Hi,
We are trying to configure an authenticated SIP trunk towards the ITSP. The registration looks good. We are able to receive incoming calls but outbound calls are failing.
Here is the relevant configuration:
sip-ua
credentials username 55507210 password 7 xxx realm sipprovider.com
retry register 10
registrar 1 dns:sipprovider.com:6321 expires 3600
sip-server dns:sipprovider.com:6321
dial-peer voice 2001 voip
translation-profile incoming IN
translation-profile outgoing OUT
max-conn 10
destination-pattern [925678].......
modem passthrough nse codec g711ulaw
session protocol sipv2
session target sip-server
session transport udp
incoming called-number .
dtmf-relay sip-notify sip-kpml rtp-nte h245-signal
codec g711ulaw
no vad
And here is the debug ccsip mess output:
----------------------------------------------------------------------------------------
Sent:
INVITE sip:22224444@sipprovider.com:6321 SIP/2.0
Via: SIP/2.0/UDP 166.166.166.166:5060;branch=z9hG4bK657A671
Remote-Party-ID: "Office" <sip:55507210@166.166.166.166>;party=calling;screen=no;privacy=off
From: "Office" <sip:55507210@166.166.166.166>;tag=1716A8D4-2347
To: <sip:22224444@sipprovider.com>
Date: Mon, 05 Jul 2021 12:56:52 GMT
Call-ID: 4DAEF96A-DCC711EB-8353AF28-E51E2F2F@166.166.166.166
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 1287072012-3704033771-2202971944-3843960623
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: 1625489812
Contact: <sip:55507210@166.166.166.166:5060>
Call-Info: <sip:166.166.166.166:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Expires: 180
Allow-Events: kpml, telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 298
v=0
o=CiscoSystemsSIP-GW-UserAgent 666 7875 IN IP4 166.166.166.166
s=SIP Call
c=IN IP4 166.166.166.166
t=0 0
m=audio 17166 RTP/AVP 0 100 101
c=IN IP4 166.166.166.166
a=rtpmap:0 PCMU/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
Jul 5 15:56:52.696: //26030/4CB7290C834E/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 No relaying
Via: SIP/2.0/UDP 166.166.166.166:5060;received=166.166.166.166;rport=56169;branch=z9hG4bK657A671
From: "Office" <sip:55507210@166.166.166.166>;tag=1716A8D4-2347
To: <sip:22224444@sipprovider.com>;tag=9e08.c5bf3bfd769785537187c6c20318433e
Call-ID: 4DAEF96A-DCC711EB-8353AF28-E51E2F2F@166.166.166.166
CSeq: 101 INVITE
Server : SIPProvider proxy
Content-Length: 0
Jul 5 15:56:52.712: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:22224444@sipprovider.com:6321 SIP/2.0
Via: SIP/2.0/UDP 166.166.166.166:5060;branch=z9hG4bK657A671
From: "Office" <sip:55507210@166.166.166.166>;tag=1716A8D4-2347
To: <sip:22224444@sipprovider.com>;tag=9e08.c5bf3bfd769785537187c6c20318433e
Date: Mon, 05 Jul 2021 12:56:52 GMT
Call-ID: 4DAEF96A-DCC711EB-8353AF28-E51E2F2F@166.166.166.166
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: kpml, telephone-event
Content-Length: 0
-----------------------------------------------------------------------------
Im not able to find any documentation for the error "SIP/2.0 403 No relaying"
Kindly assist.
Regards
SAIF
