[asterisk-dev] [Code Review] Missing T.38 -> audio fall back function for 1.4

[Kevin Fleming]

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/branches/1.4/channels/chan_sip.c
<https://reviewboard.asterisk.org/r/514/#comment3835>

    Please fix these two lines with unnecessary whitespace (shown as large red blocks on Review Board).


- Kevin


On 2010-02-24 04:17:25, vrban wrote:
> 
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> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/514/
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> 
> (Updated 2010-02-24 04:17:25)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
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> 
> When a T.38 re-INVITE failed with an 488 or 606 answer, we should fallback to audio fax by sending a re-re-INVITE with audio.
> 
> 
> This addresses bug 16692.
>     https://issues.asterisk.org/view.php?id=16692
> 
> 
> Diffs
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>   /branches/1.4/channels/chan_sip.c 246298 
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> Diff: https://reviewboard.asterisk.org/r/514/diff
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> 
> Testing
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> 
> The testing if have done: I use 1.4 asterisk with this patch between our carrier (british telecom in germany) SIP gateway with calls coming from PSTN. And if the endpoint (Linksys PAP2 ATA) want to talk T.38, we talk T.38 pass-through *1.4. And under specific circumstances our carrier can not talk T.38 with us, then we need this fall back to audio fax. 
> 
> I have a my smallest production server (100 user) now running three days with this patch. No problems so far.  
> 
> haggard has reported here:
> https://issues.asterisk.org/view.php?id=16692
> that the patch works also for him.
> 
> You can test this patch: 
> Just use two T.38 device and the one that is the callee with T.38 enabled, and the caller fax with T.38 disabled. Without the patch, the call will be hangup up by chan_sip when the caller answer "488" or "606"
> to the T.38 re-INVITE, and chan_sip hang up. With the patch chan_sip try a fall back re-re-INVITE with audio, and then the fax runs in audio mode between both fax
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> 
> Thanks,
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> vrban
> 
>