07-03-2013 02:05 AM - last edited on 03-25-2019 08:23 PM by ciscomoderator
Dear Experts,
I have configured SIP trunk with authentication. the authentication is username and password. I use CUBE to acheive this.
the call flow is like the following
phone ---> CCM 6.1 -----> H323 ----> CUBE -----> Firewall -----> Internet ------> SIP provider.
while I do troubleshooting, I found a message told me that the firewall traversal is not enabled.
I do stun configuration by using the following commands
voice sevice voip
stun
stun flowdata agent-id 10
voice class stun-usage 1
stun usage firewall-traversal flowdata
after that, the firewall traversal message disappear from the logs. but the call is not working.
the sip provider has a STUN server, how we can use it ? and also what is the STUN ?
thanks in advance
Anas
07-03-2013 08:12 AM
Do you have captures from CUBE for a failed call?
Tapan
07-03-2013 08:56 AM
It shows me a dead call, when I do.debug ccsip calls.
regards
Anas
Sent from Cisco Technical Support Android App
07-03-2013 11:04 AM
That reply is not helpful,
capture debug ccsip messages
And provide the logs.
--
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.
07-07-2013 01:24 AM
07-10-2013 04:31 AM
Hi Guys,
can any one help me with this case ?
thanks in advance
regards
Sent from Cisco Technical Support Android App
07-10-2013 05:16 AM
Hi
The version of the CUBE is 15.x?
Please rate all useful posts
Regards
Chrysostomos
""The Most Successful People Are Those Who Are Good At Plan B""
07-10-2013 05:20 AM
Here's the issue:
Sent:
INVITE sip:009627XXXXXXX@sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.127:5060;branch=z9hG4bKF1147
Remote-Party-ID: <00494XXXXXXX1>;party=calling;screen=yes;privacy=off00494XXXXXXX1>
From: <>>00494XXXXXXX1@sipgate.de>;tag=3A5AA8-139A
To: <>>009627XXXXXXX@sipgate.de>
Date: Sun, 07 Jul 2013 08:54:45 GMT
Call-ID: B47C60AE-E61911E2-8014F945-EAF1556@192.168.33.127
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 15705573-2133102877-100672513-3232243830
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: 1373187285
Contact: <00494XXXXXXX1>00494XXXXXXX1>
Expires: 180
Allow-Events: telephone-event
Proxy-Authorization: Digest username="1430052e1",realm="sipgate.de",uri="sip:009627XXXXXXX@sipgate.de:5060",response="90b394921ae0523993f5f44fe1715f41",nonce="51d926d735a262f3a6904b27adba2bb98012f4a7",algorithm=md5
Content-Length: 0
Received:
SIP/2.0 403 Forbidden (check from field)
Via: SIP/2.0/UDP 192.168.33.127:5060;rport=49786;received=192.168.33.127;branch=z9hG4bKF1147
From: <>>00494XXXXXXX1@sipgate.de>;tag=3A5AA8-139A
To: <>>009627XXXXXXX@sipgate.de>;tag=6d6e7f8f352adddb20da2b196524dfa8.b769
Call-ID: B47C60AE-E61911E2-8014F945-EAF1556@192.168.33.127
CSeq: 102 INVITE
Content-Length: 0
Seems like the SIP Server does'n like the FROM header:
From: <>>00494XXXXXXX1@sipgate.de>;tag=3A5AA8-139A
You can modify the FROM header with a SIP Profile:
voice class sip-profiles 1111
request ANY sip-header From modify "sip:(.*)@" "sip:[whatever they want to receive]@"
You may ask your Service Provider what they don't like on the INVITE, then you can modify it accordingly.
Some information if STUN:
http://www.voip-info.org/wiki/view/STUN
HTH
--
Jorge Armijo
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