[Asterisk-Users] SIP call disconnected after answer

[William Piper]

Sounds like there maybe a codec issue. If you are using g729, make sure you
have licenses.

bp


On 6/14/06, Mimmus <dviggiani at tiscali.it> wrote:
>
> 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
>
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