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