Álvaro Palma
2006-Aug-23 16:25 UTC
[asterisk-users] Getting strange behavior on SIP channels after upgrade to 1.2.11
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