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 --- "Krystian.Filiks" <[EMAIL PROTECTED]> wrote:Hi, This is the scenario I have the SJlabs phone with g711ulaw active and the rest disabled. I have * with chan_h323 I have a Quintum DX that supports, g723.1 , g729AB, ulaw and alaw. The problem is that, it does not mather what I put in the extensions.conf I have tried all possible ways that I so far could find using the net. I tried all possible codecs ulaw, alaw, g723 and g729 always the same result. The phone rings but as soon as answered it dissconnects.__________________________________ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! 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
