[Asterisk-Users] Satellite, SIP Invite 488 Codec Rejection, SIP Timing Issue??

[Jason p]

Trust me this is on the ATA. set both lines to use 729 but dont fource them
to only use that codec (in the ata config) I spent days trying to figure
this out the first time i ran accross it, and after that config change on
the ata i haven't had problems. I have seen this on most of the sipura's
that are in the linksys routers.

Jason

On 11/23/05, Kevin P. Fleming <kpfleming at digium.com> wrote:
>
> Aaron Clauson wrote:
>
> > m=audio 16450 RTP/AVP 0 2 4 8 18 96 97 98 100 101
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:4 G723/8000
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:18 G729a/8000
> > a=rtpmap:96 G726-40/8000
> > a=rtpmap:97 G726-24/8000
> > a=rtpmap:98 G726-16/8000
> > a=rtpmap:100 NSE/8000
>
> I don't know what this (NSE) is, but Asterisk certainly doesn't support
> it.
>
> The only way we can debug this is by getting a complete 'sip debug' and
> 'set verbose' console trace; read the bug posting guidelines at
> bugs.digium.com and open a bug there with the required information.
> _______________________________________________
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> 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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051123/439bc9a0/attachment.htm