Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.comwrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ 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] DTMF Issue with SIP TRUNK (CUCM and CUE)
Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ 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 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no supp services” would have an impact on his DTMF issue. I’m trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn’t supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote: Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ 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 ___ 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] DTMF Issue with SIP TRUNK (CUCM and CUE)
no supplementary service affect only call forwarding and call transfer , i do not know how it solve DTMF Regards, Moataz Tolba On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com wrote: I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no supp services” would have an impact on his DTMF issue. I’m trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn’t supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote: Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ 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 ___ 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___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
Hello All, I have attached the debug ccsip messages output before and after using the command. I do not have the answer why it resolved the dtmf-issue. If you guys find something, please share it. Thanks, Viki On Thu, Jan 30, 2014 at 4:16 PM, Moataz moataz_m...@yahoo.com wrote: no supplementary service affect only call forwarding and call transfer , i do not know how it solve DTMF Regards, Moataz Tolba On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com wrote: I understand how DTMF works on SIP Trunks, what I'm not clear on is how no supp services would have an impact on his DTMF issue. I'm trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn't supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.comwrote: Something doesn't seem to add up in my head. Supp Services shouldn't effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.comwrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ 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 ___ 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 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc dtmf Description: Binary data dtmf Description: Binary data ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube:
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
In the larger debug attachment the SDP includes a=fmtp:18 in the 200 OK coming from the CME site (IP 3.3.3.3). In the other capture I didn’t see any SDP. If no DTMF offer is present during call setup, this would assume plain old in-band DTMF, which won’t work on a compressed codec like G.729. You press digits and nothing happens. G729 requires RFC 2833, SIP NOTIFY, or KPML to function properly. On Jan 30, 2014, at 1:05 PM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello All, I have attached the debug ccsip messages output before and after using the command. I do not have the answer why it resolved the dtmf-issue. If you guys find something, please share it. Thanks, Viki On Thu, Jan 30, 2014 at 4:16 PM, Moataz moataz_m...@yahoo.com wrote: no supplementary service affect only call forwarding and call transfer , i do not know how it solve DTMF Regards, Moataz Tolba On Thursday, 30 January 2014, 15:17, Mark Holloway m...@markholloway.com wrote: I understand how DTMF works on SIP Trunks, what I’m not clear on is how “no supp services” would have an impact on his DTMF issue. I’m trying to understand the logic of something changing with RFC2833 or SIP NOTIFY to the point where # is now recognized, yet without changing anything related to DTMF. Wouldn’t supp services only impact the signlaing behavior of the SIP 302 message itself? But not DTMF? On Jan 30, 2014, at 8:00 AM, Bill Lake whl...@gmail.com wrote: Inbound SIP trunk from ITSP and CUE http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml He would see the issue in the debugs On Thu, Jan 30, 2014 at 6:51 AM, Mark Holloway m...@markholloway.com wrote: Something doesn’t seem to add up in my head. Supp Services shouldn’t effect DTMF. Did you change anything related to the SIP Trunk on CUCM? Or anything DTMF related on a dial-peer? On Jan 30, 2014, at 6:22 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Hello Somphol/Justin, I have resolved the issue by adding the command no supplementary-service sip moved-temporarily. Thanks a lot Somphol for pointing the document to me. Thank you Justin for providing me the inputs. Regards, Viki On Thu, Jan 30, 2014 at 6:15 AM, Justin Carney justin.s.car...@gmail.com wrote: I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html. Regards, --Somphol. ___ 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 ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube:
[OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
