I downloaded this two days ago from digium ftp. It reports to be 1.4.0-beta2.
I also have the g726nonstandard=yes on the sip.conf file. Now, do I need to modify or not the rtp.c that is on /main directory? I already checked, I don't have audio on g726, with both ports on g729, and with both ports on ilbc. I have audio with one port in g729 and the other on ulaw, or both on ulaw. I don't see any rtp traffic on the previous cases. Thanks, Carlos Alperin -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Wednesday, December 06, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: G.726 on Asterisk 1.4.0 Carlos Alperin wrote: > Ok, > > With everything restore on rtp.c, still I have no audio however the > call is not destroyed immediately as before. > > I'm going to put a second Granstream box, and findout if between two > boxes this happen too. > > I cannot believe that we cannot do 2 g726 on the same box at one time. > > Carlos > Make sure you are using the latest 1.4 branch, I already fixed a G726-32 related bug in there and you must have the g726nonstandard set to yes in sip.conf I do believe. -- Joshua Colp Software Developer Digium, Inc. _______________________________________________ --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