[Asterisk-Users] H323 call dropped when answered

[Krystian.Filiks]

aaaaaaaaaaaaaaa............
Thanks for that.
That clears things up in my head a little.

To change to oh323 do I have to recompile the * CVS sources without the 
chan_h323 or can I just install oh323 abnd remove the chan_h323 from the 
module directory?

What was the problem with oh323, was it just a config problem, can you 
give me a pointer hov to configure it?
Did g711 and g732 finaly work with oh323?

how did you get the "Executing Dial("SIP/sj1-4ff7", "H323/h") in new 
stack" to go away?

I need all the help I can get, this is my first asterisk.

Newb. warning :-)

Thanks
Krystian

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.
>
>  
>
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