[asterisk-users] Help configuring Audiocodes MP-104 FXO

[John Balogh]

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:jdb at psu.edu

 

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 [7991696900 at total:1] Playback("SIP/101-09dd8918",
"beep") in new stack
    -- <SIP/101-09dd8918> Playing 'beep' (language 'es')
    -- Executing [7991696900 at 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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