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

[programmer_ted]

>>
>>>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?  :)
>>
>>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.
>
>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.
>
>...snipped non-relevant signature info etc...

Learning as I go.  It appears I don't have access to the capabilities value 
from the ast_channel structure.  I'm just gonna go ahead and have the 
SLINEAR data translate to the channel's writeformat.