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.
