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SIP call is getting disconnected

Syed
Level 3
Level 3

Dear All,

 

I am making call out, but call is getting disconnected .please let me know the reason

Thank you in advance..

 

Feb 20 09:38:05.654: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:2002@10.106.115.201 SIP/2.0

Via: SIP/2.0/UDP 10.65.35.183:5060;branch=z9hG4bK000024dc

From: "4333" <sip:4333@10.106.115.201>;tag=e82aea9ab1f400190000605d-0000123c

To: <sip:2002@10.106.115.201>

Call-ID: e82aea9a-b1f40016-00004437-00001e39@10.65.35.183

Max-Forwards: 70

Date: Sat, 20 Feb 2016 09:45:36 GMT

CSeq: 101 INVITE

User-Agent: Cisco-SIPIPCommunicator/9.1.1

Contact: <sip:990889B0-18C4@10.65.35.183:5060;transport=udp>

Expires: 180

Accept: application/sdp

Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO

Remote-Party-ID: "4333" <sip:4333@10.106.115.201>;party=calling;id-type=subscriber;privacy=off;screen=yes

Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.5.1

Allow-Events: kpml,dialog

Content-Length: 376

Content-Type: application/sdp

Content-Disposition: session;handling=optional

 

v=0

o=Cisco-SIPUA 8821 0 IN IP4 10.65.35.183

s=SIP Call

t=0 0

m=audio 28504 RTP/AVP 18 0 8 9 116 124 101

c=IN IP4 10.65.35.183

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:116 iLBC/8000

a=fmtp:116 mode=20

a=rtpmap:124 ISAC/16000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

 

 

 

Feb 20 09:38:05.658: //3867/7A4395D294C0/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 10.65.35.183:5060;branch=z9hG4bK000024dc

From: "4333" <sip:4333@10.106.115.201>;tag=e82aea9ab1f400190000605d-0000123c

To: <sip:2002@10.106.115.201>

Date: Sat, 20 Feb 2016 09:38:05 GMT

Call-ID: e82aea9a-b1f40016-00004437-00001e39@10.65.35.183

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.4.3.M3

Content-Length: 0

 

 

 

Feb 20 09:38:05.662: //3867/7A4395D294C0/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 503 Service Unavailable

Via: SIP/2.0/UDP 10.65.35.183:5060;branch=z9hG4bK000024dc

From: "4333" <sip:4333@10.106.115.201>;tag=e82aea9ab1f400190000605d-0000123c

To: <sip:2002@10.106.115.201>;tag=9996E100-EF3

Date: Sat, 20 Feb 2016 09:38:05 GMT

Call-ID: e82aea9a-b1f40016-00004437-00001e39@10.65.35.183

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-15.4.3.M3

Reason: Q.850;cause=47

Content-Length: 0

4 Replies 4

Chris Deren
Hall of Fame
Hall of Fame

You are getting "SIP/2.0 503 Service Unavailable" which can mean many things, most likely that the destination you are sending the call to (10.106.115.201) is incorrect or not configured to take your calls.

What is this destination, as it appears you are only sending 4 digits to it (2002), please describe the topology.

Hi Deren,

Please find the attached call flow and running-config..

Hi,

Your CME isn't provisioned for SIP endpoints. I don't see any 'voice register ...' command in your CME. I am not sure where is your CIPC registered but based on your config it can't be on CME.

There is no extension 2002 defined in your configuration, and since the INVITE is to the internal IP address I am assuming this is an inbound call, and since this is from extension 4333 then it is within the system. What/who are you trying to call?