Actually meant o= line is the origin line. On Tue, Mar 26, 2019 at 12:39 PM Brian Meade <[email protected]> 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:[email protected] SIP/2.0 > Via: SIP/2.0/UDP 172.27.0.15:11347 > ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm > From: "Gabriel Querol" <sip:[email protected]>;tag=2792862 > To: <sip:[email protected]> > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]: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:[email protected]> > 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 <[email protected]> > wrote: > >> Here´s the trace file with the bad call >> >> >> >> >> >> >> >> *De:* Brian Meade <[email protected]> >> *Enviado el:* lunes, 25 de marzo de 2019 23:39 >> *Para:* ROZA, Ariel <[email protected]> >> *CC:* Jonatan Quezada <[email protected]>; cisco-voip ( >> [email protected]) <[email protected]> >> *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 <[email protected]> >> 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 <[email protected]> >> *Enviado el:* lunes, 25 de marzo de 2019 19:24 >> *Para:* ROZA, Ariel <[email protected]> >> *CC:* cisco-voip ([email protected]) <[email protected] >> > >> *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 <[email protected]> >> 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:[email protected] SIP/2.0 >> >> Via: SIP/2.0/UDP 172.27.0.15:11347 >> ;rport;branch=z9hG4bKddsEurK20207f00.sa17387ptb7Rm >> >> From: "XXXX XXXX" <sip:[email protected]>;tag=2792862 >> >> To: <sip:[email protected]> >> >> Call-ID: [email protected] >> >> CSeq: 1 INVITE >> >> Contact: <sip:[email protected]: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:[email protected]> >> >> 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:[email protected]>;tag=2792862 >> >> To: <sip:[email protected]>;tag=573234994 >> >> Date: Fri, 22 Mar 2019 19:00:23 GMT >> >> Call-ID: [email protected] >> >> 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:*[email protected]: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%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628314437&sdata=olXlS7JOxjHDQbCleDv1CJq6yZi%2B8FEzIftvZ%2FIXu8A%3D&reserved=0> >> SIP/2.0 >> >> Via: SIP/2.0/UDP 10.4.128.27:5060;branch=z9hG4bK668c2eabb0b8 >> >> From: "XXXX XXXX (3307)" <sip:[email protected] >> >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 >> >> To: <sip:*[email protected]> >> >> Date: Mon, 25 Mar 2019 10:40:36 GMT >> >> Call-ID: [email protected] >> >> 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:[email protected]> >> >> Remote-Party-ID: "XXXX XXXX (3307)" <sip:[email protected] >> >;party=calling;screen=yes;privacy=off >> >> Contact: <sip:[email protected]: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%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628324445&sdata=RZBbBCLIvH5tdrdh3bGUzyVnNrGWExWeVHkcLfyFvAU%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:[email protected] >> >;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220 >> >> To: <sip:*[email protected]>;tag=43743456 >> >> Call-ID: [email protected] >> >> 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 >> [email protected] >> https://puck.nether.net/mailman/listinfo/cisco-voip >> <https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628324445&sdata=iC5W%2B4n%2FMC%2BLlQ09phy%2BNAnT55BEdkW%2FtVhatcYPCy0%3D&reserved=0> >> >> >> >> >> -- >> >> 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 >> <https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fprojects.chemeketa.edu%2Fservicedesk%2Fcustomer%2Fportals&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628334454&sdata=V%2FCVqhnRGbXstML4SU4XxaKNA3SsfWpxii%2FyBedno8A%3D&reserved=0> >> >> >> >> Johnny Q >> >> Voice Technology Analyst II >> >> Network, Infrastructure, Routing Devices, and Servers >> >> Chemeketa Community College >> >> [email protected] >> >> Building 22 Room 130 >> >> Work 5033995294 >> >> Mobile 9712182110 >> >> SIP 5035406689 >> >> FAX 5033995549 >> >> _______________________________________________ >> cisco-voip mailing list >> [email protected] >> https://puck.nether.net/mailman/listinfo/cisco-voip >> <https://nam01.safelinks.protection.outlook.com/?url=https%3A%2F%2Fpuck.nether.net%2Fmailman%2Flistinfo%2Fcisco-voip&data=02%7C01%7CAriel.ROZA%40la.logicalis.com%7Cef4a8b91d5ee4c568bbf08d6b1944007%7C2e3290cb8d404058abe502c4f58b87e3%7C0%7C0%7C636891647628344462&sdata=45bP%2BIUxvsqVNJjsLOzeijBTJLP%2BN8ZgAPQvQbEB7ic%3D&reserved=0> >> >>
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