[asterisk-users] video calls not working
Paul Belanger
pabelanger at digium.com
Sat Dec 3 14:37:31 CST 2011
On 11-11-21 10:07 AM, Danny Nicholas wrote:
> Two items
>
> #1 you only need 1 disallow=all in your sip.conf definition
>
> #2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an xlite
> response to Asterisk starting music-on-hold during the connect pause. The r
> on the dial command attempts to do a "faux ring" which xlite interprets as a
> MOH request, so if you don't want to patch/recompile, just take the r off of
> Dial.
>
Why are you manually patching asterisk? Have you created an issue in
JIRA about this?
--
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org
More information about the asterisk-users
mailing list