windows registry hack) (More Involved) - Configure NTP on the UCCX app itself during Integration. (Faster Seems to work) (Disadvantage : windows time does not seem to sync up since only app is synced) -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Better Voice Lab Locations
In SJ they generally take you to the cafe. In RTP they generally get the food catered and serve it inside an ancillary room. (atleast they used to) On Fri, Jun 11, 2010 at 10:26 AM, Jeff Price (jeffpric) jeffp...@cisco.comwrote: I believe it depends on your location, but normally they walk you to a local Cisco cafeteria with a voucher for your lunch (up to a certain price). -Original Message- From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Jon1992 Sent: Friday, June 11, 2010 4:10 AM To: Amp; ccie voice Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations During lunch are we stuck in the lab area or can we go and buy? -- From: Amp amccar...@cciequest.com Sent: Thursday, June 10, 2010 11:01 PM To: ccie voice cci...@gmail.com Cc: ccie_voice@onlinestudylist.com; Mouhammad Nasser engnasse...@hotmail.com Subject: Re: [OSL | CCIE_Voice] Better Voice Lab Locations No not based on lunch. With the longer lunch time I will be able to have some time to think about what I have completed, what I need to complete, and if I need to change anything that I have done. Quoting ccie voice cci...@gmail.com: @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 ___ 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 -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[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 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] NTP on UCCX
you generally use them if you are doing NTP authentication and using ACL to control who your NTP peers are. Doesn't hurt to add but technically they are not required. I usually add them -Pavan On Fri, Jun 11, 2010 at 10:31 AM, Dani Bug daniyal.vo...@gmail.com wrote: thx i forgot to add these cmd but wondering if we need to add ntp source loopback0 ntp server x.x.x.x source int fa0/0.100 these cmd also required ?? On Fri, Jun 11, 2010 at 11:27 AM, Pavan K pav.c...@gmail.com wrote: you would also need clock timezone and summertime (if asked) On Fri, Jun 11, 2010 at 10:22 AM, Dani Bug daniyal.vo...@gmail.comwrote: go with option 2 NTP UCCX also i just want to confirm in NTP on R1 R2 and R3 which commands we required lab perspective view ... ntp server x.x.x.x wht else command we required to configure Thx Dani On Fri, Jun 11, 2010 at 9:28 AM, Pavan K pav.c...@gmail.com wrote: When configuring NTP on the UCCX server, i see two approaches when asked to configure NTP. Which way would you go ? - Configure NTP on the Windows OS (Using windows registry hack) (More Involved) - Configure NTP on the UCCX app itself during Integration. (Faster Seems to work) (Disadvantage : windows time does not seem to sync up since only app is synced) -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- - Pavan -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUPC and presence status
Thanks angel. Sent from my phone On Jun 11, 2010, at 10:55 AM, Angel Perez gorr...@hotmail.com wrote: 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. Si gn up now. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE - Destination unreachable error
Can you ping it. Did you issue a no shut? Sent from my phone On Jun 11, 2010, at 1:56 PM, Steve Denney (stdenney) stden...@cisco.com wrote: Working on Vol 2 Lab 2 Question 8.2. When trying to session into the CUE, I get this error: BR2-RTR#service-module service-Engine 0/0 sess Trying 10.10.202.1, 2194 ... % Destination unreachable; gateway or host down Module status looks good: BR2-RTR#service-module service-Engine 0/0 status Service Module is Cisco Service-Engine0/0 Service Module supports session via TTY line 194 Service Module is in Steady state Service Module heartbeat-reset is enabled Getting status from the Service Module, please wait.. Cisco Unity Express 7.0.1 CUE Running on AIM IP route looks good: BR2-RTR#sh ip route 10.10.202.2 Routing entry for 10.10.202.2/32 Known via static, distance 1, metric 0 (connected) Routing Descriptor Blocks: * directly connected, via Service-Engine0/0 Route metric is 0, traffic share count is 1 Config is plain enough: interface Service-Engine0/0 ip unnumbered Vlan400 service-module ip address 10.10.202.2 255.255.255.0 service-module ip default-gateway 10.10.202.1 Have reloaded the router, and did a ser ser 0/0 reset – still no jo y. What obvious thing am I missing? cheers, sd ___ 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
[OSL | CCIE_Voice] CUE integration with CCM problems
If i place all my CTI ports in the NULL partition everything works, If i put
them in PT-VM i get BUSY (CTI rejecting call).
