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List:       asterisk-dev
Subject:    Re: [asterisk-dev] [Code Review] Missing T.38 -> audio fall back
From:       "Russell Bryant" <russell () digium ! com>
Date:       2010-05-28 8:17:47
Message-ID: 20100528081747.10316.95476 () hotblack ! digium ! com
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> On 2010-03-25 10:55:22, Kevin Fleming wrote:
> > 

This got a ship it a long time ago.  Has someone committed this?  If not, can someone \
please do so?


- Russell


-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/514/#review1754
-----------------------------------------------------------


On 2010-02-24 04:17:25, vrban wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/514/
> -----------------------------------------------------------
> 
> (Updated 2010-02-24 04:17:25)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Summary
> -------
> 
> 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
> -----
> 
> /branches/1.4/channels/chan_sip.c 246298 
> 
> Diff: https://reviewboard.asterisk.org/r/514/diff
> 
> 
> Testing
> -------
> 
> 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 
> 
> Thanks,
> 
> vrban
> 
> 


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