[OpenSIPS-Users] 17 sec, recieve a bye and a hangup

[Peter den Hartog]

I'm trying to intergrate opensips with a allready running Asterisk server.
The two servers are both on the same machine. 

I can recieve calls fine, Asterisk send them to my opensips installation,
and the opensips forwards the phone call to the right user. I can call
between the users on the network, with out any issue's so far so good.

I have a sip trunk registered on Asterisk, and i use that for my in and
outgoing calls.

But when i make an outside call, the call ends after 17 seconds. Looking at
the sip messages i see that i recieve a bye, then the call is gone.

Am i doing something wrong, should the sip trunk be directly in opensips?
and add that as a rewritehost? Or is this an Asterisk issue?

My opensips is running on port 5090 (so are the phones) and my
asterisk+outside trunk is on 5060.
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