I would change preferred codec to 711a and see what happens.
From: ROZA, Ariel <[email protected]> Sent: Monday, March 25, 2019 1:37 PM To: NateCCIE <[email protected]>; 'cisco-voip' <[email protected]> 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 <[email protected] <mailto:[email protected]> > Enviado el: lunes, 25 de marzo de 2019 14:33 Para: ROZA, Ariel <[email protected] <mailto:[email protected]> >; 'cisco-voip' <[email protected] <mailto:[email protected]> > 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 <[email protected] <mailto:[email protected]> > On Behalf Of ROZA, Ariel Sent: Monday, March 25, 2019 11:17 AM To: cisco-voip ([email protected] <mailto:[email protected]> ) <[email protected] <mailto:[email protected]> > 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:[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] <mailto:[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] <mailto:[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 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-adbf28ac4c95-50893 220 To: <sip:*[email protected]> Date: Mon, 25 Mar 2019 10:40:36 GMT Call-ID: [email protected] <mailto:[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> 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-adbf28ac4c95-50893 220 To: <sip:*[email protected]>;tag=43743456 Call-ID: [email protected] <mailto:[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
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