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

[John Balogh]

> I want to do single-stage dialing. I've just realized I 

> have the two-stage running now (I get dial tone and then,

> when i introduce the number, the call get through).

 

Right. 

 

According to the SIP User's Manual

LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf

page 67/294

 

"

Enable Digit Delivery to Tel [EnableDigitDelivery]

 Disable [0] = Disabled (default).

 Enable [1] = Enable Digit Delivery feature for MediaPack/FXO & FXS.

The digit delivery feature enables sending of DTMF digits to the
gateway's port after the line is offhooked (FXS) or seized (FXO). For
IP->Tel calls, after the line is offhooked / seized, the MediaPack plays
the DTMF digits (of the called number) towards the phone line.

[...]

To use this feature with FXO gateways, configure the gateway to work in
one

stage dialing mode.

"

 

You probably need to set the above.

 

The FXO parameter (from page 107/294):

 

"

Dialing Mode [IsTwoStageDial]

 One Stage [0] = One-stage dialing.

 Two Stage [1] = Two-stage dialing (default).

Used for IP->FXO gateways calls.

 

If two-stage dialing is enabled, then the FXO gateway seizes one of the
PSTN/PBX lines without performing any dial, the remote user is connected
over IP to PSTN/PBX, and all further signaling (dialing and Call
Progress Tones) is performed directly with the PBX without the gateway's
intervention.

 

If one-stage dialing is enabled, then the FXO gateway seizes one of the
available lines (according to Channel Select Mode parameter), and dials
the destination phone number received in INVITE message. Use the
'Waiting For Dial Tone' parameter to specify whether the dialing should
come after detection of dial tone, or immediately after seizing of the
line.

"

 

So you probably need to clear that parameter (it is not configured in
your .INI file now, so you need to add it, or change the web interface
drop-down control).

 

Let us know if this helps.

 

JDB

 

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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