Ok, thanks, also if i do not have rtp debug (i'm using asterisk 1.0.9) Hi
-----Messaggio originale----- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Joshua Colp Inviato: venerdì 28 luglio 2006 12.54 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] Canreinvite ----- Original Message ----- From: Giordano Grandis [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Fri, 28 Jul 2006 07:01:08 -0300 Subject: [asterisk-users] Canreinvite > How can I check if SIP re-invite is really working ? If you do a sip debug you should see two INVITEs to each side after the call is established with the IP address of the GXP2000 in the SDP. You can also run rtp debug to see if the RTP audio stream is running through Asterisk. > I'm trying it with two grandstream gxp2000. > > Thanks > Joshua Colp Digium _______________________________________________ --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 _______________________________________________ --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
