03-28-2012 02:48 AM - edited 03-21-2019 05:34 AM
Hi,
I have a UC540 working over a SIP trunk to Internode. I am having major problems with being able to call forward to any outside number.
I thought this was a natting issue as the UC540 sits behind a C887 router and the router was controlling the nat. I have since added a routed subnet so there is no natting involved on the UC540 or the Router. Still no change to the call forward issue
I have attached a debug of a call coming into the UC540 and the being sent back out with a call forward. On the call forward I get a SIP/2.0 401 Unauthorized.
I have also attached the config of UC540.
Thanks, Ron.
03-28-2012 05:15 AM
After received the 401 unauthorized message, your UC sends the new invite with authentication but it doesn't get any answer.
Can you enable the debug ip nat sip on your 877 and post the output?
Regards.
03-28-2012 06:17 AM
Hello,
I don't get anything from that debug, I am not doing any natting on the Cisco 887.
The UC540 has a public IP address and not being translated.
When I was using a natted environment & was translating for the UC540 I had the same symptoms, always call forward to external number fails.
I have attached diagram showing setup of network.
03-28-2012 07:08 AM
Are normal outgoing call working?
Can be a problem of ITSP authorization?
In which way your ITSP block or permit a call? IP ADDRESS? User part of FROM Header?
Can we compare a normal outgoing call and a forwarding call?
Regards.
03-28-2012 07:13 AM
Hi there, yes outgoing calls are working, only call forwarded doen't work.
I have attached a debug of normal call and a call forward.
Thanks
PS attchments 82365012 calls 0400733889 is a normal call that is successful
0400733889 calls 82365012 which is diverted to 82027371 is a call forward that fails.
03-28-2012 09:52 AM
Sound like an ITSP problem. These are the two INVITE:
failed outgoing invite
INVITE sip:82027371@sipconnect.internode.on.net:5060 SIP/2.0
Via: SIP/2.0/UDP 150.101.237.61:5060;branch=z9hG4bKD63119AA
From: "0400733889" <>>0882365000@sipconnect.internode.on.net>;tag=4E738118-13C2
To: <>>82027371@sipconnect.internode.on.net>
Date: Wed, 28 Mar 2012 09:04:42 GMT
Call-ID: E44329E1-77EB11E1-A54DABCC-BC919705@sipconnect.internode.on.net
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3829526929-2011894241-2773068748-3163657989
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Timestamp: 1332925482
Contact: <0882365000>0882365000>
Diversion: <>>0882365000@sipconnect.internode.on.net>;privacy=off;reason=unconditional;counter=1;screen=no
Expires: 180
Allow-Events: telephone-event
Authorization: Digest username="W4184527",realm="BroadWorks",uri="sip:82027371@sipconnect.internode.on.net:5060",response="e0aaf0db0d9297e51544c185ad57d1b0",nonce="BroadWorksXh0c5fhxzTwqv75fBW",cnonce="DF5C3CB9",qop=auth,algorithm=MD5,nc=00000001
Max-Forwards: 8
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287
v=0
o=CiscoSystemsSIP-GW-UserAgent 4228 9362 IN IP4 150.101.237.61
s=SIP Call
t=0 0
m=audio 17460 RTP/AVP 8 0 18 101
c=IN IP4 150.101.237.61
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Working outgoing invite:
INVITE sip:0400733889@sipconnect.internode.on.net:5060 SIP/2.0
Via: SIP/2.0/UDP 150.101.237.61:5060;branch=z9hG4bKD61F13BD
From: "Payments B" <>>0882365000@sipconnect.internode.on.net>;tag=4E6BDE24-832
To: <>>0400733889@sipconnect.internode.on.net>
Date: Wed, 28 Mar 2012 08:56:22 GMT
Call-ID: B9F498F6-77EA11E1-A53AABCC-BC919705@sipconnect.internode.on.net
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3107494923-2011828705-2771758028-3163657989
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1332924982
Contact: <0882365000>0882365000>
Expires: 180
Allow-Events: telephone-event
Authorization: Digest username="W4184527",realm="BroadWorks",uri="sip:0400733889@sipconnect.internode.on.net:5060",response="59c990ca0b7b1210c5228a1067217564",nonce="BroadWorksXh0c54rrlTwg35h7BW",cnonce="B6CA980C",qop=auth,algorithm=MD5,nc=00000001
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 287
v=0
o=CiscoSystemsSIP-GW-UserAgent 5877 1433 IN IP4 150.101.237.61
s=SIP Call
t=0 0
m=audio 16822 RTP/AVP 8 0 18 101
c=IN IP4 150.101.237.61
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
I highlighted the differences but I don't see any problem.
Simply your provider doesn't answer to your INVITE. Do you already contact the ITSP?
Maybe it doesn't support the DIVERSION header.
Try to open a ticket and see what they say.
Regards.
03-28-2012 04:35 PM
Hi Ron,
On the failed called:
82027371@sipconnect.internode.on.net:5060 SIP/2.0
The problem is there is no Zero, this is the same problem we faced with Internode some time ago, you have to even for local calls sometimes add the STD "0" code in there, just make sure that the leading steering "0" is stripped so the call does not go out looking like "0082027371" there should only be 1 "0" there
Other than that Internet have a robust SIP trunk that works very well
Cheers,
David.
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