[Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects

Fri Dec 2 14:50:11 MSK 2011

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/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport   
(LINE# 35)

Then it changes to

Via: 
SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport 
(LINE# 187)

It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (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 at attackplan.net 
>> <mailto:charlie.orford at 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
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
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>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
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>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
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>>
>> AIM: anthm
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>>
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>> pstn:+19193869900
>>
>>
>> _________________________________________________________________________
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>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
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>
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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