On Mon, 2006-02-20 at 17:04 +0100, Marc Patino Gómez wrote: > Hi, > > Can you post your working config, I'm wasting my time to config h323->sip >
Is working now :) I'm using asterisk-oh323 0.7.3 on my asterisk 1.2.4 box. I've to configure in oh323.conf with gatekeeper=DISABLED and the context of my sip clients. The H.323 device is configured to use the asterisk ip address as gateway. With this config I can use SIP/IAX2 trunks to call outside from the h.323 device and can call from SIP/IAX2 to H.323 and from H.323 to my SIP/IAX2 devices :) sip*CLI> oh323 show conf sip*CLI> Configuration of OpenH323 channel driver ------------------------------------------ Version: 0.7.3 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: ON/ON/ON Supported formats in pref. order: alaw<0> ulaw<1> gsm<2> g723<3> g729<4> Jitter buffer limits (min/max): 20-100 ms TCP port range: 10000 - 20000 UDP (RAS) port range: 10000 - 20000 UDP (RTP) port range: 10000 - 20000 IP Type-of-Service value: 0 User input mode: tone Max number of inbound H.323 calls: 100 Max number of outbound H.323 calls: 100 Max number of simultaneous H.323 calls: 100 Max call rate (ingress direction): 1.00/30 Default language: es Default music class: default Default context: from-internal sip*CLI> I've to create the h.323 extentions for the two ports of my H.323 device (ext 103 and 104 for port 1 and port 2) : [ext-local] include => ext-local-custom exten => 101,1,Macro(exten-vm,novm,101) exten => 101,hint,SIP/101 exten => 102,1,Macro(exten-vm,novm,102) exten => 102,hint,SIP/102 exten => 103,1,Macro(exten-vm,novm,103) exten => 103,hint,OH323/[EMAIL PROTECTED] exten => 104,1,Macro(exten-vm,novm,104) exten => 104,hint,OH323/[EMAIL PROTECTED] exten => 555,1,Macro(exten-vm,novm,555) exten => 555,hint,SIP/555 > > Thanks > > Guillermo Salas M wrote: > > >Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can > >make calls from one h.323 device to the world using sip trunks :) > > > >I can call to sip devices from the h.323 one. Now I want to make calls > >from sip to h.323 but it does not work. Maybe one of us have a > >configuration example to do this? > > > >I'm using the latest svn version (compiled yesterday). > > > >========================================================================= > >Connected to Asterisk SVN-branch-1.2-r10487 currently running on sip > >(pid = 29977) > >nip*CLI> > > > > > > > >Best regards, > > > > > > > > > > -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