Any additional CSS settings needed ?
=
Placed all my CTI Ports in PT-VM
Placed all my CTI Routepoints in Null partition.
CSS for Routepoints and Ports is CSS-VM { PT-VM + PT-Phones }
Doesn't work
==
--
- Pavan
___
For more information regarding industry leading CCIE Lab training, please visit
www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE integration with CCM problems
No it does not have that partition.
Shouldnt the css of the route point be used when the call is
redirected to cti port?
Sent from my phone
On Jun 9, 2010, at 8:58 PM, bkvalent...@gmail.com bkvalent...@gmail.com
wrote:
Do your phones and gateways have the pt-vm partition in their css?
Brian
Sent from my Verizon Wireless Phone
- Reply message -
From: Pavan K pav.c...@gmail.com
Date: Wed, Jun 9, 2010 9:14 pm
Subject: [OSL | CCIE_Voice] CUE integration with CCM problems
To: osl osl ccie_voice@onlinestudylist.com
If i place all my CTI ports in the NULL partition everything works,
If i put
them in PT-VM i get BUSY (CTI rejecting call).
Any additional CSS settings needed ?
=
Placed all my CTI Ports in PT-VM
Placed all my CTI Routepoints in Null partition.
CSS for Routepoints and Ports is CSS-VM { PT-VM + PT-Phones }
Doesn't work
==
--
- Pavan
___
For more information regarding industry leading CCIE Lab training, please visit
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Re: [OSL | CCIE_Voice] CUE integration with CCM problems
The route point is in the null partition and its css contains the partition of cti ports and regular phones Sent from my phone On Jun 9, 2010, at 9:10 PM, Brian Valentine bkvalent...@gmail.com wrote: What pt is the route point in? Your VM profile has a pilot and css associated to it. Does it contain the route point partition? Brian On Jun 9, 2010 10:02 PM, Pavan pav.c...@gmail.com wrote: No it does not have that partition. Shouldnt the css of the route point be used when the call is redirected to cti port? Sent from my phone On Jun 9, 2010, at 8:58 PM, bkvalent...@gmail.com bkvalent...@gmail.com wrote: Do your phon... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUE integration with CCM problems
Thank you.
Will reset the cue module and try again tomorrow.
Sent from my phone
On Jun 9, 2010, at 9:51 PM, wolfsrudel wolfsru...@gmail.com wrote:
according to what you're saying it *should* work, or i'm too tired
to reckon.
the only entity that should be seen from the phones' perspective (in
this case) is the cti rp, in turn it would have a css such that it
could reach both cti ports and phones (and likely mwi).
css_internal { pt_phones }
css_voicemail { pt_voicemail, pt_phones }
cti_rp / pt_phones = css_voicemail
cti ports / pt_voicemail = css_voicemail
mwi / pt_voicemail = css_voicemail
works like a charm.
On Wed, Jun 9, 2010 at 10:14 PM, Pavan K pav.c...@gmail.com wrote:
If i place all my CTI ports in the NULL partition everything works,
If i put
them in PT-VM i get BUSY (CTI rejecting call).
Any additional CSS settings needed ?
=
Placed all my CTI Ports in PT-VM
Placed all my CTI Routepoints in Null partition.
