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

Charlie Orford charlie.orford at attackplan.net
Mon Dec 5 15:33:54 MSK 2011 

Hi Joegen,

I just wanted to say a quick thank you for your advice - it worked. As 
soon as I turned off all the NAT related stuff on the Aastra phone 
(basically turned off rport and removed the stun and turn servers), the 
phone stopped changing its VIA header and CANCEL worked as advertised.

In case it's useful to anyone else in a similar position, I have the 
following NAT related stuff enabled on the internal FS sip profile 
(note: my FS server is on a public IP i.e. not itself behind NAT):

<param name="NDLB-received-in-nat-reg-contact" value="true"/>
<param name="NDLB-force-rport" value="true"/>
<param name="nat-options-ping" value="true"/>

Charlie

On 03/12/2011 12:37, Charlie Orford wrote:
> Bummer - we've got an office full of these phones :(
>
> As you suggest, I'll turn off all the NAT stuff when I'm there on 
> Monday and see if that fixes it - hope so, otherwise I guess it looks 
> like I'll have to try and get Aastra to look at the problem (doubt 
> they are going to be very interested though!).
>
> Thanks,
> Charlie
>
>
> On 03/12/2011 06:29, Joegen Baclor wrote:
>> It's definitely a bug on the phone.  I would try setting all Aastra 
>> NAT related features off and see if it stops getting confused like 
>> this.  Freeswitch should be able to handle NAT from the far end.
>>
>> On 12/02/2011 08:18 PM, Charlie Orford wrote:
>>> Thanks Joegen, I hadn't picked up on that.
>>>
>>> Is there any valid reason why the phone would swap to using it's 
>>> internal IP (192.168.0.175) for the Via and Contact headers when 
>>> responding to the auth challenge? Or does this look more like a bug 
>>> with the phone?
>>>
>>>
>>>
>>> On 02/12/2011 12:50, Joegen Baclor wrote:
>>>> 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:
>>>>>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>>>>     http://www.freeswitchsolutions.com
>>>>>>
>>>>>>     
>>>>>>     
>>>>>>
>>>>>>     Official FreeSWITCH Sites
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>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> -- 
>>>>>> Anthony Minessale II
>>>>>>
>>>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>>>
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>>>>>>
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>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
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>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
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>>>>>
>>>>>
>>>>>
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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
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>>>>
>>>
>>>
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>>>
>>> 
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>
>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
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