Egbert Groot
2008-Dec-29 23:48 UTC
[asterisk-users] Bug in contact header from Asterisk 1.6.0.3-rc1 ?
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:12329123 at 123.123.123.123> 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:12329123 at 123.123.123.123'. 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 ?nd port. Can someone acknowledge my observations and conclusion is right? thanks, Egbert Groot.
Egbert
2009-Jan-01 22:40 UTC
[asterisk-users] Bug in contact header from Asterisk 1.6.0.3-rc1 ?
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:12329123 at 123.123.123.123> 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:12329123 at 123.123.123.123'. 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 ?nd port. Can someone acknowledge my observations and conclusion is right? thanks, Egbert Groot.