CSS for Routepoints and Ports is CSS-VM { PT-VM + PT-Phones }
Doesn't work
==
--
- Pavan
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Re: [OSL | CCIE_Voice] Lan QOS Scenario
Looks good as farvas i can tell. Normally you would also enabl priority-queue on the interface Sent from my phone On Jun 8, 2010, at 8:20 PM, jammer jones jammerjone...@gmail.com wrote: Trying to understand this a little better. Cisco's documentation is not written in very clear english. Very frustrating trying to understand the threshold values as well as the shape versus share bandwidth values. QOS. Cos 5 for queue 1 queue 2 queue 3 queue 4 0 similar to lab 2 . Queue one has the 25% of the bandwidth. other bandwidth is shared as 30 40 30. If the queue 2 is saturated by 60% then the cos 4 has to be dropped. Here is what I think it is. Can someone please correct me if i am wrong and provide any positive feedback. ! mls qos srr-queue output cos-map queue 2 threshold 2 4 !maps cos 4 to queue 2 and threshold 2 mls qos srr-queue output cos-map queue 4 threshold 1 0 !maps cos 0 to queue 4 mls qos queue-set output 1 threshold 2 40 60 100 200 ! when queue 2 threshold 2 exceeds 60% cos packets with cos 4 will be dropped mls qos ! ! interface GigabitEthernet1/0/1 description Office_912_lab_a switchport access vlan 48 switchport mode access switchport voice vlan 51 srr-queue bandwidth share 1 30 40 30 ! sets queues 2 - 4 to 30 40 30 srr-queue bandwidth shape 4 0 0 0 ! sets queue 1 to 25% of the link mls qos trust cos spanning-tree portfast ___ 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
Re: [OSL | CCIE_Voice] VPIM error on CUE (554 Bad Sender's System) [Solved]
Had to change domain name on Unity connection under SMTP settings and reboot the box. Restarting the Conversation Manager service (as instructed by the GUI) didn't make any difference. -Pavan On Sat, Jun 5, 2010 at 7:41 PM, Pavan K pav.c...@gmail.com wrote: Trying VPIM Sending messages from CUE to UnityConnection works perfectly. Messages from UnityConnection to CUE get an error message and generate a NDR (non-delivery receipt) Looking through the SMTP traces, i see a 554 error. (Screenshot attached). Anybody seen this before ? -- - Pavan -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] How to send a secure message in Unity Connection ?
I can send messages as Private / urgent. How does one send a secure message ? I haven't been able to find any useful docs on this yet ! -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Gatekeeper 4 Digit call
Could you post gk config debugs for both the calls? Sent from my phone On Jun 6, 2010, at 5:36 PM, Dani Bug daniyal.vo...@gmail.com wrote: Hi Guys, i have issue with gatekeeper 4 digit call site to site i don't know why my call some times going and sometime not for example when i call from BR2 to HQ using 4 Digit it's ringing but when i hangup and then dial again i am hearing busy/engage tone . HQ to BR2 is working prefect and I remember Ii had this issue before but i forgot how to resolve . any advise will be appreciated Thx Dani ___ 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
Re: [OSL | CCIE_Voice] CME 7.0 Presence caller-list is not working ...
As others have pointed out it works fine on 7.0. Just do a create cnf-files and reset if your SK phones dont show presence in call-list. -Pavan On Sat, Jun 5, 2010 at 5:41 AM, Angel Perez gorr...@hotmail.com wrote: 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. Get it now.https://signup.live.com/signup.aspx?id=60969 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] vol 2 lab 1: gatekeeper and H.323 GW on the same router
in your existing config, remove the h323gw bind source interface command Sent from my phone On Jun 4, 2010, at 9:50 AM, kobel findko...@gmail.com wrote: I described from CUCM perspective: incoming calls - call from GW to CUCM. On Fri, Jun 4, 2010 at 4:43 PM, Ashar Siddiqui siddas...@gmail.com wrote: You are talking about inbound calls or outbound calls from the gateway? Sorry it’s not clear for me. ___ 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
Re: [OSL | CCIE_Voice] MVA works but I don't hear prompts
Angel, I use my own gear with vmware esxi and mva works great. I have tried it atleast 10-15 times and never had a problem with it. Sent from my phone On Jun 3, 2010, at 3:33 AM, Angel Perez gorr...@hotmail.com wrote: 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. Si gn 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. ___ 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
Re: [OSL | CCIE_Voice] Attendant console link ?
