[Asterisk-Users] G729 to G711 bridge
Miranda Gomez Miguel Angel
mmiranda at americatel.com.sv
Tue Oct 12 16:55:06 MST 2004
Excuse me, but can you explain a little more, what is the problem? so i can
search for the solutions,
regards
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Brian West
Sent: Tuesday, October 12, 2004 5:43 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] G729 to G711 bridge
> Called 2001
> -- SIP/2001-0a50 is ringing
> -- SIP/2001-0a50 answered SIP/2008-24fc
> -- Attempting native bridge of SIP/2008-24fc and SIP/2001-0a50
> Oct 12 17:15:59 WARNING[76823]: rtp.c:1392 ast_rtp_bridge: codec0 = 277 is
> not codec1 = 8, cannot native bridge.
> == Spawn extension (from-sip, 2001, 1) exited non-zero on 'SIP/2008-
> 24fc'
>
>
> the g729 codecs is showed as number 256 and not 277 in show codecs, what
> can be wrong?
That's because the codec is a bitmask.
bkw
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list