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 <cisco-voip-boun...@puck.nether.net> En nombre de ROZA, Ariel Enviado el: martes, 26 de marzo de 2019 14:04 Para: Brian Meade <bmead...@vt.edu> CC: cisco-voip (cisco-voip@puck.nether.net) <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 <bmead...@vt.edu<mailto:bmead...@vt.edu>> Enviado el: martes, 26 de marzo de 2019 13:56 Para: ROZA, Ariel <ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) <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 <bmead...@vt.edu<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" <sip:86329@172.27.0.15<mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: <sip:3366@10.4.128.27<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: <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: "Gabriel Querol" <sip:86329@172.27.0.15<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&data=02%7C01%7Cariel.roza%40la.logicalis.com%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878451827&sdata=YILRc5UxLiA30sXHmbaGaM9AAyJcHE7P4mc2VEjM8To%3D&reserved=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 <ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> wrote: Here´s the trace file with the bad call De: Brian Meade <bmead...@vt.edu<mailto:bmead...@vt.edu>> Enviado el: lunes, 25 de marzo de 2019 23:39 Para: ROZA, Ariel <ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> CC: Jonatan Quezada <jonatan.quez...@chemeketa.edu<mailto:jonatan.quez...@chemeketa.edu>>; cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) <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 <ariel.r...@la.logicalis.com<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 <jonatan.quez...@chemeketa.edu<mailto:jonatan.quez...@chemeketa.edu>> Enviado el: lunes, 25 de marzo de 2019 19:24 Para: ROZA, Ariel <ariel.r...@la.logicalis.com<mailto:ariel.r...@la.logicalis.com>> CC: cisco-voip (cisco-voip@puck.nether.net<mailto:cisco-voip@puck.nether.net>) <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 <ariel.r...@la.logicalis.com<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: "XXXX XXXX" <sip:86329@172.27.0.15<mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: <sip:3366@10.4.128.27<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: <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<mailto:sip%3A86329@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<mailto:sip%3A86329@172.27.0.15>>;tag=2792862 To: <sip:3366@10.4.128.27<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&data=02%7C01%7Cariel.roza%40la.logicalis.com%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878461836&sdata=NveSUI5N%2FNPWtQ5j7yCw7a%2BcGyv%2Bvpf6tSoJ3ywRKBw%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<mailto:sip%3A3307@10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 To: <sip:*86329@172.27.0.12<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-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<mailto:sip%3A3307@10.4.128.27>> Remote-Party-ID: "XXXX XXXX (3307)" <sip:3307@10.4.128.27<mailto:sip%3A3307@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%7Cf2f9bcaf3b114e4c1bf608d6b20d60e7%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636892167878461836&sdata=J9V6cLg3fBgJsrLxbc1%2FZJTL3%2BQm2899EgwSFmS7jbY%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<mailto:sip%3A3307@10.4.128.27>>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 To: <sip:*86329@172.27.0.12<mailto:86329@172.27.0.12>>;tag=43743456 Call-ID: 6b366f80-c981b024-4f13-1b80040a@10.4.128.27<mailto: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<mailto:cisco-voip@puck.nether.net> 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