[asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11

[Álvaro Palma]

I upgraded to 1.2.11 and now I see two behaviors different than the
existent in 1.2.10:

1.- I get 183 Session Progress instead of 180 Ringing.
2.- If I have three extensions, A, B and C. A using codec X, B using
also codec X and C using codec Y. If C dials to B and A tries to pick
up the call (using *8#), it start getting an endless output of:

chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native
formats is 256 (read/write = 64/64)

(in this case, C was using GSM, B and A, G729).

I tried this making all the combinations between A, B and C calling each
other, and I only get the problem when the picked conversation needs to
be transcoded (it means, if A calls to C and B pick it up, it worked
fine). For some reason, I guess somebody initializes a variable as
SLINEAR (64) in all cases. The result is that it's impossible to pick up
the calls!!!

Has anybody experienced this issue? Is this a bug in 1.2.11? I looked
through Mantis, but didn't find a clue.

Thanks a lot for your attention.

-- 
Atly.
Alvaro Palma