Please post your BOARD.INI file (configuration of AudioCodes).
Also, do you expect to do single-stage dialing (MP104 takes SIP invite information and turns that into DTMF output), or two-stage dialing (MP104 only answers and connects the audio/RTP path)? John Balogh, Sr. Systems Engineer PSU, ITS, TNS, Network Planning sip:[email protected] From: [email protected] [mailto:[email protected]] On Behalf Of Daniel - Asterisk Sent: Wednesday, December 02, 2009 12:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help configuring Audiocodes MP-104 FXO Hi list, I'm trying to get ready the MP-104 FXO to use qith my box, but when I send calls I hear only dial tone and after a few seconds I get busy signal. I very appreciate your advices. Command line results and SIPconfigurations follows: CLI> -- Executing [7991696...@total:1] Playback("SIP/101-09dd8918", "beep") in new stack -- <SIP/101-09dd8918> Playing 'beep' (language 'es') -- Executing [7991696...@total:4] Dial("SIP/101-09dd8918", "SIP/201/991696900") in new stack -- Called 201/991696900 -- SIP/201-09ddc890 answered SIP/101-09dd8918 sip.conf [201] secret = **** callerid = Mobile_01 <201> type = friend host = dynamic context = total dtmfmode=rfc2833 qualify = yes call-limit=5 disallow = all allow = gsm allow = ulaw allow = alaw allow = g729
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