Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
Hi, Thank you for the information. There are ts in Dial command in extensions.conf. When I deleted these ts, each sip phones were directly communicating. I just wrote these ts from the examples. 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? Regards, Zen Girish Gopinath [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] canreinvite=yes in sip.conf still go through asterisk
t and T are for # transfers. Other types of transfer are done in other ways. Zap FLASH transfers are set in /etc/asterisk/zapata.conf. I don't know how you enable/disable SIP or other types of transfers. On Thu, 2004-03-04 at 06:51, Zen Kato wrote: Hi, Thank you for the information. There are ts in Dial command in extensions.conf. When I deleted these ts, each sip phones were directly communicating. I just wrote these ts from the examples. 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? Regards, Zen Girish Gopinath [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- For Asterisk PBX related documentation go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section also see http://www.voip-info.org/wiki-Asterisk also see my site at http://www.fnords.org/~eric/asterisk/ BTEL Consulting ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [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! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
