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

Jonatan Quezada jonatan.quezada at chemeketa.edu
Mon Mar 25 18:24:06 EDT 2019


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.ROZA at 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 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
>
> 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
>
> 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
>
> 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
>
> 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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>


-- 
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

Johnny Q
Voice Technology Analyst II
Network, Infrastructure, Routing Devices, and Servers
Chemeketa Community College
Johnny.Q at chemeketa.edu
Building 22 Room 130
Work 5033995294
Mobile 9712182110
SIP 5035406689
FAX 5033995549
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