Zen Kato
2004-Mar-03 15:53 UTC
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, I am trying to confirm the command 'canreinvite=yes' in sip.conf using grandstream BT101/2s and snom100s. In either case, no description nor 'canreinvite=yes', media stream always go through *. Do I need another settings for confirming sip clients directly communicate each other? -- Zen
Girish Gopinath
2004-Mar-03 17:56 UTC
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Zen,>I am trying to confirm the command 'canreinvite=yes' in sip.conf >using grandstream BT101/2s and snom100s. In either case, no description >nor 'canreinvite=yes', media stream always go through *. > >Do I need another settings for confirming sip clients directly >communicate each other?Do you have a Dial statement that has "t" or "T" in it? This will force the media stream to pass through Asterisk. Regards, Girish _________________________________________________________________ Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag Only on www.shaadi.com. Register now!
Girish Gopinath
2004-Mar-04 06:46 UTC
[Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi Zen,>From: Zen Kato <zenkato@pis.bekkoame.ne.jp><snip>>Does these "t" and "T" are used for transfer(blind/consaltation) from >called user and calling user, respectively? If we don't have these >'t' and 'T', can't we do transfer?'T' and 't' are used for transfer using # The 'T' allows the calling user to transfer the call. 't' allows the called user to transfer the call. Andy Powell's guide to Asterisk http://www.automated.it/guidetoasterisk.htm has these details, It is simple, and contains some good basic things about Asterisk. Regards, Girish>Regards, > >Zen > >"Girish Gopinath" <gopinath_girish@hotmail.com> wrote : > > > Zen, > > > > >I am trying to confirm the command 'canreinvite=yes' in sip.conf > > >using grandstream BT101/2s and snom100s. In either case, no description > > >nor 'canreinvite=yes', media stream always go through *. > > > > > >Do I need another settings for confirming sip clients directly > > >communicate each other? > > > > Do you have a Dial statement that has "t" or "T" in it? > > This will force the media stream to pass through Asterisk. > > > > Regards, Girish > > > > _________________________________________________________________ > > Contact brides & grooms FREE! http://www.shaadi.com/ptnr.php?ptnr=hmltag > > Only on www.shaadi.com. Register now! > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >_________________________________________________________________ Skin is in! Bollywood is sizzling. http://server1.msn.co.in/slideshow/striptease/ Check out these hot pics!
Maxime R
2004-Mar-04 08:28 UTC
[Asterisk-Users] 2 Linphones communicating through Asterisk?
Both are allowed but for readability => is used on objects. Maxime ----- Original Message ----- From: "Tor Houghton" <torh@bogus.net> To: <asterisk-users@lists.digium.com> Sent: Thursday, March 04, 2004 10:19 AM Subject: Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?> On Thu, Mar 04, 2004 at 02:06:52PM -0000, Jon Shamash wrote: > > > > [snip] > > > > it should be > > exten = 66,1,Dial(SIP/66) > > > > Incidentally, is there a difference between => and =, or are both allowed? > > Tor > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users