[cisco-voip] SIp Trunk call failing after PBX upgrade

Mon Mar 25 15:37:14 EDT 2019

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 <nateccie at gmail.com>
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM>; 'cisco-voip' <cisco-voip at puck.nether.net>
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 <cisco-voip-bounces at puck.nether.net<mailto:cisco-voip-bounces at puck.nether.net>> On Behalf Of ROZA, Ariel
Sent: Monday, March 25, 2019 11:17 AM
To: cisco-voip (cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>) <cisco-voip at puck.nether.net<mailto:cisco-voip at puck.nether.net>>
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:3366 at 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 at 172.27.0.15>;tag=2792862
To: <sip:3366 at 10.4.128.27>
Call-ID: 501227892-15 at 172.27.0.15<mailto:501227892-15 at 172.27.0.15>
CSeq: 1 INVITE
Contact: <sip:86329 at 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 at 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 at 172.27.0.15>;tag=2792862
To: <sip:3366 at 10.4.128.27>;tag=573234994
Date: Fri, 22 Mar 2019 19:00:23 GMT
Call-ID: 501227892-15 at 172.27.0.15<mailto:501227892-15 at 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 at 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 at 10.4.128.27>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: <sip:*86329 at 172.27.0.12>
Date: Mon, 25 Mar 2019 10:40:36 GMT
Call-ID: 6b366f80-c981b024-4f13-1b80040a at 10.4.128.27<mailto:6b366f80-c981b024-4f13-1b80040a at 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 at 10.4.128.27>
Remote-Party-ID: "XXXX XXXX (3307)" <sip:3307 at 10.4.128.27>;party=calling;screen=yes;privacy=off
Contact: <sip:3307 at 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 at 10.4.128.27>;tag=429005~b085ab57-efd9-4eb3-9a97-adbf28ac4c95-50893220
To: <sip:*86329 at 172.27.0.12>;tag=43743456
Call-ID: 6b366f80-c981b024-4f13-1b80040a at 10.4.128.27<mailto:6b366f80-c981b024-4f13-1b80040a at 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20190325/92f3b0f3/attachment.html>