[Asterisk-Dev] Bugfix for CVS-HEAD-06/26/04-21:56:45

[programmer_ted]

At 01:07 AM 6/30/2004, you wrote:
>Hi,
>
>My friend and I were getting a warning when calling his Sipura from a PSTN 
>line (connecting to Asterisk through BroadVoice), that said:
>
>Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)
>
>and was followed by a hangup (type 64 is 16-bit Signed Linear PCM, type 4 
>is G711u).  I found that many people have had similar issues, but these 
>were never resolved.  So, I figured that because Asterisk is open-source, 
>I'd dive into the code and try to fix the bug.
>
>After a couple of hours of familiarizing myself with the Asterisk code and 
>tracing the problem, I found that for some reason the tone generator, 
>which uses 16-bit Signed Linear PCM, was still being allocated and 
>playtones_generator (indications.c) was still getting called, regardless 
>that the Sipura doesn't take SLINEAR data (in my case, it accepts 
>G711u).  So, I ended up adding an if conditional to the beginning of the 
>playtones_alloc function (indications.c) to check if SLINEAR was supported 
>by the channel, and if not, return 0 (which, when received by the 
>ast_activate_generator function (channel.c), causes the channel 
>generatordata to remain empty, effectively stopping the SLINEAR data from 
>being sent in the most nonintrusive way possible).
>
>NOTICE: this bugfix will work for similar issues involving format 64 
>(16-bit Signed Linear PCM) being sent even if channel capabilities don't 
>allow it, if the generator is involved - it's not limited to my situation 
>(dialing the Sipura from Asterisk).
>
>This patch should be applied to indications.c under the main asterisk 
>source directory (usually /usr/src/asterisk):
>
>68a69
> >       if (!(chan->nativeformats & AST_FORMAT_SLINEAR)) return 0;
>
>Oh, and finally, here's a shameless plug to a good friend's website (which 
>includes a VOIP forum!): http://outcast.ws
>
>Comments?  Questions?  :)
>
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Just a quick update.  I was looking things over again and it appears this 
fix also disables the generator when I'm calling in on PSTN over BroadVoice 
(when dialing the Sipura), not just disabling it for the Sipura.  This 
basically disables the dialing sound while waiting for the Sipura to pick 
up.  I have an idea that I should have used chan->capabilities rather than 
chan->nativeformats, but it's too late to check at the moment.  I'll try it 
out first thing tomorrow and update you guys, but for now, that's one 
drawback of using this fix.