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

[Klaus Darilion]

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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>