[Asterisk-Users] Re: rfc3389 and ATA config: Fixed
Dan Fernandez
danfernandez00 at hotmail.com
Wed May 7 16:45:18 MST 2003
I had to disable G711 silence supression by changing bit 0 from 1 to 0 of the AudioCode variable.
AudioCode changed from 0x00150015 to 0x00140014
----- Original Message -----
From: Dan Fernandez
To: asterisk-users at lists.digium.com
Sent: Wednesday, May 07, 2003 8:05 PM
Subject: rfc3389 and ATA config
I use g723.1 for my SIP to SIP calls. To place calls to * i do a SIP_CODEC=ulaw.
I have two clients, and ATA (v2.16) and MSN.
The problem I have is with the ATA. When I dial an * app (voicemail for example) from the ATA , I keep getting the following NOTICE:
rtp.c (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible. The phone call drops after a few seconds.
I don´t get the problem with MSN.
Does anyone know what configuration do I need to change on the ATA. I have looked at the rfc3389 and the ATA admin guide and couldn´t figure it out.
Thanks
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030507/dd8be4cd/attachment.htm
More information about the asterisk-users
mailing list