Hello All, I am facing an issue with dtmf-relay. PhoneA registered to CUCM (SiteA) calls PhoneD registered to CUCME (Site C). Between Regions G729 codec is negotiated. PhoneD call-forward no answer to Voicemail. CUE integrated with CME. After leaving the Voicemail from PhoneA to PhoneD, when I press # key to send the Voicemail, it is not recognized. Here is my scenario and the configuration. (PhoneA) -- CUCM SIP TRUNK CUCME (PhoneD) --- CUE. voice service voip ip address trusted list ipv4 0.0.0.0 0.0.0.0 allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none h323 sip bind control source-interface Loopback0 bind media source-interface Loopback0 registrar server expires max 600 min 60 ! voice class codec 1 codec preference 1 g711ulaw codec preference 2 g729r8 ! telephony-service sdspfarm units 3 sdspfarm transcode sessions 4 max-ephones 5 max-dn 10 ip source-address 3.3.3.3 port 2000 load 7945 term45.default.loads time-zone 28 time-format 24 date-format dd-mm-yy voicemail 3600 max-conferences 8 gain -6 call-forward pattern .T moh music-on-hold.au dn-webedit transfer-system full-consult transfer-pattern .T create cnf-files ! sccp local Loopback0 sccp ccm 10.0.10.160 identifier 2 version 7.0 sccp ccm 3.3.3.3 identifier 1 version 7.0 sccp ip precedence 3 sccp sccp ccm group 1 associate ccm 1 priority 1 associate ccm 2 priority 2 associate profile 1 register BR2-IOS-XCODE associate profile 2 register BR2-IOS-CFB keepalive retries 5 switchover method immediate switchback method immediate switchback interval 15 ! dspfarm profile 1 transcode codec g729r8 codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 maximum sessions 4 associate application SCCP ! dspfarm profile 2 conference codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 2 associate application SCCP ! dial-peer voice 1000 voip destination-pattern [15]...$ session protocol sipv2 session target ipv4:10.0.10.160 incoming called-number . voice-class codec 1 dtmf-relay sip-notify no vad ! dial-peer voice 3600 voip destination-pattern 3[16]00$ session protocol sipv2 session target ipv4:10.10.202.100 incoming called-number 399[89] dtmf-relay sip-notify codec g711ulaw no vad ! *On the CUCM, I did the following,*Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw DTMF Signaling MethodRequired Field: No preference Non Secure SIP Trunk Profile: I am using TCP+UDP for INCOMING + Accept Unsolicited Notification (Checked). Please let me know what I am missing. Thanks, Viki ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.comwrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ Free CCIE RS, Collaboration, Data Center, Wireless Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
Re: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE)
I concur with Somphol's suggestion and that mtp shouldn't be required. You stated you can record the voicemail but I don't see the sdspfarm tag 1 BR2-IOS-XCODE command under telephony-service. Is your transcoder showing its registered with show sccp command? I'm guessing that it is registered else you wouldn't be getting to cue using g729 that is coming over the wan (maybe the tag command just got lost on the copy/paste of the config to the email?). (Also for the sccp config you're missing the same tag command for the cfb and the conference hardware command. You have the sccp ccm pointing to the cucm ip after cme, are you trying to register sccp resources to cucm?) You can run debug ccsip messages on cme to ensure you see the dtmf comes across the sip trunk from cucm. Dial peer 3600 for cue lists dtmf-relay sip-notify and just double check this is set the same inside cue. For an alternate test, when you place the same call can you leave a message ( 2 sec) and hang up without pressing pound? Does the mwi come on and can the cme phone retrieve the voicemail after entering the pin? If so use the same debug ccsip messages cmd to see the expected/normal debug output for the dtmf on this working scenario. Hope this helps... -Justin (Sent from my phone, please excuse and/or laugh at any typos.) On Jan 29, 2014 5:40 PM, Somphol Boonjing somp...@gmail.com wrote: On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman sethuvign...@gmail.com wrote: Media Termination Point Required (Checked) MTP Preferred Originating CodecRequired Field: g711ulaw Hi Vignesh, I think if you can set these two to default settings which is MTP Required [uncheck], and MTP Prefered Codec: None, Leave the DTMF Signaling Method to No Preference. Reset the SIP Trunk. You shouldn't need MTP for this operation. Then, if you really want to experiment with MTP insertion, I think you may find this article interesting - http://www.ucguerrilla.com/2012/12/ccie-v-i-shoulda-checked-that-tip-3-mtp.html . Regards, --Somphol. ___ 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] DTMF Issue with SIP TRUNK (CUCM and CUE)
Hello What do you see when you do 'debug ccsip messages' on cucme Sent using BlackBerry® from mobinil -Original Message- From: Vignesh Sethuraman sethuvign...@gmail.com Sender: ccie_voice-boun...@onlinestudylist.com Date: Wed, 29 Jan 2014 22:48:46 To: ccievoiceccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE) ___ 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
