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 <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 <jonatan.quez...@chemeketa.edu> > *Enviado el:* lunes, 25 de marzo de 2019 19:24 > *Para:* ROZA, Ariel <ariel.r...@la.logicalis.com> > *CC:* cisco-voip (cisco-voip@puck.nether.net) <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 <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 regiĆ³n 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: "XXXX XXXX" <sip:86329@172.27.0.15>;tag=2792862 > > To: <sip:3366@10.4.128.27> > > Call-ID: 501227892-15@172.27.0.15 > > CSeq: 1 INVITE > > Contact: <sip:86329@172.27.0.15:11347;transport=udp> > > 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: "XXXX XXXX" <sip:86329@172.27.0.15> > > P-Mitrol-idLlamada: 190322160050689_MIT_07437 > > P-Mitrol-LoginID: XXXX > > 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" <sip:86329@172.27.0.15>;tag=2792862 > > To: <sip:3366@10.4.128.27>;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&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C70e1772c8c6d42083a1308d6b1709fcd%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891494629446120&sdata=AO4MLkjYlNIvMkLH5FTGzhrftQtRKkh4XhPrzaJRoCw%3D&reserved=0> > SIP/2.0 > > Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8 > > From: "XXXX XXXX (3307)" <sip:3307@10.4.128.27 > >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 > > To: <sip:*86329@172.27.0.12> > > Date: Mon, 25 Mar 2019 10:40:36 GMT > > Call-ID: 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-Events: presence, kpml > > Supported: X-cisco-srtp-fallback,X-cisco-original-called > > Cisco-Guid: 1798729600-0000065536-0000010811-0461374474 > > Session-Expires: 1800 > > P-Asserted-Identity: "XXXX XXXX (3307)" <sip:3307@10.4.128.27> > > Remote-Party-ID: "XXXX XXXX (3307)" <sip:3307@10.4.128.27 > >;party=calling;screen=yes;privacy=off > > Contact: <sip:3307@10.4.128.27:5060 > <https://nam01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fsip%3A3307%4010.4.128.27%3A5060&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7C70e1772c8c6d42083a1308d6b1709fcd%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891494629456129&sdata=tZVxfGiCVMA6uxvQR3KN%2FLNVL7AVW9O6bozXRy4j7i8%3D&reserved=0> > > > > Max-Forwards: 69 > > Content-Type: application/sdp > > Content-Length: 212 > > v=0 > > o=CiscoSystemsCCM-SIP 429005 1 IN IP4 10.4.128.27 > > s=SIP Call > > c=IN IP4 10.4.128.12 > > t=0 0 > > m=audio 30530 RTP/AVP 8 101 > > a=rtpmap:8 PCMA/8000 > > a=ptime:20 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > > > Answer from the PBX > > ---------------------- > > > > SIP/2.0 183 Session Progress > > Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8 > > From: "Gabriel Querol (3307)" <sip:3307@10.4.128.27 > >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 > > To: <sip:*86329@172.27.0.12>;tag=43743456 > > Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27 > > CSeq: 101 INVITE > > Server: MitE1x v4.4.5.1062 > > Allow: INVITE,ACK,CANCEL,BYE,REGISTER,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO > > P-Mitrol-idLlamada: 190325074112281_MIT_02447 > > Content-Length: 217 > > Content-Type: application/sdp > > v=0 > > o=CiscoSystemsCCM-SIP 429005 1 IN IP4 172.27.0.12 > > s=MitE1x Call > > c=IN IP4 172.27.0.12 > > t=0 0 > > m=audio 36508 RTP/AVP 8 101 > > a=sendrecv > > a=rtpmap:8 PCMA/8000/1 > > a=rtpmap:101 telephone-event/8000 > > a=fmtp:101 0-15 > > > > _______________________________________________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip > 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