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 <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 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: "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 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> > 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 > > > -- > For immediate assistance please reach out to Chemeketa IT Help Desk at > 5033997899 > -or- > Visit the help center > > https://projects.chemeketa.edu/servicedesk/customer/portals > > Johnny Q > Voice Technology Analyst II > Network, Infrastructure, Routing Devices, and Servers > Chemeketa Community College > johnn...@chemeketa.edu > Building 22 Room 130 > Work 5033995294 > Mobile 9712182110 > SIP 5035406689 > FAX 5033995549 > > _______________________________________________ > cisco-voip mailing list > cisco-voip@puck.nether.net > https://puck.nether.net/mailman/listinfo/cisco-voip
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