Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 53, Issue 12
Hi, Pls send the running config from both the routers. Things to check: 1. Since you are getting unallocated number first thing to check is the outgoing sip dial-peer (with session protocol sipv2) pointing to the call-manager) 2. do u have the following commands under voice service voip on the routers: Allow connection h323 to h323 Allow connections h323 to sip Allow connection sip to h323 Allow connections sip to sip 3. I guess u must be using g729 codec from ccm to cme. In this case call manager MTP will not work and u will need IOS software MTP with G729 codec. Include MTP in the MRGL of sip trunk. Moreover make sure that ur call uses codec G729 throughout. For this to work, u need to make sure that u are not hitting default dial-peer from either direction. Configure an incoming dial-peer that gets hit from both sides i.e CM to CME and CME to CM. 4. Next step collect ccm traces to see what is going on. Thanks and regards, Vishal Preenja ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 53, Issue 7 - Music on hold from router flash (Piano music
Hi, If you want to play music from Router's flash, first off u shd stop the MOH RTP stream to reach remore router from ccm. 3 ways to achieve this are: 1. reduce max hops on the MOH server to 1 2. remove ip pim commands on the serial interfaces that connects HQ and that remote sire 3. access list Then u need ti make sure that you have the following cmds under telephony service/call-manager fallback Telephone-service Moh music-on-hold.au multicast moh 239.1.1.1 port 16384 route lo0 voice vlan ip ip pim command under all the interfaces where u want to direct multicast traffic. Which interface are you using for sccp phones to register i.e I this command under telephony service - ip source ip address port 2000. If its lo0 , the try using voice vlan ip and then try. When playing moh from flash on the remote router, run the following show command to verify from where MOH is being played: show call active voice | include RemoteMedia RemoteMediaIPAddress=239.1.1.1 RemoteMediaPort=16384 Thanks and Regards, Vishal Preenja From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Friday, July 02, 2010 9:03 PM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 53, Issue 7 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: SRST (Ashar Siddiqui) 2. Music on hold from router flash (Piano music) (Afzal Bhutta) 3. Re: SRST (kobel) 4. Re: SRST (Ashar Siddiqui) -- Message: 1 Date: Fri, 2 Jul 2010 15:29:17 +0100 From: Ashar Siddiqui siddas...@gmail.com To: 'sean hurricane' shurric...@gmail.com, ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] SRST Message-ID: 005c01cb19f2$f6b8d740$e42a85...@com Content-Type: text/plain; charset=us-ascii Sean, Do srst auto-prov none and then just create ephones (as many as required) and put the following in there: Ephone 1 No privacy ! Ephone 2 No privacy ! You will need to do all the basic requirements for Cbarge like conference hardware, sdspfarm units etc and configuring dspfarm with telephony-service address on third priority if the requirement is that Cbarge is working during normal mode as well. Give it a go and let us know how it works. Ash From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of sean hurricane Sent: 02 July 2010 14:59 To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] SRST I have an SRST requirement which states the following: A. when in SRST phone should should be able to utilize cbarge and Park and live-record B. Your are not allowed to allowed to have information for learned phones in running configuration (no manual ephone configuration) C. Phones must support same number of incoming and outgoing calls in SRST as it does in UCCM. What is the best way to accomplish this without using srst auto-provision all which clearly does not meet the requirementI have tried to use auto-provison none but then cbarge does not work. any thoughts? -- next part -- An HTML attachment was scrubbed... URL: http://onlinestudylist.com/pipermail/ccie_voice/attachments/20100702/76d6cf 06/attachment-0001.html -- Message: 2 Date: Fri, 2 Jul 2010 10:52:44 -0400 From: Afzal Bhutta azhar.bhu...@gmail.com To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Music on hold from router flash (Piano music) Message-ID: aanlktik0iogc1taute6gcsm6c4hdmynxu186s8k9h...@mail.gmail.com Content-Type: text/plain; charset=windows-1252 Sorry Folks not providing details in first attempt. Thanks for all and special thanks to Matthew Berry and Randall Saborio for their interest and figured out this issue. Let?s make thing more understandable. I am working in my home lab. I am trying to spoof call manager. My target is to get music from router flash for HQ and for siteB not from call manager. Call manager is configured as I explain below. I have adjusted my ServerMax Hops to 15 for the M.cast it is working fine for HQ and SiteB but I am hearing music from Call manger not from router flash (Piano music) What I performed on the routers. I have enabled Muticast-routing on HQ and site B I have use IP pim dense mode commands on Voice-vlan interfaces, Loopback0, and serial interfaces which
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68
Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] IPCCx 7.x
