Paweł Gołaszewski
2003-Sep-04 04:31 UTC
[Asterisk-Users] oh323 <-> sip communication problem
I've got problem with connections h323 -> sip and sip -> h323. I've Cisco 7940 phone with sip soft and Netmeeting as h323 node. As gatekeeper I've gnugk and brand new asterisk from cvs + chan_oh323 0.5.5 When I call from Cisco (SIP) to h323 node by alias registered on gatekeeper and h323 node will answer the phone... I have on my Cisco still Ringing. Call termination, no matter from which side works fine. koga*CLI> oh323 show info koga*CLI> ^^^^^^^^^^^^^^^^^^^^^^^ why is here empty line? :) Information about active OpenH323 channel(s) -------------------------------------------- Num. Token State Init RX/TX Format Remote RTP Addr. Local RTP Addr. 0 ip$localhost/21538 RING Local 0/160 NULL 0.0.0.0:0 0.0.0.0:0 koga*CLI> show channels Channel (Context Extension Pri ) State Appl. Data H323:21538 (voip-h323 s 1 ) Ringing AppDial (Outgoing Line) SIP/blue-ebfa (default marosin 2 ) Ring Dial OH323/marosin I can't call from h323 phone to sip. I get that user is not registered on gatekeeper... My configuration: oh323.conf: [register] alias=asterisk alias=123 alias=blue alias=blues ;alias=marosin extensions.conf: [voip-h323] exten => marosin,1,Ringing exten => marosin,2,Dial,OH323/marosin exten => marosin,3,Hangup marosin is h323 phone blues and blue are sip phones. -- pozdr. Pawe? Go?aszewski --------------------------------- worth to see: againsttcpa.com CPU not found - software emulation...