Hi,
I have setup a SIP trunk for outbound calls. When I dial a call everything works fine and remote phone rings. but as soon as someone picks up the phone on the remote side, he gets disconnected. I am trying to figure a solution for two weeks now, but no succuess 
. I would really appreciate if someone can help me out. I am fairly new to Asterisk.
When calling from an internal softphone (ext:1000), I have noticed this in debug
Any ideas whats wrong ???
DEBUG[12760] devicestate.c: Notification of state change to be queued on device/channel SIP/tktech-b7820c50
DEBUG[3550] devicestate.c: No provider found, checking channel drivers for SIP - tktech-b7820c50
DEBUG[12760] devicestate.c: Notification of state change to be queued on device/channel SIP/tktech
DEBUG[3550] chan_sip.c: Checking device state for peer tktech-b7820c50
VERBOSE[12760] logger.c: – SIP/tktech-b7820c50 answered SIP/1000-b7825318
WARNING[12760] channel.c: No path to translate from SIP/1000-b7825318(4) to SIP/tktech-b7820c50(256)
WARNING[12760] app_dial.c: Had to drop call because I couldn’t make SIP/1000-b7825318 compatible with SIP/tktech-b7820c50
DEBUG[12760] channel.c: Hanging up channel 'SIP/tktech-b7820c50’
DEBUG[12760] chan_sip.c: Hangup call SIP/tktech-b7820c50, SIP callid 0e594a7d50127cdc4130491b1587dfdc@10.1.1.4)
VERBOSE[12760] logger.c: Scheduling destruction of SIP dialog ‘0e594a7d50127cdc4130491b1587dfdc@10.1.1.4’ in 32000 ms (Method: INVITE)
Anyone ??
Incompatible codecs. Please use sip set debug on, or sip set debug ip, to find out exactly what the mismatch is. However, my guess is that one side only supports G.729 and the other doesn’t, as, in most other cases, Asterisk could transcode.
Thanks for your reply.