I compiled the channel on usr/src/asterisk/channels/h323, which I believe is the Nufone Channel. Previously I did compile the PWLIB and OH323 packets.
Is that correct ? Regards, Jorge A. -----Mensaje original----- De: Paul Mahler [mailto:[EMAIL PROTECTED] Enviado el: Sunday, November 21, 2004 10:56 PM Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper Are you using oh323 ? Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Jorge Alayon > Sent: Friday, November 19, 2004 4:33 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Asterisk and H.323 Gatekeeper > > Hello, > > I am new to this list and to asterisk and going through the > archive file I did not find an answer to my problem. > > I have a VoIP network working fine with multiple gateways > registered to a Cisco H.323 Gatekeeper. I have successfully > registered Asterisk as a GW in that network and also > successfully registered two X-Lite SIP Client to asterisk > that call to each other. > > I want to connect to the H.323 network but call does not > progress from the SIP to the H.323 network. > > This is the ASterisk console output. > > -- Registered SIP '1154538511' at 192.168.11.46 port 5060 > expires 1800 > -- Executing Wait("SIP/1154538511-ed8a", "2") in new stack > -- Executing Dial("SIP/1154538511-ed8a", > "h323/01145568423") in new stack > -- Called 01145568423 > == No one is available to answer at this time > -- Timeout on SIP/1154538511-ed8a > == CDR updated on SIP/1154538511-ed8a > -- Executing Goto("SIP/1154538511-ed8a", "#|1") in new stack > -- Goto (default,#,1) > -- Executing Playback("SIP/1154538511-ed8a", > "demo-thanks") in new stack > -- Playing 'demo-thanks' (language 'en') > -- Executing Hangup("SIP/1154538511-ed8a", "") in new stack > == Spawn extension (default, #, 2) exited non-zero on > 'SIP/1154538511-ed8a' > > If I dial from an ATA, An AS5300, or an Audiocodes GW the > number 01145568423 through the Gatekeeper, it works. > > Any ideas ? > > Regards, > > Jorge A. > _______________________________________________ > 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 _______________________________________________ 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 _______________________________________________ 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