Jonathan Miller
2006-Jun-16 13:10 UTC
[Asterisk-Users] no IVR audio but phone to phone fine
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 I'm having trouble getting my IVR to produce audio onto my cisco 7940 handset using SIP. I've got CentOS 4.3 running * 1.2.9.1 and the corresponding latest releases of libpri, zaptel, addons and sounds. I'm using two extensions to test that successfully dial each other and can talk back and forth fine. When I dial into the IVR, I get a normal looking output from the console: -- Executing Goto("SIP/2601-588e", "demo|s|1") in new stack -- Goto (demo,s,1) -- Executing Wait("SIP/2601-588e", "4") in new stack -- Executing Answer("SIP/2601-588e", "") in new stack -- Executing Set("SIP/2601-588e", "TIMEOUT(digit)=5") in new stack -- Digit timeout set to 5 - -- Executing Set("SIP/2601-588e", "TIMEOUT(response)=10") in new stack -- Response timeout set to 10 - -- Executing BackGround("SIP/2601-588e", "demo-congrats") in new stack -- Playing 'demo-congrats' (language 'en') All seems as if the output should be directed onto the SIP channel. That's not happening. I changed the default RTP ports to be the same as those on the phones, 16384 > 32778 but am still having trouble getting the audio to play on this machine. I suspect a package missing, but don't know what that could be and I'm not able to find much help on this in the history of the list. Other issues have been resolved, but I can't find anything about this. Please help! Sincerely, Jonathan Miller -----BEGIN PGP SIGNATURE----- Version: PGP Universal 2.0.6 iQEVAwUBRJMCZ5JhYmFK+jfsAQgAzAf+NlNbr4f9TAP83/atcmkImOG3fF8+jTs+ /1KqPWyEny53961vs4xNcY8j0fE0MdIaW9XxeiTZysUcQc1cQlflU7t590gEt7jh 20yhzNTvH77W2/p2dHnagTCg+CeBKI+3T0344W4m0d1HeFw/oGMyvmu8Er2i1RDx flWmz42TvHUuTwhNiLPiTStBIkBfr5lzpc+LlD8dIazHuZCsXHc64b0wPHvPm6f1 zYOTp5XwCP9B4cZQ+AFkxF345ZUJkyotkTXEOit86BloNoo61ms41fBDkqPGKmpk wWOFr6UHE3O8GMSX7/PENXfIQLs36I1N+w8H26V9iB4aqC26gAOGkQ==MBqK -----END PGP SIGNATURE-----