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CUCM not sending ACK for an outgoing INVITE

Hi,

 

I am facing a problem where the CUCM ignore the response 200 OK coming from a SIP trunk, at an INVITE previously sent. Thus the call fail:

CUCM not answering picture

 

The thing is that I don't find any traces relaying the problem (apart this SIP trace). Where can I find the relevant logs (in RTMT for instance) which would output the error, so that I get a starting point ? And what could be the cause of this behaviour ?

The CUCM version is 10.5.1.10000-7.

8 Replies 8

The SIP 480 Temporarily unavailable indicates that the callee was contacted but is temporarily unavailable.
 

You can collect the CCM trace from RTMT to start with.
RTMT->Trace & Log Central->collect files->Select Cisco Call manager and collect the traces.

 

-Venkatesh

Thank you for your help venperum.

 

The problem is that I don't understand why the CUCM is ignoring the responses, although it receives a correct 200 OK with the SDP negociation. I think it only needs to accept it and warn the client which initiated the call, but instead the answer is ignored.

 

To illustrate, here is the SIP INVITE sent to the callee by the CUCM, and the answer sent by the callee to the CUCM:

 

INVITE sip:username@my.sip.domain SIP/2.0
Via: SIP/2.0/TCP 197.17.144.4:5060;branch=z9hG4bK27755eb2053
From: "Anita Perez - x6017" <sip:+14085556017@197.17.144.4>;tag=693~67f921d2-679e-4fdb-a0e9-0df0d4a1fe0a-19642095
To: <sip:username@my.sip.domain>
Date: Wed, 12 Aug 2015 07:10:37 GMT
Call-ID: 39efda00-5ca1f16d-278-38512c6@197.17.144.4
Supported: 100rel,timer,resource-priority,replaces
Min-SE:  1800
User-Agent: Cisco-CUCM10.5
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
Call-Info: <sip:197.17.144.4:5060>;method="NOTIFY;Event=telephone-event;Duration=500"
Call-Info: <urn:x-cisco-remotecc:callinfo>;x-cisco-video-traffic-class=DESKTOP
Cisco-Guid: 0972020224-0000065536-0000000001-0059052742
Session-Expires:  1800
P-Asserted-Identity: "Anita Perez - x6017" <sip:+14085556017@197.17.144.4>
Remote-Party-ID: "Anita Perez - x6017" <sip:+14085556017@197.17.144.4>;party=calling;screen=yes;privacy=off
Contact: <sip:+14085556017@197.17.144.4:5060;transport=tcp>;video;audio;bfcp
Max-Forwards: 69
Content-Type: application/sdp
Content-Length: 879

v=0
o=CiscoSystemsCCM-SIP 693 1 IN IP4 197.17.144.4
s=SIP Call
c=IN IP4 10.16.7.210
b=TIAS:384000
b=AS:384
t=0 0
m=audio 24608 RTP/AVP 0 8 18 104 105 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 G7221/16000
a=fmtp:105 bitrate=24000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 14216 RTP/AVP 126 97
b=TIAS:384000
a=label:11
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42E01F;packetization-mode=1;max-fs=3601;max-rcmd-nalu-size=32000;level-asymmetry-allowed=1
a=imageattr:126 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42E01F;packetization-mode=0;max-fs=3601;level-asymmetry-allowed=1
a=imageattr:97 recv [x=[32:1:1280],y=[18:1:720],par=1.7778,q=1.00]

a=content:main

 

And the answer:

 

SIP/2.0 200 Ok
Via: SIP/2.0/TCP 197.17.144.4:5060;received=173.38.218.1;rport=39207;branch=z9hG4bK27755eb2053
From: "Anita Perez - x6017" <sip:+14085556017@197.17.144.4>;tag=693~67f921d2-679e-4fdb-a0e9-0df0d4a1fe0a-19642095
To: <sip:username@my.sip.domain>;tag=cUh~ZLg
Call-ID: 39efda00-5ca1f16d-278-38512c6@197.17.144.4
CSeq: 101 INVITE
User-Agent: Linphone/3.8.5 (belle-sip/1.4.1)
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: <sip:username@my.sip.domain;alias=172.16.4.100~5060~1>;+sip.instance="<urn:uuid:b1a2a96b-81b7-4715-900e-a1e7cfadf619>"
Content-Type: application/sdp
Content-Length: 303
Record-route: <sip:172.16.5.60;r2=on;lr;nat=yes>
Record-route: <sip:172.16.5.60;transport=tcp;r2=on;lr;nat=yes>

v=0
o=username 2578 3554 IN IP4 172.16.4.100
s=Talk
c=IN IP4 172.16.4.100
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 126 97
a=rtpmap:126 H264/90000
a=fmtp:126 profile-level-id=42801F
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F

 

It seems ok, but still, the CUCM acts as if it were not receiving any response (hence the 480). See the several 200 OK on the picture of my first post.

 

Thanks.

 

Note: I replaced the IP addresses and domains by fake ones.

Well, I was wrong. The above SIP message is indeed not correct: the IP in the Record-route header, as well as the IP is SDP, was the private IP of my Kamailio.

I finally saw in RTMT logs that the CUCM was trying to send the ACK to this private IP, which obviously was failing. So the issue had nothing to do with the CUCM, but with Kamailio which was not giving the correct IP.

 

(For those interested, the actual culprit was RTPproxy, being behind a nat. More info here).

So what if the person on the other side is available? Are you saying that if I left this for a week or month is would be available at some point?

Why are you responding to a post from way back in 2015? Also that’s not at all what the 480 Temporary Unavailable response means and likely you know that already, based on your response IMHO I get a sense that your trying to make fun of someone, which is totally uncalled for in my view.



Response Signature


I responded because I was having the same issue and I was hoping there would be some helpful troubleshooting or a little more knowledge as to why it happened. Not someone wording the error in a different way. I see you on a lot of the community post, is that a job in cisco, or are you that enthusiastic about your job?


@Adrian Michael Santiago wrote:

I see you on a lot of the community post, is that a job in cisco, or are you that enthusiastic about your job?


Just being enthusiastic I guess.



Response Signature


Good day Roger, please keep on being an enthusiast! Your contributions are greatly needed and very much appreciated in this community (and I'm sure, any other community you are a part of and graciously serve). We all thank you Roger, please don't stop!