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. -- View this message in context: http://n2.nabble.com/17-sec-recieve-a-bye-and-a-hangup-tp3774964p3774964.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
