Hi, I've just now stated reading about CallWeaver project. My target is to join OpenSer with a SIP PBX and Asterisk makes it difficult to me because some issues. I'd like to know how Callweaver handles these issues so I list some questions. Thanks a lot for any explanation about them:
1) Sip spiral: Asterisk chan_sip doesn't allow SIP spiral (not a loop). This is: - Asterisk calls a user of a SIP proxy. - This SIP proxy has a forwarding for this user that correspond with a PSTN number, so the INVITE is URI modified and sent back to Asterisk (the PSTN gateway). - Asterisk receives the same INVITE it sent before (same fromt and to tags and call-id, but different URI so NOT the same INVITE) and rejects it with "482 Loop Detected". This is a pain since it could be a really cool feature that Asterisk makes impossible. There is a related bug and patch not accepted and updated to trunk version: http://bugs.digium.com/view.php?id=7403 Since Callweaver uses Sofia SIP I hope this stack understands "482" and accept SIP spiral. Does it? 2) Native transfer and direct RTP: Asterisk allows native transfer with options "t" and/or "T" in "Dial" command. This native transfer is done by DTMF and Asterisk remains in the media path in order to get those DTMF's even if "canreinvite=yes" for both callee and callee. But using "dtmfmode=info" the DTMF goes as SIP INFO messages (not in the RTP) so there is no reason Asterisk to remain into the media path. Anyway Asterisk remains in it :( Reported bug in Asterisk: http://bugs.digium.com/view.php?id=11172 (I reported it today and it seems fixed now !!!) 3) Multidomain support - Virtual hosts: Asterisk support for multidomain is really limited, just by asignig context to incoming calls based on the domain, no more. Is there more about it in CallWeaver? 4) Support for SIP Session Timers: Asterisk doesn't support it, so if a UAC crashes while being in-hold (not sending RTP) then Asterisk has no way to know it so the channel remains open (a pain for CDR). Does Callweaver support SIP Session Timers? http://www.faqs.org/rfcs/rfc4028.html 5) Support for outbound proxy in [general]: Asterisk doesn't allow using a outbound proxy for ALL outgoing calls, just for peers. Does CallWeaver allow it? 6) Big vulnerability with native transfer: Explained here: http://bugs.digium.com/view.php?id=10198 Does Callweaver fix it? Best regards. -- Iñaki Baz Castillo _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
