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

Mon Mar 25 16:06:05 EDT 2019

Cause No. 65 - bearer capability not implemented.
This cause indicates that the equipment sending this cause does not support
the bearer capability requested.

What it means:

1.	In most cases, the number being called is not an ISDN number but an
analog destination.
2.	The equipment is dialing at a faster rate than the circuitry allows,
for example, dialing at 64K when only 56K is supported.

Where is the call going, out a gateway or just a Cisco phone?

From: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM> 
Sent: Monday, March 25, 2019 2:03 PM
To: NateCCIE <nateccie at gmail.com>; 'cisco-voip' <cisco-voip at puck.nether.net>
Subject: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

That was the original setting, and the results is what I included in the
mail

De: NateCCIE <nateccie at gmail.com <mailto:nateccie at gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 17:01
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM
<mailto:Ariel.ROZA at LA.LOGICALIS.COM> >; 'cisco-voip'
<cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> >
Asunto: RE: [cisco-voip] SIp Trunk call failing after PBX upgrade

I would change preferred codec to 711a and see what happens.

From: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM
<mailto:Ariel.ROZA at LA.LOGICALIS.COM> > 
Sent: Monday, March 25, 2019 1:37 PM
To: NateCCIE <nateccie at gmail.com <mailto:nateccie at gmail.com> >; 'cisco-voip'
<cisco-voip at puck.nether.net <mailto:cisco-voip at puck.nether.net> >
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 <nateccie at gmail.com <mailto:nateccie at gmail.com> > 
Enviado el: lunes, 25 de marzo de 2019 14:33
Para: ROZA, Ariel <Ariel.ROZA at LA.LOGICALIS.COM
<mailto:Ariel.ROZA at LA.LOGICALIS.COM> >; 'cisco-voip'
<cisco-voip at puck.nether.net <mailto: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-adbf28ac4c95-50893
220

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-adbf28ac4c95-50893
220

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

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