Hi, I have a little problem related to quintum a400 gateway. I have installed asterisk-1.2.8. Have configured it with SIP and H323 channels to recieve and make calls over lan using softphone (shphone for both SIP and H323). H323 driver version is openh323-v1.17.1 and pwlib-v1.9.0. pc to pc calls thru asterisk are established without any problem. Recently i connected a quintum a400 gateway to the lan. quintum is programmed like whenever it recieves a call request, it should forward the request to asterisk server, and it does with no error. after call being forwarded to asterisk, asterisk uses 's' extensions to handle the call. initially i am using the following extension: exten=>s,1,Dial(H323/192.168.0.23,20) ;23 is the ip address of pc using softphone. a digital phone (simple one which we use for direct pstn comm) is connected with the 1st pbx port of quintum. we dial quintum extension, quintum(using h323) forwards the call to asterisk, asterisk dials the ip .23, softphone rings, as we answere the phone the call gets disconnected atuomatically. SIP account ends up with the same result. here is the log info: H323 LOG == Starting H323/ip$192.168.0.22:24602/21 at default,15,1 failed so falling back to exten 's' -- Executing Dial("H323/ip$192.168.0.22:24602/21", "H323/192.168.0.23/20") in new stack -- Called 192.168.0.23/20 Jul 4 16:17:23 WARNING[2955]: channel.c:2693 ast_channel_make_compatible: No path to translate from H323/192.168.0.23-2(-2033656) to H323/ip$192.168.0.22:24602/21(-2033656) -- H323/192.168.0.23-2 is ringing -- H323/192.168.0.23-2 is ringing -- H323/192.168.0.23-2 answered H323/ip$192.168.0.22:24602/21 == Spawn extension (default, s, 1) exited non-zero on 'H323/ip$192.168.0.22:24602/21' I ALSO DONT KNOW THE REASON WHAT THIS WARNING IS ABOUT SIP LOG the same thing happens for sip account but without the warning. H323 Channel Configuration [laptopAsus] type=friend host=192.168.0.23 context=default SIP Channel Configuration [Ammad] type=friend secret=tu qualify=4000 nat=yes host=dynamic canreinvite=no context=default I have no idea how to solve this problem. already tried to use different codecs but no progress......plz help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060704/e34061d5/attachment.htm