Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
In the end, my customer finally realized the problema was on the PBX side. The unupgraded PBX worked fine. Thanks Brian for your help. Regards, Ariel. De: cisco-voip En nombre de ROZA, Ariel Enviado el: martes, 26 de marzo de 2019 14:04 Para: Brian Meade CC: cisco-voip (cisco-voip@puck.nether.net) Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade ´ll check with my customer, and report back. I saw that negative parameter on the o= line, but I wasn´t completely certain how to handle it. Thanks for the help! De: Brian Meade mailto:bmead...@vt.edu>> Enviado el: martes, 26 de marzo de 2019 13:56 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade Actually meant o= line is the origin line. On Tue, Mar 26, 2019 at 12:39 PM Brian Meade mailto:bmead...@vt.edu>> wrote: It's definitely failing at parsing the SDP on that invite and finding an invalid parameter: 07517620.001 |16:00:23.657 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 932 from 172.27.0.15:[5060]: [1031135,NET] INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>> Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>> P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: gquerol P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 07517621.007 |16:00:23.657 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - sdp_ret=SDP_INVALID_PARAMETER You may need to use a SIP Normalization script to clean up what they are sending. I think it's the o= line (organization line). That's 2nd value (-835641967) should be a positive number I believe. That session-id parameter is supposed to match NTP format- https://tools.ietf.org/html/rfc4566#section-5.2<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Frfc4566%23section-5.2=02%7C01%7Cariel.roza%40la.logicalis.com%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878451827=YILRc5UxLiA30sXHmbaGaM9AAyJcHE7P4mc2VEjM8To%3D=0> Maybe just check their server has NTP synced okay to start? Thanks, Brian Meade On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: Here´s the trace file with the bad call De: Brian Meade mailto:bmead...@vt.edu>> Enviado el: lunes, 25 de marzo de 2019 23:39 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: Jonatan Quezada mailto:jonatan.quez...@chemeketa.edu>>; cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade Can you send the trace file you pulled the bad call from? Is MTP Required set on the SIP Trunk? On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one that was updated (a local in-house development, called Mitrol). The system worked fine before the upgrade, and after that it went bonkers. De: Jonatan Quezada mailto:jonatan.quez...@chemeketa.edu>> Enviado el: lunes, 25 de marzo de 2019 19:24 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade we are seeing a similar issues to one of our nodes. we did our during production, Brave but totally doable. After figuring out that we needed to point the EM profiles to the node we were keeping up for the upgrade, we took down the other ucs down, all went well for upgrade. All VM on my ucs are all done now, but there is this huge jitter issues that has risen from the ashes of the upgrade. Its as if my media RTP streams are being forked and the forking is causing the jitter and delay? I have calls where I lose second of audio but signaling seems fine, Im ju
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
´ll check with my customer, and report back. I saw that negative parameter on the o= line, but I wasn´t completely certain how to handle it. Thanks for the help! De: Brian Meade Enviado el: martes, 26 de marzo de 2019 13:56 Para: ROZA, Ariel CC: cisco-voip (cisco-voip@puck.nether.net) Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade Actually meant o= line is the origin line. On Tue, Mar 26, 2019 at 12:39 PM Brian Meade mailto:bmead...