Hi all, I'm not sure wether it is a bug or not, so I'm asking for your opinion before submitting it to the bugtracker.
The problem: I use asterisk with in sip.conf a non standard bind port of 5070 set. Now when asterisk sends out an Invite message to my sip proxy, the contact header in de request is something like: Contact: <sip:[email protected]> The call succeeds and gets answered. So far so good. By using the 'Via' headers the 200 OK repsonse gets properly routed to asterisk. But now the client wants to end the call, and sends 'BYE sip:[email protected]'. Now the proxy can't route the messages by means of the Via header (because this is a new transaction? and Asterisk didn't insert a record-route header). The proxy forwards the 'Bye' to the default sip port on '123.123.123.123', with no success. The other way round, when the client initiates the call, asterisk answers with a '200 OK'. This response includes a correct 'Contact' header, consisting of both username,domain/ip ánd port. Can someone acknowledge my observations and conclusion is right? thanks, Egbert Groot. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
