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 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 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 > 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 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 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 >> 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 >>> 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" wrote: >>> >>> On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman >>> 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: >>> >>> iPexpert on YouTube: www.youtube.com/ipexpertinc >>> >>> ___ >>> Free CCIE R&S, 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 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 > 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 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 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 > 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 > 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" 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > > > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > > > > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > > > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > dtmf Description: Binary data dtmf Description: Binary data ___ Free CCIE R&S, Collaboration, Data Center, Wireless & Security Vide
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 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 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 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 >>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 >>>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" wrote: > >On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman > 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: > >iPexpert on YouTube: www.youtube.com/ipexpertinc > >>> ___ >>>Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: >>> >>>iPexpert on YouTube: www.youtube.com/ipexpertinc >> >>___ >>Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: >> >>iPexpert on YouTube: www.youtube.com/ipexpertinc >> > ___ Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc___ Free CCIE R&S, 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 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 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 > 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 >> 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" wrote: >> >> On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman >> 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: >> >> iPexpert on YouTube: www.youtube.com/ipexpertinc >> >> ___ >> Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: >> >> iPexpert on YouTube: www.youtube.com/ipexpertinc > > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > ___ Free CCIE R&S, 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 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 > 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" wrote: > > On Thu, Jan 30, 2014 at 8:48 AM, Vignesh Sethuraman > 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > > ___ > Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE R&S, 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 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 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" 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: >> >> iPexpert on YouTube: www.youtube.com/ipexpertinc >> > ___ Free CCIE R&S, 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 Sender: ccie_voice-boun...@onlinestudylist.com Date: Wed, 29 Jan 2014 22:48:46 To: ccievoice Subject: [OSL | CCIE_Voice] DTMF Issue with SIP TRUNK (CUCM and CUE) ___ Free CCIE R&S, Collaboration, Data Center, Wireless & Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc ___ Free CCIE R&S, 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" 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: > > iPexpert on YouTube: www.youtube.com/ipexpertinc > ___ Free CCIE R&S, 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 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: , 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 R&S, Collaboration, Data Center, Wireless & Security Videos :: iPexpert on YouTube: www.youtube.com/ipexpertinc