So in the lab exam do we have a cisco unified attendant console server ? -pavan Sent from my phone On Jun 1, 2010, at 3:29 AM, kerboute kerboute naoufal.kerbo...@cbi.ma wrote: attendant console is end of life for CUCM 7 You need Cisco unified attendant console server On 05/31/2010 11:28 PM, Pavan K wrote: Does anybody have a link to / copy of the attendant console plugin ? --Thanks in advance. - Pavan ___ 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
Re: [OSL | CCIE_Voice] Attendant console link ?
Looks like our 4x machine got scrapped. Does anybody have the plugin handy and willing to provide me a copy if I can prove that I have legitimate access to cisco software. Sent from my phone On Jun 1, 2010, at 6:01 AM, Pavan pav.c...@gmail.com wrote: Correct. That was exactly why I was looking for it. I just found this link which offers more info. http://ciscoblog.globalknowledge.com/2009/07/13/cucm-7-and-attendant-console/ I Will pull the AC plugon from 4.x and try it out. Sent from my phone On Jun 1, 2010, at 5:45 AM, Roger Källberg roger.kallb...@cygate.se wrote: AC is still a testable topic, see this url https://supportforums.cisco.com/message/3012407#3012407 . I guess that's why he asked for it. Roger Källberg Consultant Cygate AB Från: kerboute kerboute [naoufal.kerbo...@cbi.ma] Skickat: den 1 juni 2010 10:29 Till: Pavan K Kopia: ccie_voice@onlinestudylist.com Ämne: Re: [OSL | CCIE_Voice] Attendant console link ? attendant console is end of life for CUCM 7 You need Cisco unified attendant console server On 05/31/2010 11:28 PM, Pavan K wrote: Does anybody have a link to / copy of the attendant console plugin ? --Thanks in advance. - Pavan ___ 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
Re: [OSL | CCIE_Voice] Attendant console link ?
Thanks guys for the response. I got access to the file. I am using this for the lab exam so cant use third party. -pavan On Tue, Jun 1, 2010 at 1:43 PM, johan_claes fb922...@skynet.be wrote: better to buy peter connect for reception, cheaper and better, johan claes ccie#5437 - Original Message - *From:* kerboute kerboute naoufal.kerbo...@cbi.ma *To:* Pavan K pav.c...@gmail.com *Cc:* ccie_voice@onlinestudylist.com *Sent:* Tuesday, June 01, 2010 10:29 AM *Subject:* Re: [OSL | CCIE_Voice] Attendant console link ? attendant console is end of life for CUCM 7 You need Cisco unified attendant console server On 05/31/2010 11:28 PM, Pavan K wrote: Does anybody have a link to / copy of the attendant console plugin ? --Thanks in advance. - Pavan ___ 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 -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] MVA on SIPGW
Has any body tried this ? -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Attendant console link ?
Does anybody have a link to / copy of the attendant console plugin ? --Thanks in advance. - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] MVA on SIPGW
When using SIPGW and trying to transfer a call, The INVITE with the diversion header reaches CCM but is getting blocked in there due to a Top level domain mismatch. Wondering if anybody got it to work ? -Pavan On Mon, May 31, 2010 at 1:55 PM, Pavan K pav.c...@gmail.com wrote: Has any body tried this ? -- - Pavan -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] High Traffic ?
Looks like your ccm ran into code yellow. You probably have a routing loop somewhere. Sent from my phone On May 30, 2010, at 11:17 AM, Erwan Erwan e_er...@yahoo.com wrote: hi, Does anyone experience this - Call from HQ 5001 to 5600 (VM) said High Traffic Try Again Later I checked in cisco web, this can cause by lots of hunt group and loop in Hunt group How to disable the loop and make sure if no loop in Hunt Grup and VM ? (As I only config VM thru wizard) tks ___ 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
Re: [OSL | CCIE_Voice] High Traffic ?
