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