Re: [asterisk-users] How to make RTP does not go thru asterisk server
correcting an error cause by my own ambiguity: Mojo with Horan Company, LLC wrote: clarifying that you CANNOT put t or T in there if you want canreinvite=no to have no effect. you cannot put t or T in there if you want canreinvite=no to have ANY effect. If you want the stream to skip asterisk, and first you've told it not to allow reinvites with this canreinvite option, then you have to make sure asterisk isn't also being TOLD to listen in on the stream for transfer requests (t and T) Moj Anuj Jain wrote: Hi All I am using trixbox asterisk 1.2 I have enabled canreinvite=yes and no tT in the dialplan as it has been described in the various forums. Still the voice call goes thru the asterisk server. How can i really make the call between 2 grandstream devices( i am using HT 488, HT286 and SIP extensions) after the initial handshake. Thanks Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,4524455a101385315134984! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make RTP does not go thru asterisk server
Hi AllI am using trixbox asterisk 1.2I have enabled canreinvite=yes and no tT in the dialplan as it has been described in the various forums.Still the voice call goes thru the asterisk server. How can i really make the call between 2 grandstream devices( i am using HT 488, HT286 and SIP extensions) after the initial handshake.Thanks Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to make RTP does not go thru asterisk server
Hi AllI am using trixbox asterisk 1.2I have enabled canreinvite=yes and no tT in the dialplan as it has been described in the various forums.Still the voice call goes thru the asterisk server. How can i really make the call between 2 grandstream devices( i am using HT 488, HT286 and SIP extensions) after the initial handshake.My asterisk setup.Asterisk server is in one network and my 2 grandstreams are in 2 different networks and behind the firewall and natting enabled, all of them are connected thru the internet.Thanks Regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to make RTP does not go thru asterisk server
I'm not clear on your usage of t and T in the dial command, so clarifying that you CANNOT put t or T in there if you want canreinvite=no to have no effect. Anuj Jain wrote: Hi All I am using trixbox asterisk 1.2 I have enabled canreinvite=yes and no tT in the dialplan as it has been described in the various forums. Still the voice call goes thru the asterisk server. How can i really make the call between 2 grandstream devices( i am using HT 488, HT286 and SIP extensions) after the initial handshake. Thanks Regards !DSPAM:500,4524455a101385315134984! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:500,4524455a101385315134984! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
