02-20-2016 10:28 AM - edited 03-18-2019 11:48 AM
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
02-20-2016 02:33 PM
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.
02-21-2016 02:56 AM
02-21-2016 05:30 AM
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.
02-21-2016 06:42 AM
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?
Discover and save your favorite ideas. Come back to expert answers, step-by-step guides, recent topics, and more.
New here? Get started with these tips. How to use Community New member guide