[asterisk-dev] fallback to audio faxing when T.38 INVITE fail with 488/606 Not acceptable

[Kristijan Vrban]

I was struggling to get the T.38 fallback working with a Siemens Media
Gateway hiQ 9200 (used by british telecom in germany)
and my fallback re-re-INVITE to PCMA always faild with a "500 Internal
Server Error" then i read again:

http://tools.ietf.org/html/draft-ietf-sipping-realtimefax-01#section-6.2

An the i added hardcoded

silenceSupp:off - - - -

into the SDP in the fallback INVITE. Then it was working!

So i want to ask, if you agree than we need to add this in case of a
T.38 fallback INVITE? Then i can try to make a patch.

Kristijan


2010/1/25 Kristijan Vrban <vrban.lkml at googlemail.com>:
> hello klaus,
>
> check: https://issues.asterisk.org/view.php?id=16692
>
> kristijan
>
> 2010/1/21 Klaus Darilion <klaus.mailinglists at pernau.at>:
>>
>>
>> Kristijan Vrban schrieb:
>>> i can offer the patch for 1.4 on issue.asterisk.org but digium must
>>> decide if this is a bugfix or a feature.
>>> @digium bugfix or feature?
>>
>> a missing ACK is always a bug.
>>
>> please post the patch at the bugtracker
>>
>> regards
>> klaus
>>
>>>
>>> Kristijan
>>>
>>> p.s.
>>> and yes, i also would like to upgrade to 1.6. but this decision taken
>>> by the management team, and unfortunately not the technical staff :-(
>>>
>>> 2010/1/18 Klaus Darilion <klaus.mailinglists at pernau.at>:
>>>>
>>>> Kristijan Vrban schrieb:
>>>>> I took a look into 1.6 source. 1.6 does a re-re-INVITE to audio after
>>>>> its T.38 re-INVITE failed. A backport is simple. But i think in
>>>>> allready know the answer. No, this is a new feature, no new features
>>>>> into 1.4 So one more backport patch into my patch folder for my
>>>>> asterisk 1.5 :)
>>>> IMO this isn't a new feature, but a bugfix.
>>>>
>>>> I experienced same problems too.
>>>>
>>>> regards
>>>> klaus
>>>>
>>>>> kristijan
>>>>>
>>>>> 2010/1/13 Kevin P. Fleming <kpfleming at digium.com>:
>>>>>> Kristijan Vrban wrote:
>>>>>>>> However, the original message was a bit unclear if it was a re-invite or an invite.
>>>>>>> re-invite
>>>>>>>
>>>>>>> And there is no difference if chan_sip get a 488 or 606 for the T.38
>>>>>>> re-INVITE (both are valid response) . As far as i can see, there is
>>>>>>> simply no logic that handle a fallback to audio fax when the caller
>>>>>>> can not handle T.38? I examined how a Cisco/Linksys SPA2120 handel
>>>>>>> this: It ACK the 488/606 for its rejected T.38 re-INVITE and send a
>>>>>>> re-re-INVITE with PCMU, and then the fax is transported via audio RTP.
>>>>>> If Asterisk sends a re-INVITE to T.38, and the other end rejects it,
>>>>>> there is nothing to be done to 'fallback' to audio mode; the call is
>>>>>> still in audio mode, because it never left audio mode.
>>>>>>
>>>>>> Also, you are doing this testing with Asterisk 1.4, which has very
>>>>>> limited support for T.38; the T.38 support in Asterisk 1.6.x is vastly
>>>>>> improved, and I'd highly encourage you to use instead if you can.
>>>>>>
>>>>>> --
>>>>>> Kevin P. Fleming
>>>>>> Digium, Inc. | Director of Software Technologies
>>>>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
>>>>>> skype: kpfleming | jabber: kpfleming at digium.com
>>>>>> Check us out at www.digium.com & www.asterisk.org
>>>>>>
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>>>
>>
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