[asterisk-users] g729 registered

Ralph Liebessohn ralphliebessohn at gmail.com
Mon Nov 20 08:22:03 MST 2006


On 11/20/06, Alex Robar <alex.robar at gmail.com> wrote:
>
> Hi Ralph,
>
> Have you setup your PAP2 to allow the 729 codec? I believe you actually
> have to tell it that it's allowed to use that codec before it will work.
>
> Cheers,
> Alex
>
> On 11/20/06, Ralph Liebessohn <ralphliebessohn at gmail.com> wrote:
>
> > Hi guys,
> >
> > I've registered some g729 licenses, during register process everything
> > worked fine.
> >
> > astk2*CLI> show g729
> > 0/0 encoders/decoders of 20 licensed channels are currently in use
> >
> > But I'm not able to use this codec. I'm trying to use a linksys PAP2 to
> > talk using g729 but I got this answer from asterisk:
> >
> > Got SIP response 488 "Not Acceptable Here" back from 192.168.10.126
> >
> > And when I try to forward a call using IAX I got it:
> >
> >     -- Executing Dial("SIP/teste1-081b23c8",
> > "IAX2/192.168.5.12/6551|30|tT") in new stack
> >     -- Called 192.168.5.12/6551
> > Nov 20 01:29:48 WARNING[8297]: chan_iax2.c:7099 socket_read: Call
> > rejected by 192.168.5.12: Unable to negotiate codec
> >
> > Should asterisk translate to another codec when trying to make a new
> > call with iax? Why can't asterisk make a call using g729 and sip?
> >
> > Some configuration.
> >
> > SIP.CONF
> > [teste]
> > type=friend
> > secret=blah
> > host=dynamic
> > username=teste
> > context=from-spo
> > disallow=all
> > allow=g729
> > allow=ulaw
> >
> > [teste1]
> > type=friend
> > secret=blah
> > host=dynamic
> > username=teste1
> > context=from-spo
> > disallow=all
> > allow=g729
> > allow=ulaw
> >
> > EXTENSIONS.CONF
> > exten => 1,1,Dial(SIP/teste1)
> > exten => 0,1,Dial(SIP/teste)
> >
> > Thanks.
> >
> > --
> > Ralph Liebessohn
> > ICQ: 74835911
> > Skype: liebessohn
> >
>
> --
> Alex Robar
> alex.robar at gmail.com
>


Hi Alex,

I set on Audio configuration to enable g729a, g729a as preferred codec and
use only preferred codec. Is only that right?
With ulaw all calls work fine.

-- 
Ralph Liebessohn
ICQ: 74835911
Skype: liebessohn
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061120/2d4a7575/attachment.htm


More information about the asterisk-users mailing list