@vt.edu>> wrote: It's definitely failing at parsing the SDP on that invite and finding an invalid parameter: 07517620.001 |16:00:23.657 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 932 from 172.27.0.15:[5060]: [1031135,NET] INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>> Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>> P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: gquerol P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 07517621.007 |16:00:23.657 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - sdp_ret=SDP_INVALID_PARAMETER You may need to use a SIP Normalization script to clean up what they are sending. I think it's the o= line (organization line). That's 2nd value (-835641967) should be a positive number I believe. That session-id parameter is supposed to match NTP format- https://tools.ietf.org/html/rfc4566#section-5.2<https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Ftools.ietf.org%2Fhtml%2Frfc4566%23section-5.2=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C3ba4f63173ea43f148f608d6b20bf629%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892161791009351=b1UUK085P%2BCGGFW9RRZcg1MFIXkSIE8FU%2BVUtV1ntUQ%3D=0> Maybe just check their server has NTP synced okay to start? Thanks, Brian Meade On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: Here´s the trace file with the bad call De: Brian Meade mailto:bmead...@vt.edu>> Enviado el: lunes, 25 de marzo de 2019 23:39 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: Jonatan Quezada mailto:jonatan.quez...@chemeketa.edu>>; cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade Can you send the trace file you pulled the bad call from? Is MTP Required set on the SIP Trunk? On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one that was updated (a local in-house development, called Mitrol). The system worked fine before the upgrade, and after that it went bonkers. De: Jonatan Quezada mailto:jonatan.quez...@chemeketa.edu>> Enviado el: lunes, 25 de marzo de 2019 19:24 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade we are seeing a similar issues to one of our nodes. we did our during production, Brave but totally doable. After figuring out that we needed to point the EM profiles to the node we were keeping up for the upgrade, we took down the other ucs down, all went well for upgrade. All VM on my ucs are all done now, but there is this huge jitter issues that has risen from the ashes of the upgrade. Its as if my media RTP streams are being forked and the forking is causing the jitter and delay? I have calls where I lose second of audio but signaling seems fine, Im just losing a ton of packets between the nodes now that they(the pub and sub) are load balancing the media resources, or rather seeming to load ballance. After some dial peer and server group re pointing, all devices finally were on the one node and we were able to upgrade the UCS, but the other is left to do. all of my CUCM On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: Hi, guys and gals. I have a customer with a CUCM 9.0(2) cluster. It is connected to a SIP
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
Actually meant o= line is the origin line. On Tue, Mar 26, 2019 at 12:39 PM Brian Meade wrote: > It's definitely failing at parsing the SDP on that invite and finding an > invalid parameter: > 07517620.001 |16:00:23.657 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: > Incoming SIP UDP message size 932 from 172.27.0.15:[5060]: > [1031135,NET] > INVITE sip:3366@10.4.128.27 SIP/2.0 > Via: SIP/2.0/UDP 172.27.0.15:11347 > ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > From: "Gabriel Querol" ;tag=2792862 > To: > Call-ID: 501227892-15@172.27.0.15 > CSeq: 1 INVITE > Contact: > Max-Forwards: 70 > User-Agent: MitE1x v4.4.5.1062 > Expires: 300 > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO > P-Early-Media: Supported > P-Asserted-Identity: "Gabriel Querol" > P-Mitrol-idLlamada: 190322160050689_MIT_07437 > P-Mitrol-LoginID: gquerol > P-Mitrol-PerfilRuteo: 100 > Content-Length: 233 > Content-Type: application/sdp > > v=0 > o=86329 -835641967 1 IN IP4 172.27.0.15 > s=MitE1x Call > c=IN IP4 172.27.0.15 > t=0 0 > m=audio 36112 RTP/AVP 0 8 101 > a=sendrecv > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > 07517621.007 |16:00:23.657 |AppInfo > |//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - > sdp_ret=SDP_INVALID_PARAMETER > > You may need to use a SIP Normalization script to clean up what they are > sending. > > I think it's the o= line (organization line). That's 2nd value > (-835641967) should be a positive number I believe. That session-id > parameter is supposed to match NTP format- > https://tools.ietf.org/html/rfc4566#section-5.2 > > Maybe just check their server has NTP synced okay to start? > > Thanks, > Brian Meade > > > > On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel > wrote: > >> Here´s the trace file with the bad call >> >> >> >> >> >> >> >> *De:* Brian Meade >> *Enviado el:* lunes, 25 de marzo de 2019 23:39 >> *Para:* ROZA, Ariel >> *CC:* Jonatan Quezada ; cisco-voip ( >> cisco-voip@puck.nether.net) >> *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade >> >> >> >> Can you send the trace file you pulled the bad call from? >> >> >> >> Is MTP Required set on the SIP Trunk? >> >> >> >> On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel >> wrote: >> >> My issue is not a CUCM upgrade. The other side from the SIP Trunk was the >> one that was updated (a local in-house development, called Mitrol). The >> system worked fine before the upgrade, and after that it went bonkers. >> >> >> >> *De:* Jonatan Quezada >> *Enviado el:* lunes, 25 de marzo de 2019 19:24 >> *Para:* ROZA, Ariel >> *CC:* cisco-voip (cisco-voip@puck.nether.net) > > >> *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade >> >> >> >> we are seeing a similar issues to one of our nodes. we did our during >> production, Brave but totally doable. After figuring out that we needed to >> point the EM profiles to the node we were keeping up for the upgrade, we >> took down the other ucs down, all went well for upgrade. All VM on my ucs >> are all done now, but there is this huge jitter issues that has risen from >> the ashes of the upgrade. Its as if my media RTP streams are being forked >> and the forking is causing the jitter and delay? >> >> >> >> I have calls where I lose second of audio but signaling seems fine, Im >> just losing a ton of packets between the nodes now that they(the pub and >> sub) are load balancing the media resources, or rather seeming to load >> ballance. >> >> >> >> After some dial peer and server group re pointing, all devices finally >> were on the one node and we were able to upgrade the UCS, but the other is >> left to do. all of my CUCM >> >> >> >> On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel >> wrote: >> >> Hi, guys and gals. >> >> >> >> I have a customer with a CUCM 9.0(2) cluster. >> >> It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or >> otherwise). The PBX has four different nodes, all configured in the SIP >> TRUNK >> >> >> >> They claim it was working fine until last Thursday, where they did an >> upgrade to one of the nodes of the PBX. After that, calls going from PBX to >> CUCM fail with a 488 Media Not Acceptable error. >> >> They also ha
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
It's definitely failing at parsing the SDP on that invite and finding an invalid parameter: 07517620.001 |16:00:23.657 |AppInfo |//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 932 from 172.27.0.15:[5060]: [1031135,NET] INVITE sip:3366@10.4.128.27 SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347 ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" ;tag=2792862 To: Call-ID: 501227892-15@172.27.0.15 CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: "Gabriel Querol" P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: gquerol P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 07517621.007 |16:00:23.657 |AppInfo |//SIP/SIPHandler/ccbId=0/scbId=0/extract_sdp: sdp_parse failed - sdp_ret=SDP_INVALID_PARAMETER You may need to use a SIP Normalization script to clean up what they are sending. I think it's the o= line (organization line). That's 2nd value (-835641967) should be a positive number I believe. That session-id parameter is supposed to match NTP format- https://tools.ietf.org/html/rfc4566#section-5.2 Maybe just check their server has NTP synced okay to start? Thanks, Brian Meade On Tue, Mar 26, 2019 at 10:33 AM ROZA, Ariel wrote: > Here´s the trace file with the bad call > > > > > > > > *De:* Brian Meade > *Enviado el:* lunes, 25 de marzo de 2019 23:39 > *Para:* ROZA, Ariel > *CC:* Jonatan Quezada ; cisco-voip ( > cisco-voip@puck.nether.net) > *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade > > > > Can you send the trace file you pulled the bad call from? > > > > Is MTP Required set on the SIP Trunk? > > > > On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel > wrote: > > My issue is not a CUCM upgrade. The other side from the SIP Trunk was the > one that was updated (a local in-house development, called Mitrol). The > system worked fine before the upgrade, and after that it went bonkers. > > > > *De:* Jonatan Quezada > *Enviado el:* lunes, 25 de marzo de 2019 19:24 > *Para:* ROZA, Ariel > *CC:* cisco-voip (cisco-voip@puck.nether.net) > *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade > > > > we are seeing a similar issues to one of our nodes. we did our during > production, Brave but totally doable. After figuring out that we needed to > point the EM profiles to the node we were keeping up for the upgrade, we > took