Hi Jorge, The oh323 channel and h323 channel by NuFone are different. As far as your problem, this looks like a codec problem i had. Try to look that way.
K. ----- Original Message ----- From: "Jorge Alayon" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, November 22, 2004 11:06 AM Subject: RE: [Asterisk-Users] Asterisk and H.323 Gatekeeper > 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 _______________________________________________ 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
