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

[Klaus Darilion]

Am 09.02.2010 11:52, schrieb 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.

I do not understand why the hiQ rejects the normal INVITE. But if it 
works, why not.

klaus

>
> 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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>