down the other ucs down, all went well for upgrade. All VM on my ucs > are all done now, but there is this huge jitter issues that has risen from > the ashes of the upgrade. Its as if my media RTP streams are being forked > and the forking is causing the jitter and delay? > > > > I have calls where I lose second of audio but signaling seems fine, Im > just losing a ton of packets between the nodes now that they(the pub and > sub) are load balancing the media resources, or rather seeming to load > ballance. > > > > After some dial peer and server group re pointing, all devices finally > were on the one node and we were able to upgrade the UCS, but the other is > left to do. all of my CUCM > > > > On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel > wrote: > > Hi, guys and gals. > > > > I have a customer with a CUCM 9.0(2) cluster. > > It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or > otherwise). The PBX has four different nodes, all configured in the SIP > TRUNK > > > > They claim it was working fine until last Thursday, where they did an > upgrade to one of the nodes of the PBX. After that, calls going from PBX to > CUCM fail with a 488 Media Not Acceptable error. > > They also have tried making calls from one of the not upgraded nodes, with > the same error. > > I have been looking into the SIP traces, and I see nothing really telling > of a problem there. > > > > We reseted the SIP trunk with no success. > > I have looked at the región configuration, and all regions are set to the > System Default (G722, G711) > > I also tried changing the preferred codec in the SIP trunk, with no > success. > > > > Following this, I am pasting the SIP messages of a failed call from PBX -> > CUCM and a successfull call in the reverse, from CUCM -> PBX. > > > > Can you see if anything is wrong or odd? > > > > Regards, > > > > Ariel. >
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
I did not check all those options because the trunk was working before my customer upgraded the PBX I have the logs, thats where I got the SIP messages from. I´ll try to upload them to the list De: UC Penguin Enviado el: lunes, 25 de marzo de 2019 21:40 Para: ROZA, Ariel CC: Jonatan Quezada ; cisco-voip (cisco-voip@puck.nether.net) Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade Is there a SIP normalization profile attached to the SIP trunk used for “Failed Call from PBX”? Are changes required to that profile after the remote PBX was modified? For the “Failed Call from PBX”: This is a SIP early offer invite. Does the CUCM trunk support early offer? This invite has advertises it supports early media. Does the CUCM SIP trunk support early media? There is no ptime listed in the SIP invite. How does CUCM know what ptime to use? Are MTP resources available for this trunk? Have you pulled CallManager SDL Logs? On Mar 25, 2019, at 18:13, ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one that was updated (a local in-house development, called Mitrol). The system worked fine before the upgrade, and after that it went bonkers. De: Jonatan Quezada mailto:jonatan.quez...@chemeketa.edu>> Enviado el: lunes, 25 de marzo de 2019 19:24 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) mailto:cisco-voip@puck.nether.net>> Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade we are seeing a similar issues to one of our nodes. we did our during production, Brave but totally doable. After figuring out that we needed to point the EM profiles to the node we were keeping up for the upgrade, we took down the other ucs down, all went well for upgrade. All VM on my ucs are all done now, but there is this huge jitter issues that has risen from the ashes of the upgrade. Its as if my media RTP streams are being forked and the forking is causing the jitter and delay? I have calls where I lose second of audio but signaling seems fine, Im just losing a ton of packets between the nodes now that they(the pub and sub) are load balancing the media resources, or rather seeming to load ballance. After some dial peer and server group re pointing, all devices finally were on the one node and we were able to upgrade the UCS, but the other is left to do. all of my CUCM On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: Hi, guys and gals. I have a customer with a CUCM 9.0(2) cluster. It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or otherwise). The PBX has four different nodes, all configured in the SIP TRUNK They claim it was working fine until last Thursday, where they did an upgrade to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail with a 488 Media Not Acceptable error. They also have tried making calls from one of the not upgraded nodes, with the same error. I have been looking into the SIP traces, and I see nothing really telling of a problem there. We reseted the SIP trunk with no success. I have looked at the región configuration, and all regions are set to the System Default (G722, G711) I also tried changing the preferred codec in the SIP trunk, with no success. Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM and a successfull call in the reverse, from CUCM -> PBX. Can you see if anything is wrong or odd? Regards, Ariel. Failed Call from PBX INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: " " mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>> Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: " " mailto:sip%3A86329@172.27.0.15>> P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Reply from CUCM --- SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>>;tag=573234994 Date: Fri
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
Can you send the trace file you pulled the bad call from? Is MTP Required set on the SIP Trunk? On Mon, Mar 25, 2019 at 7:14 PM ROZA, Ariel wrote: > My issue is not a CUCM upgrade. The other side from the SIP Trunk was the > one that was updated (a local in-house development, called Mitrol). The > system worked fine before the upgrade, and after that it went bonkers. > > > > *De:* Jonatan Quezada > *Enviado el:* lunes, 25 de marzo de 2019 19:24 > *Para:* ROZA, Ariel > *CC:* cisco-voip (cisco-voip@puck.nether.net) > *Asunto:* Re: [cisco-voip] SIp Trunk call failing after PBX upgrade > > > > we are seeing a similar issues to one of our nodes. we did our during > production, Brave but totally doable. After figuring out that we needed to > point the EM profiles to the node we were keeping up for the upgrade, we > took down the other ucs down, all went well for upgrade. All VM on my ucs > are all done now, but there is this huge jitter issues that has risen from > the ashes of the upgrade. Its as if my media RTP streams are being forked > and the forking is causing the jitter and delay? > > > > I have calls where I lose second of audio but signaling seems fine, Im > just losing a ton of packets between the nodes now that they(the pub and > sub) are load balancing the media resources, or rather seeming to load > ballance. > > > > After some dial peer and server group re pointing, all devices finally > were on the one node and we were able to upgrade the UCS, but the other is > left to do. all of my CUCM > > > > On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel > wrote: > > Hi, guys and gals. > > > > I have a customer with a CUCM 9.0(2) cluster. > > It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or > otherwise). The PBX has four different nodes, all configured in the SIP > TRUNK > > > > They claim it was working fine until last Thursday, where they did an > upgrade to one of the nodes of the PBX. After that, calls going from PBX to > CUCM fail with a 488 Media Not Acceptable error. > > They also have tried making calls from one of the not upgraded nodes, with > the same error. > > I have been looking into the SIP traces, and I see nothing really telling > of a problem there. > > > > We reseted the SIP trunk with no success. > > I have looked at the región configuration, and all regions are set to the > System Default (G722, G711) > > I also tried changing the preferred codec in the SIP trunk, with no > success. > > > > Following this, I am pasting the SIP messages of a failed call from PBX -> > CUCM and a successfull call in the reverse, from CUCM -> PBX. > > > > Can you see if anything is wrong or odd? > > > > Regards, > > > > Ariel. > > > > Failed Call from PBX > > > > > > INVITE sip:3366@10.4.128.27 SIP/2.0 > > Via: SIP/2.0/UDP 172.27.0.15:11347 > ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > > From: " " ;tag=2792862 > > To: > > Call-ID: 501227892-15@172.27.0.15 > > CSeq: 1 INVITE > > Contact: > > Max-Forwards: 70 > > User-Agent: MitE1x v4.4.5.1062 > > Expires: 300 > > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO > > P-Early-Media: Supported > > P-Asserted-Identity: " " > > P-Mitrol-idLlamada: 190322160050689_MIT_07437 > > P-Mitrol-LoginID: > > P-Mitrol-PerfilRuteo: 100 > > Content-Length: 233 > > Content-Type: application/sdp > > v=0 > > o=86329 -835641967 1 IN IP4 172.27.0.15 > > s=MitE1x Call > > c=IN IP4 172.27.0.15 > > t=0 0 > > m=audio 36112 RTP/AVP 0 8 101 > > a=sendrecv > > a=rtpmap:0 PCMU/8000/1 > > a=rtpmap:8 PCMA/8000/1 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > > > Reply from CUCM > > --- > > > > SIP/2.0 488 Not Acceptable Media > > Via: SIP/2.0/UDP 172.27.0.15:11347 > ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > > From: "Gabriel Querol" ;tag=2792862 > > To: ;tag=573234994 > > Date: Fri, 22 Mar 2019 19:00:23 GMT > > Call-ID: 501227892-15@172.27.0.15 > > CSeq: 1 INVITE > > Allow-Events: presence > > Warning: 304 10.4.128.27 "Media Type(s) Unavailable" > > Reason: Q.850;cause=65 > > Content-Length: 0 > > > > > > > > > > SUCESSFULL CALL FROM CUCM > > - > > INVITE sip:*86329@172.27.0.12:5060 > <https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2F86329%40172.27.0.12%3A5060=02%7
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
Is there a SIP normalization profile attached to the SIP trunk used for “Failed Call from PBX”? Are changes required to that profile after the remote PBX was modified? For the “Failed Call from PBX”: This is a SIP early offer invite. Does the CUCM trunk support early offer? This invite has advertises it supports early media. Does the CUCM SIP trunk support early media? There is no ptime listed in the SIP invite. How does CUCM know what ptime to use? Are MTP resources available for this trunk? Have you pulled CallManager SDL Logs? > On Mar 25, 2019, at 18:13, ROZA, Ariel wrote: > > My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one > that was updated (a local in-house development, called Mitrol). The system > worked fine before the upgrade, and after that it went bonkers. > > De: Jonatan Quezada > Enviado el: lunes, 25 de marzo de 2019 19:24 > Para: ROZA, Ariel > CC: cisco-voip (cisco-voip@puck.nether.net) > Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade > > we are seeing a similar issues to one of our nodes. we did our during > production, Brave but totally doable. After figuring out that we needed to > point the EM profiles to the node we were keeping up for the upgrade, we took > down the other ucs down, all went well for upgrade. All VM on my ucs are all > done now, but there is this huge jitter issues that has risen from the ashes > of the upgrade. Its as if my media RTP streams are being forked and the > forking is causing the jitter and delay? > > I have calls where I lose second of audio but signaling seems fine, Im just > losing a ton of packets between the nodes now that they(the pub and sub) are > load balancing the media resources, or rather seeming to load ballance. > > After some dial peer and server group re pointing, all devices finally were > on the one node and we were able to upgrade the UCS, but the other is left to > do. all of my CUCM > > On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel > wrote: > Hi, guys and gals. > > I have a customer with a CUCM 9.0(2) cluster. > It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or > otherwise). The PBX has four different nodes, all configured in the SIP TRUNK > > They claim it was working fine until last Thursday, where they did an upgrade > to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail > with a 488 Media Not Acceptable error. > They also have tried making calls from one of the not upgraded nodes, with > the same error. > I have been looking into the SIP traces, and I see nothing really telling of > a problem there. > > We reseted the SIP trunk with no success. > I have looked at the región configuration, and all regions are set to the > System Default (G722, G711) > I also tried changing the preferred codec in the SIP trunk, with no success. > > Following this, I am pasting the SIP messages of a failed call from PBX -> > CUCM and a successfull call in the reverse, from CUCM -> PBX. > > Can you see if anything is wrong or odd? > > Regards, > > Ariel. > > Failed Call from PBX > > > INVITE sip:3366@10.4.128.27 SIP/2.0 > Via: SIP/2.0/UDP > 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > From: " " ;tag=2792862 > To: > Call-ID: 501227892-15@172.27.0.15 > CSeq: 1 INVITE > Contact: > Max-Forwards: 70 > User-Agent: MitE1x v4.4.5.1062 > Expires: 300 > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO > P-Early-Media: Supported > P-Asserted-Identity: " " > P-Mitrol-idLlamada: 190322160050689_MIT_07437 > P-Mitrol-LoginID: > P-Mitrol-PerfilRuteo: 100 > Content-Length: 233 > Content-Type: application/sdp > v=0 > o=86329 -835641967 1 IN IP4 172.27.0.15 > s=MitE1x Call > c=IN IP4 172.27.0.15 > t=0 0 > m=audio 36112 RTP/AVP 0 8 101 > a=sendrecv > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > > > Reply from CUCM > --- > > SIP/2.0 488 Not Acceptable Media > Via: SIP/2.0/UDP > 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > From: "Gabriel Querol" ;tag=2792862 > To: ;tag=573234994 > Date: Fri, 22 Mar 2019 19:00:23 GMT > Call-ID: 501227892-15@172.27.0.15 > CSeq: 1 INVITE > Allow-Events: presence > Warning: 304 10.4.128.27 "Media Type(s) Unavailable" > Reason: Q.850;cause=65 > Content-Length: 0 > > > > > SUCESSFULL CALL FROM CUCM > - > INVITE sip:*86329@172.27.0.12:5060 SIP/2.0 > Via: SIP/2.0/UD