Hi Thomas, Are ladies also allowed to reply? :-) Pls check the hardware compatibility matrix below: http://www.cisco.com/en/US/docs/voice_ip_comm/cust_contact/contact_center/cr s/express_compatibility/matrix/crscomtx.pdf HP DL380 G3 is not supported with IPCC 7.x onwards. Thanks and Regards, Vishal Preenja _ From: Thomas Koch [mailto:koch1...@comcast.net] Sent: Friday, March 12, 2010 1:50 PM To: 'Vishal Preenja'; 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] IPCCx 7.x Gent's, Has anyone had any issue loading IPCC express 7.x on non Cisco hardware..? I tried to load it on an HP DL380 G3 (MCS-7845 equivalent) and it said non Cisco hardware install will now stop T E-mail:thomas.k...@compucom.com From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Vishal Preenja Sent: Friday, March 12, 2010 12:46 PM To: 'Omotayo' Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Yes, I am referring to MRG that is there in the MRGL of the trunk. Just check that you should not have a MTP allocated for that call from CCM to CME. You can verify by making a call and then check in RTMT that whether Transcoder is being invoked or MTP. Thanks and Regards, Vishal Preenja From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 1:25 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello, Also what do you mean by MTP above transcoder. Are you reffering to the MRGL? On Fri, Mar 12, 2010 at 7:09 PM, Vishal Preenja vpree...@cisco.com wrote: It will work as I described. Can you send me the detailed ccm traces from all servers in the clusters or get me access of your box. Thanks and Regards, Vishal Preenja _ From: Omotayo [mailto:adefilabi...@gmail.com] Sent: Friday, March 12, 2010 12:33 PM To: Vishal Preenja Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 68 Hello Visha, when i did it as you described. when sccp phone call sip phone on the cme, i get a reorder tone when sip phone on the cme calls the sccp phone on the hq, it disconnects when hwq phone is picked and the sip phone continues to ring How can this be fixed On Fri, Mar 12, 2010 at 4:57 PM, Vishal Preenja vpree...@cisco.com wrote: Hi, While making a call from the UCM to CME Sip phone ( because you have G711ulaw configured in the voice register pool), if you are getting disconnect then you will need transcoder in the MRGL of Sip trunk/ GK trunk and also make sure that you don't have MTP listed above transcoder. If there is MTP configured above transcoder, it will be allocated when transcoder is requested and the call will fail. Thanks and regards, Vishal Preenja. Hi Jeff, Would you please tell us more about the call flow and the end to end codec requirements for this call. If doing g.729 over the wan, and your sip phone is using g.711 you should transcode at br2, Please let us know, Thanks, On Thu, Mar 11, 2010 at 8:29 PM, Jeff Cotter jcot...@voxns.com wrote: Can anybody tell me if a PVDM2-32 can be used as a hardware transcoder on UCM. Can?t seem to get a call from Call Manager to CME sip phone working. I can call from CME to UCM but not the other way around. Rings but disconnects when answered. Transcoder shows registered in Call manager. Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com/ ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 49, Issue 31
Try with the demoted called number 1#3002 Seems like translation profile under voip dial-peer 200 is not doing its job of converting 1#3002 to 3002 Mar 4 01:44:01.043: //188/004C02671500/CCAPI/ccCallDisconnect: Cause Value=1, Call Entry(Responsed=TRUE, Cause Value=1) Cause Value=1 means unassingned number ( that's obvious because you don't have 1#3002 as DN Thanks and Regards, Vishal Preenja From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of ccie_voice-requ...@onlinestudylist.com Sent: Thursday, March 04, 2010 8:36 AM To: ccie_voice@onlinestudylist.com Subject: CCIE_Voice Digest, Vol 49, Issue 31 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: Calls from hq to br2 thru gatekeeper (Roger K?llberg) 2. Troubleshooting Book (Ken Kov) 3. Re: Troubleshooting Book (Tanner Ezell) 4. SIP phone registered with CUCME (iy...@nationwide.com) -- Message: 1 Date: Thu, 4 Mar 2010 15:45:59 +0100 From: Roger K?llberg roger.kallb...@cygate.se Subject: Re: [OSL | CCIE_Voice] Calls from hq to br2 thru gatekeeper To: Omotayo adefilabi...@gmail.com, OSL Group ccie_voice@onlinestudylist.com Message-ID: 79fa99add19eda4c9880d26d736e50ef2d2cbbb...@ex2-sth.domain.root Content-Type: text/plain; charset=windows-1252 When does the call fail? If it fails just after you pick up then please try to add a voice-class codec list to your voip dial-peers that holds both g711 and g729. Roger K?llberg Fr?n: Omotayo [adefilabi...@gmail.com] Skickat: den 4 mars 2010 02:46 Till: OSL Group ?mne: Re: [OSL | CCIE_Voice] Calls from hq to br2 thru gatekeeper Hello, Now the call is getting to the cme but still says call can not be completed as dialled dial-peer voice 200 voip translation-profile incoming GK session target ras incoming called-number . dtmf-relay h245-alphanumeric no vad ! dial-peer voice 300 voip destination-pattern [15]... session target ras tech-prefix 1# dtmf-relay h245-alphanumeric no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper shutdown --More-- Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username=hq phone 2 - ccCallInfo IE subfields - cisco-ani=2123945002 cisco-anitype=0 cisco-aniplan=0 cisco-anipi=0 cisco-anisi=1 dest=1#3002 cisco-desttype=0 cisco-destplan=0 cisco-rdie= cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/cc_api_call_setup_ind_common: Interface=0x4863B3F8, Call Info( Calling Number=2123945002,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown), Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Incoming Dial-peer=200, Progress Indication=NULL(0), Calling IE --More-- Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=188 Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: In: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1/004C02671500/CCAPI/ccCheckClipClir: Out: Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed) Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :cc_get_feature_vsa malloc success Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: cc_get_feature_vsa count is 1 Mar 4 01:44:01.035: //-1//CCAPI/cc_get_feature_vsa: Mar 4 01:44:01.035: :FEATURE_VSA attributes are: feature_name:0,feature_time:1248553792,feature_id:36 Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=2123945002(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed), Called Number=1#3002(TON=Unknown, NPI=Unknown)) Mar 4 01:44:01.039: //188/004C02671500/CCAPI/cc_process_call_setup_ind: Event=0x499F5288 Mar 4 01:44:01.039: //-1//CCAPI/cc_setupind_match_search: Try with the demoted called number 1#3002 Mar 4 01:44:01.039
