[Asterisk-Users] SIP problem picking up the call

[Moises Silva]

sip debug
rtp debug
enable all the log levels in console in logger.conf

regards

On 1/20/06, RumaTech <asterisk at rumatech.com> wrote:
> Hi, all
>
> I am trying to call to particular destination via SIPNET (one of the VoIP
> providers).
>
> I can succesfully dial and I can hear waiting tone, however nothing happens
> beyond it.
>
> Here is what asterisk shows:
>
>  Executing Dial("SIP/phone2-fa85", "SIP/sipnet/84959951017") in new stack
>     -- Called sipnet/84959951017
>     -- SIP/sipnet-f021 is ringing
>     -- SIP/sipnet-f021 answered SIP/phone2-fa85
>     -- Attempting native bridge of SIP/phone2-fa85 and SIP/sipnet-f021
>
> Basically, RTP is not being set up.
> It is not a codec problem.
>
> Any ideas where to look next?
> Thanks,
> Rudolf
>
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