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
My issue is not a CUCM upgrade. The other side from the SIP Trunk was the one that was updated (a local in-house development, called Mitrol). The system worked fine before the upgrade, and after that it went bonkers. De: Jonatan Quezada Enviado el: lunes, 25 de marzo de 2019 19:24 Para: ROZA, Ariel CC: cisco-voip (cisco-voip@puck.nether.net) Asunto: Re: [cisco-voip] SIp Trunk call failing after PBX upgrade we are seeing a similar issues to one of our nodes. we did our during production, Brave but totally doable. After figuring out that we needed to point the EM profiles to the node we were keeping up for the upgrade, we took down the other ucs down, all went well for upgrade. All VM on my ucs are all done now, but there is this huge jitter issues that has risen from the ashes of the upgrade. Its as if my media RTP streams are being forked and the forking is causing the jitter and delay? I have calls where I lose second of audio but signaling seems fine, Im just losing a ton of packets between the nodes now that they(the pub and sub) are load balancing the media resources, or rather seeming to load ballance. After some dial peer and server group re pointing, all devices finally were on the one node and we were able to upgrade the UCS, but the other is left to do. all of my CUCM On Mon, Mar 25, 2019 at 10:17 AM ROZA, Ariel mailto:ariel.r...@la.logicalis.com>> wrote: Hi, guys and gals. I have a customer with a CUCM 9.0(2) cluster. It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or otherwise). The PBX has four different nodes, all configured in the SIP TRUNK They claim it was working fine until last Thursday, where they did an upgrade to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail with a 488 Media Not Acceptable error. They also have tried making calls from one of the not upgraded nodes, with the same error. I have been looking into the SIP traces, and I see nothing really telling of a problem there. We reseted the SIP trunk with no success. I have looked at the región configuration, and all regions are set to the System Default (G722, G711) I also tried changing the preferred codec in the SIP trunk, with no success. Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM and a successfull call in the reverse, from CUCM -> PBX. Can you see if anything is wrong or odd? Regards, Ariel. Failed Call from PBX INVITE sip:3366@10.4.128.27<mailto:sip%3A3366@10.4.128.27> SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: " " mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>> Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: " " mailto:sip%3A86329@172.27.0.15>> P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Reply from CUCM --- SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: mailto:sip%3A3366@10.4.128.27>>;tag=573234994 Date: Fri, 22 Mar 2019 19:00:23 GMT Call-ID: 501227892-15@172.27.0.15<mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Allow-Events: presence Warning: 304 10.4.128.27 "Media Type(s) Unavailable" Reason: Q.850;cause=65 Content-Length: 0 SUCESSFULL CALL FROM CUCM - INVITE sip:*86329@172.27.0.12:5060<https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2F86329%40172.27.0.12%3A5060=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C70e1772c8c6d42083a1308d6b1709fcd%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891494629446120=AO4MLkjYlNIvMkLH5FTGzhrftQtRKkh4XhPrzaJRoCw%3D=0> SIP/2.0 Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8 From: " (3307)" mailto:sip%3A3307@10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 To: mailto:86329@172.27.0.12>> Date: Mon, 25 Mar 2019 10:40:36 GMT Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27<mailto:6b366f80-c981b024-4f13-1b80040a@10.4.128.27> Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM9.1 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-E
Re: [cisco-voip] SIp Trunk call failing after PBX upgrade
