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

Klaus Darilion klaus.mailinglists at pernau.at
Thu Jan 21 12:57:05 CST 2010



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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