[Asterisk-Users] Problem with SIP register

[Baris Simsek]

Diego Andrés Asenjo González wrote:

>Hi!
>
>I'm registering an asterisk server in a Sysmaster with a SIP account.
>The registration succeeds and I can establish a call that come from the
>Sysmaster.
>
>After around 80 seconds the Sysmaster sends a BYE SIP message and the
>call hang up. This does not occur to the hard/soft SIP phones registered
>in the sysmaster.
>
>I debug, but the only info that I can get is the BYE message.
>
>Thanks for your suggetions soving the problem.
>
>Bye.
>  
>
Hi,

Enable SIP debug and check which peer sends BYE at first.

After call establishment, can you hear voice for 80 sec.? What about RTP 
in this duration?

-- 
Baris Simsek
http://www.enderunix.org/simsek/