Cause No. 65 - bearer capability not implemented. This cause indicates that the equipment sending this cause does not support the bearer capability requested. What it means: 1. In most cases, the number being called is not an ISDN number but an analog destination. 2. The equipment is dialing at a faster rate than the circuitry allows, for example, dialing at 64K when only 56K is supported. Where is the call going, out a gateway or just a Cisco phone? From: ROZA, Ariel Sent: Monday, March 25, 2019 2:03 PM To: NateCCIE ; 'cisco-voip' Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade That was the original setting, and the results is what I included in the mail De: NateCCIE mailto:natec...@gmail.com> > Enviado el: lunes, 25 de marzo de 2019 17:01 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip' mailto:cisco-voip@puck.nether.net> > Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade I would change preferred codec to 711a and see what happens. From: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> > Sent: Monday, March 25, 2019 1:37 PM To: NateCCIE mailto:natec...@gmail.com> >; 'cisco-voip' mailto:cisco-voip@puck.nether.net> > Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade Yes I already looked at that /1. According to the RFC, the /1 denotes the quantity of channels and it is optional when the codec uses only one channel. I looked up posible bugs related to that in the Bug Search Tool and did not find anything suitable. Already tried changing the Preferred codec to G711U and got the same results, except the output now shows PCMU/8000 from CUCM side, as expected. Thanks, Nate. De: NateCCIE mailto:natec...@gmail.com> > Enviado el: lunes, 25 de marzo de 2019 14:33 Para: ROZA, Ariel mailto:ariel.r...@la.logicalis.com> >; 'cisco-voip' mailto:cisco-voip@puck.nether.net> > Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade Non working call shows G711u and a, working call shows only a. there is also a difference of the /1 at the end not sure what that indicates. a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000 From: cisco-voip mailto:cisco-voip-boun...@puck.nether.net> > On Behalf Of ROZA, Ariel Sent: Monday, March 25, 2019 11:17 AM To: cisco-voip (cisco-voip@puck.nether.net <mailto:cisco-voip@puck.nether.net> ) mailto:cisco-voip@puck.nether.net> > Subject: [cisco-voip] SIp Trunk call failing after PBX upgrade Hi, guys and gals. I have a customer with a CUCM 9.0(2) cluster. It is connected to a SIP PBX via a direct SIP TRUNK (No SBC, CUBE or otherwise). The PBX has four different nodes, all configured in the SIP TRUNK They claim it was working fine until last Thursday, where they did an upgrade to one of the nodes of the PBX. After that, calls going from PBX to CUCM fail with a 488 Media Not Acceptable error. They also have tried making calls from one of the not upgraded nodes, with the same error. I have been looking into the SIP traces, and I see nothing really telling of a problem there. We reseted the SIP trunk with no success. I have looked at the región configuration, and all regions are set to the System Default (G722, G711) I also tried changing the preferred codec in the SIP trunk, with no success. Following this, I am pasting the SIP messages of a failed call from PBX -> CUCM and a successfull call in the reverse, from CUCM -> PBX. Can you see if anything is wrong or odd? Regards, Ariel. Failed Call from PBX INVITE sip:3366@10.4.128.27 SIP/2.0 Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: " " ;tag=2792862 To: Call-ID: 501227892-15@172.27.0.15 <mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Contact: Max-Forwards: 70 User-Agent: MitE1x v4.4.5.1062 Expires: 300 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO P-Early-Media: Supported P-Asserted-Identity: " " P-Mitrol-idLlamada: 190322160050689_MIT_07437 P-Mitrol-LoginID: P-Mitrol-PerfilRuteo: 100 Content-Length: 233 Content-Type: application/sdp v=0 o=86329 -835641967 1 IN IP4 172.27.0.15 s=MitE1x Call c=IN IP4 172.27.0.15 t=0 0 m=audio 36112 RTP/AVP 0 8 101 a=sendrecv a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Reply from CUCM --- SIP/2.0 488 Not Acceptable Media Via: SIP/2.0/UDP 172.27.0.15:11347;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm From: "Gabriel Querol" ;tag=2792862 To: ;tag=573234994 Date: Fri, 22 Mar 2019 19:00:23 GMT Call-ID: 501227892-15@172.27.0.15 <mailto:501227892-15@172.27.0.15> CSeq: 1 INVITE Allow-Events: presence Warning: 304 10.4.128.27 "Media Type(s) Unavailable" Reason:
