Hey there:
I'm having a problem with call forwarding.
If I list an external (PSTN) number, the caller gets an '...if you want
to try your call again, please hang up and dial again.' If I list an
internal extension, the caller gets the voicemail of the original
extension.
I'm stuck right now so
I'm running into a packet fragmentation problem on my OK response to an
INVITE from an ITSP.
It appears that the OK response is just over the limit and is getting
split into two IP packets. If they
arrive at the ITSP out of order (which they do about 50% of the time)
then the call fails. A
It isn't available. G.711ulaw and alaw for the media server.
Of course if your Phones and gateways support g.729 you could set them
to prefer those codecs and outbound calls would use G.729. However you
still need to leave g.711u/a for the media server (auto attendant, vm).
Mike
> -Origina
how can I use G729 codec, do I need a licenses?
___
sipx-users mailing list
sipx-users@list.sipfoundry.org
List Archive: http://list.sipfoundry.org/archive/sipx-users
Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users
