Hi, You can try to set faststart=no on ooh323.conf , i am not sure it will work but it works for me when i am geting calls from IAX2 to h.323 ( i had the same porblem before with IAX2 to h.323) ...
And plz post your problem on asterisk-users mailing list not in dev... thanks atik On 8/24/06, Juan Carlos Castro y Castro <[EMAIL PROTECTED]> wrote:
Greetings. Although this message may not initially look like a -dev worthy post, keep reading because I believe it is. I have the following problem: I have an Asterisk+GnuGK installation, in which the brand of H.323 ATAs we use (obscure OEM; I believe the maker is Aristel) are unable to receive calls. The phone rings, when I pick it up it's all mute and - that's the weird part - the originator keeps on hearing ring tones as if nobody had picked up. When the receiving (mute) end hangs up, the call is terminated. Calls in the other direction don't have this problem. It's not an obvious ATA bug, neither a GnuGK bug -- two ATAs can call each other just fine. More: Asterisk can call an H.323 softphone (SJPhone) fine too. The traffic from the latter looks mighty different from that from a call to an ATA. Here's links to Ethereal analyses of three cases: Asterisk to SJPhone (OK): http://img78.imageshack.us/img78/9452/asterisktosoftphoneka1.png Asterisk to ATA (Rings but doesn't complete): http://img78.imageshack.us/img78/8003/asterisktoatasq5.png ATA to ATA (OK): http://img168.imageshack.us/img168/1830/atatoouratajd6.png The big difference between Asterisk-to-ATA and ATA-to-ATA is, Asterisk (or, rather, chan_ooh323) never sends terminalCapabilitySet or masterSlaveDetermination packets. And, of course, there's no RTP coming from Asterisk. All captures were made at the gatekeeper. It has two IP addresses (has to because of Linux-HA). Switching to it having only one IP didn't help. Yes, I have the .cap files that originated those. I fear there will have to be some hacking in chan_ooh323 (hence me posting here) in order for it to speak properly with the ATAs like other ATAs do. If someone would like to test this, I can register my H.323 ATA with the gatekeeper you're using, and keep on making and receiving calls at your leisure. Plus chatting via MSN or the messenger of your choice, or IRC. If this wasn't -devvy enough, I'll post to -users instead. Should it also go to bugs.digium.com? Regards, Juan _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
