On 11/6/07, Iñaki Baz Castillo <[EMAIL PROTECTED]> wrote: > 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?
Why don't you send Asterisk instead a 3xx message say a 302 moved temporarily? > 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 !!!) I don't like native transfers and anyways I disagree with your logic. > > 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? > Do your multi-domain support in SER > 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? I was not aware, please explain. > > 6) Big vulnerability with native transfer: > Explained here: > http://bugs.digium.com/view.php?id=10198 > Native transfer is a hack. It behaves like a hack, doesn't it? _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
