For some reason, the aastra phone is sending a different contact  and via address after it gets challenged by freeswitch.

Original address was

Via: SIP/2.0/UDP 7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport   (LINE# 35)

Then it changes to

Via: SIP/2.0/UDP 192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport (LINE# 187)

It then sends a CANCEL using Via: SIP/2.0/UDP 7.7.7.7:6060 (LINE# 961) which freeswitch isn't able to match to a transaction because it is probably expecting 192.168.0.175.



On 12/02/2011 07:19 PM, Charlie Orford wrote:
Hi Anthony

We are using Aastra 57i phones with the latest firmware (v3.2.2.56 from June 2011).

However, it looks like a specific issue with the Aastra 57i as I have just replicated the call and cancel situation using a 3CX softphone and in this case the CANCEL request gets honoured by FS.

Here's a copy of a failed CANCEL test using the Aastra phone: http://pastebin.freeswitch.org/17922

Here's a copy of a successful CANCEL test using the 3CX softphone: http://pastebin.freeswitch.org/17923

sofia loglevel all 9 is on for both tests.

Charlie


On 01/12/2011 21:53, Anthony Minessale wrote:
try same failed call test with sofia loglevel all 9

Also, try some other phone or device that does not have the problem, and create a cancel situation the same way and see if you can find a difference.

Finally, have you tried the latest firmware on the phone?


On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford <charlie.orford@attackplan.net> wrote:
Hi list,

When we make a call from an FS extension to a PSTN number (via our ITSP
gateway provider) and hang-up before the call completes, FS replies to
the CANCEL request with "481 Call/Transaction Does Not Exist" and the
call continues to ring on the remote end. If we hangup after the call
has connected, it works with no problem.

I have looked through a SIP trace of this happening and to my (untrained
eye) nothing seems obviously wrong (i.e. tag, call id and branch values
all seem to be correct). I'm using the latest git snapshot from
2011-11-30 18-14-24 -0600.

For a sip trace showing the problem, see:
http://pastebin.freeswitch.org/17908

For a sip trace showing a successful hangup, see:
http://pastebin.freeswitch.org/17909

*Note: ip addresses, domain and called number have been altered for privacy.

Any help or insight is much appreciated.

Kind Regards,
Charlie


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_________________________________________________________________________
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consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

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http://wiki.freeswitch.org
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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