Hi, this is my fist post.
–UPDATE: sorry I inverted the scenarios, please red them again----
I’m working on Asterisk and OCS2007 (Office Communication Server) integration. I have installed Asterisk 1.6 and I have created a sip trunk to the OCS Mediation Server (MS).
First scenario: I can place an audio call from an Xlite VoIP Phone registered on Asterisk to a MOC (Microsoft Office Communicator) client registerd on OCS2007.
Second scenario: I can also place a phone call from a MOC client to an Xlite client. No audio on MOC.
In both scenarios I have an half bridging with one of the two RTP connection passing from Asterisk. I can not understand why.
About SIP and RTP:
First scenario: Xlite calls MOC.
A re- invite is sent from Asterisk to Xlite. No re-INVITE is sent from Asterisk to Mediation Server (MS). The RTP from Xlite is directed to the MS and the RTP from the MS is directed to Asterisk and then Asterisk sends the RTP to Xlite.
Can anyone help me to understand why?
Second scenario: MOC calls Xlite.
A re-invite is sent to MS. A re-invite is sent to Xlite but this re-invite “redirects” the flow again to Asterisk (not to the MS). The RTP from the MS is directed to Xlite. The RTP from Xlite is directed to Asterisk that doesn’t send it to the MS.
Why the re-Invite to Xlite is wrong? (It should redirect to the MS instead it redirects to Asterisk again)
Why Asterisk doesn’t send the RTP to the MS?
Another bit for the second scenario: when the MS starts the RTP flow it uses Destination port: 0 (0). I know this was a problem for old version of Asterisk. Is still a problem?
Thanks a lot for any help.
Best regards. Fabrizio
