[Asterisk-Users] SIP call disconnected after answer

[Mimmus]

Hi,
calling a partner on the other side of a SIP trunk, call gets disconnected
immediately after answer. This is the content of log file:

Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel:
SIP/cerved-out-6eba
Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels
SIP/232-2e41 and SIP/cerved-out-6eba
Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up channel
'SIP/cerved-out-6eba'
Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Hangup call SIP/cerved-out-6eba,
SIP callid 362258b02bbafa8117eecbb6755837a0 at 10.97.1.254)
Jun 14 16:25:14 DEBUG[14380] chan_sip.c: update_call_counter(9704) -
decrement call limit counter
Jun 14 16:25:14 DEBUG[14380] chan_sip.c: Updating call counter for outgoing
call
Jun 14 16:25:14 DEBUG[14380] app_dial.c: Exiting with DIALSTATUS=ANSWER.

I have Asterisk 1.2.8 but remote server has 1.2.4.

Any help?
-- 
Domenico Viggiani