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

[Daniel - Asterisk]

I've set at Protocol Management >> FXO Settings >> Dialing Mode ==> One
Stage and everything is fine now

Thank you very much John,

EDA

On Wed, Dec 2, 2009 at 1:43 PM, John Balogh <JDB at psu.edu> wrote:

>  > 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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