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

[programmer_ted]

At 01:32 AM 6/30/2004, you wrote:
>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?  :)
>>
>>_______________________________________________
>>Asterisk-Dev mailing list
>>Asterisk-Dev at lists.digium.com
>>http://lists.digium.com/mailman/listinfo/asterisk-dev
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>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.
>_______________________________________________
>Asterisk-Dev mailing list
>Asterisk-Dev at lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-dev
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev

I thought it over a little bit more and the optimum solution would be to 
just translate the SLINEAR data to a format that is recognized by whoever 
is receiving the data, thus eliminating all drawbacks.  I'm going to try 
using capabilities rather than nativeformats as a quick workaround (after 
debugging to see if it'll work), and then work on adding the translating 
code to sip_write.  Actually, thinking about it again, it'd probably be 
best to just translate at the playtones_generator function.  I'll keep you 
guys updated.