[asterisk-dev] Help with 240 samples on frames read from chan_iax
moises.silva at gmail.com
Sat Nov 4 12:20:53 MST 2006
Hi everyone. I know app_conference is not formal part from Asterisk,
but I think you may help me a bit to understand what is happening
app_conference is working just fine, except when receives frames
with 240 samples, it seems is somehow hardcoded to expect frames of
160 samples, so when receives 240, a buffer overflow ocurrs on the mix
buffer and crashes Asterisk. Frames with 240 samples, so far, are just
generated by my IAX2 connection with other server, so, if I enter a
conference with a SIP channel using ULAW, ZAP channel using SLIN, and
IAX2 channel (kiax softphone) with ULAW, everything works fine, but at
the moment the other IAX2 server enters the conference (ULAW also),
I can stop asterisk from crashing modifing the app_conference code
to NOT mix 240 samples frames, and works, but obviously the voice from
the IAX2 server is not received by the other parties.
Can somebody just give me a hint about where to look?
Why just that IAX2 server generates frames with 240 samples?
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org"
More information about the asterisk-dev