--- "Krystian.Filiks" <[EMAIL PROTECTED]> wrote:
> did you ever get the chan_h323 working?
No
> Asterisk . wrote:
>
> >Hello,
> >
> >I posted mails regarding the same issue just 1 week back (Sub: H323 Call Dropping,
> >and then
> OH323
> >Call Dropping). I was trying to connect from my CISCO ATA to Nextone using Asterisk
> >and the
> calls
> >were dropping immediately after the calls were answered. I used chan_h323 for about
> >4 days, but
> >could not make it. Then i changed to chan_oh323 and finally got it working after
> >trying that
> for
> >another 3 days using g729 codec. I also had issues with g711, and g723. I think
> >your problem is
> >codec. Try SIP debug also and see the packets. If nothing works, try using
> >chan_oh323.
> >
> >
> >
> >>"Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack"
> >>
> >>
> >
> >This one is really frustrating. I had no clue when it happened to me, and i had no
> >hangup
> command
> >in my dialplan.
> >
> >Good Luck!
> >
> >Girish
> >
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - 100MB free storage!
http://promotions.yahoo.com/new_mail
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