Easiest way is to fire up rtmt real time sdi trace for the active call processing node , make one test call and see how the call is getting routed by looking at the digit analysis info. Sent from my phone On May 30, 2010, at 2:08 PM, Erwan Erwan e_er...@yahoo.com wrote: Ic, how to check the loop ? what to verify ? I am using proctor lab, and call to VM 5600, showing this but other call is fine --- On Mon, 5/31/10, Pavan pav.c...@gmail.com wrote: From: Pavan pav.c...@gmail.com Subject: Re: [OSL | CCIE_Voice] High Traffic ? To: Erwan Erwan e_er...@yahoo.com Cc: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Date: Monday, May 31, 2010, 2:35 AM Looks like your ccm ran into code yellow. You probably have a routing loop somewhere. Sent from my phone On May 30, 2010, at 11:17 AM, Erwan Erwan e_er...@yahoo.com wrote: hi, Does anyone experience this - Call from HQ 5001 to 5600 (VM) said High Traffic Try Again Later I checked in cisco web, this can cause by lots of hunt group and loop in Hunt group How to disable the loop and make sure if no loop in Hunt Grup and VM ? (As I only config VM thru wizard) tks ___ 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
Re: [OSL | CCIE_Voice] Trust COS or DSCP on uplink to router ?
Yes you can, as long as you know the dscp is correctly marked. Sent from my phone On May 30, 2010, at 1:45 PM, Mike Brooks 2xcci...@gmail.com wrote: So I understand that, COS bits are set in the 802.1p field in an 802.1q encapsulated trunk but does it ever make sense to trust COS on an uplink to a router-on-a-stick type interface ? Shouldn't you always just trust DSCP on an uplink to a router and never trust cos ? Regards, Mike ___ 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
[OSL | CCIE_Voice] E164 normalization on SIP Trunk for inbound calls
Folks, For inbound calls, we can normally prefix/strip digits on the H323 / MGCP gateway page based on the calling number type (subscriber / national / ... ) When a call comes in through a SIP trunk, we lose the number type (due to SIP limitations). Does anybody have a good idea to normalize / re-classify the incoming call (subscriber / national ) in this scenario ? I am using CCM 7.0 -- - Pavan ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Problem Lab1 Vol2 Question 4.2 CUBE
Uncheck wait for farend h245 caps on h225 trunk. Sent from my phone On May 28, 2010, at 7:52 PM, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi guys, I'm having a strange behaviour on my lab, I'm working on question 4.2 I'v configured everything GK, GK and gateway. I can make calls from UCM to UCME and vice versa however when I call from UCM to UCME and answer call I can see call connected on BR2PHN2 but on UCM phones Call still ringing and then disconnect. Note: When I saw the call answered on BR2PHN2 i hear i bip is like the call on hold. Any Ideas?? Regards Naoufal ___ 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
Re: [OSL | CCIE_Voice] RE : Problem Lab1 Vol2 Question 4.2 CUBE
When interfacing with cube, you have to force ccm to send tcs as cube doesn't do it. By default cme would initiate tcs when talking to ccm. Tcs = terminal capability set. Sent from my phone On May 29, 2010, at 12:09 AM, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi Pavan, Thank you it works, what is that mean? Message d'origine De: Pavan [mailto:pav.c...@gmail.com] Date: sam. 5/29/2010 3:47 À: naoufal.kerboute Cc: ccie_voice@onlinestudylist.com Objet : Re: [OSL | CCIE_Voice] Problem Lab1 Vol2 Question 4.2 CUBE Uncheck wait for farend h245 caps on h225 trunk. Sent from my phone On May 28, 2010, at 7:52 PM, naoufal.kerboute naoufal.kerbo...@cbi.ma wrote: Hi guys, I'm having a strange behaviour on my lab, I'm working on question 4.2 I'v configured everything GK, GK and gateway. I can make calls from UCM to UCME and vice versa however when I call from UCM to UCME and answer call I can see call connected on BR2PHN2 but on UCM phones Call still ringing and then disconnect. Note: When I saw the call answered on BR2PHN2 i hear i bip is like the call on hold. Any Ideas?? Regards Naoufal ___ 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
