I would also like to add my question: In the Asterisk 1.2 branch, the rfc2833 mode apparently doesn't send a duration of the dtmf when used. Is this fixed in Callweaver? ( It is fixed in the 1.4 branch )
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? > > > > 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 >
_______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
