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UnivoiceLLC
Beginner

CISCO a=rtpmap:100 X-NSE/8000.

I am attempting to route calls from my IAD to a new SIP switch however calls are getting denied by the receiving switch and the only thing I have noticed is a=rtpmap:100 X-NSE/8000. instead of a=rtpmap:101 telephone-event/8000  anyone ave any ideas?

 

Here is the trace

U 96.88.111.196:51426 -> 8.27.107.6:5060
INVITE sip:19542662278@8.27.107.6:5060 SIP/2.0.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>.
Date: Fri, 01 Mar 2002 22:20:20 GMT.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
Supported: 100rel,timer,resource-priority,replaces.
Min-SE:  1800.
Cisco-Guid: 1457517616-748229078-2157491913-4163939833.
User-Agent: Cisco-SIPGateway/IOS-12.x.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER.
CSeq: 101 INVITE.
Max-Forwards: 70.
Timestamp: 1015021220.
Contact: <sip:13052055799@96.88.111.196:5060>.
Expires: 1800.
Allow-Events: telephone-event.
Content-Type: application/sdp.
Content-Disposition: session;handling=required.
Content-Length: 279.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 8383 3233 IN IP4 96.88.111.196.
s=SIP Call.
c=IN IP4 96.88.111.196.
t=0 0.
m=audio 17088 RTP/AVP 18 0 100.
c=IN IP4 96.88.111.196.
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=yes.
a=rtpmap:0 PCMU/8000.
a=rtpmap:100 X-NSE/8000.
a=fmtp:100 192-194.

#
U 8.27.107.6:5060 -> 96.88.111.196:5060
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>;tag=30e4149d349ff990e1fe106a3276098f.58da.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
CSeq: 101 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).
Content-Length: 0.
.

#############
U 96.88.111.196:51426 -> 8.27.107.6:5060
ACK sip:19542662278@8.27.107.6:5060 SIP/2.0.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>;tag=30e4149d349ff990e1fe106a3276098f.58da.
Date: Fri, 01 Mar 2002 22:20:20 GMT.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
Max-Forwards: 70.
CSeq: 101 ACK.
Allow-Events: telephone-event.
Content-Length: 0.

 

2 REPLIES 2
Terry Cheema
Advocate

What is your topology/Call Flows? What is the call control server CUCM/CUCME or the phones are connected to IAD? Are you using the SIP switch as a SBC? Can you post the config of the IAD and also make a test call and post the complete logs:

debug ccsip messages

From the logs, the SIP server appears to be rejecting the calls with a 503 response, which can be due to various reasons. If you can post the above information, you will get a better response.

-Terry

Please rate all helpful posts.

Nadeem Ahmed
Cisco Employee


Seems like your SIP Switch(Server) is sendng 503 Service Unavailable not because of Media Attributes value it its not understand or accept the value in the parameter ideally
it should send 488 Not acceptable.

Is SIP switch is Cisco or non-Cisco Device? are they able to communicate (Ping and all) Since 503 service unavaiable your SIP Switch is unable to process the call due to some issue
like TCP communication is not getting estabilished, port blocked, IP communication or ACL.


SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>;tag=30e4149d349ff990e1fe106a3276098f.58da.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
CSeq: 101 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).


Verify other way around sending a from other side and see if you get the same behaviour.
 

 

Br,

Nadeem

SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>;tag=30e4149d349ff990e1fe106a3276098f.58da.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
CSeq: 101 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).
Content-Length: 0. - See more at: https://supportforums.cisco.com/discussion/12523611/cisco-artpmap100-x-nse8000#sthash.D4K4xiOW.dpuf
SIP/2.0 503 Service Unavailable.
Via: SIP/2.0/UDP 96.88.111.196:5060;x-route-tag="tgrp:DEFAULT";branch=z9hG4bK13B8A.
From: <sip:13052055799@96.88.111.196>;tag=4CB2000-F96.
To: <sip:19542662278@8.27.107.6>;tag=30e4149d349ff990e1fe106a3276098f.58da.
Call-ID: 5A2DE53E-2C9911D6-809DB6C9-F830ADF9@96.88.111.196.
CSeq: 101 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).
Content-Length: 0. - See more at: https://supportforums.cisco.com/discussion/12523611/cisco-artpmap100-x-nse8000#sthash.D4K4xiOW.dpuf
Br, Nadeem Please rate all useful post.
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