[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?  :)