[asterisk-users] Help configuring Audiocodes MP-104 FXO
Matt Collins
mcollins at ccdservice.net
Fri Jan 29 12:23:09 CST 2010
Damn, where were you 6 months ago? ;)
Daniel - Asterisk wrote:
> Just if it is helps someone, based on information at the blog:
> http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html
> I've summarized the following steps:
>
> *Step 1:*
> Configure audiocodes to have registration account with asterisk, this
> can be done easily with "Protocol Management -> Protocol Definition ->
> Proxy&Registration", fill on "Proxy IP Address", "Enable Registration
> : Yes", "Username", "Password", and "Authentication Mode : Per Endpoint".
>
> *Step 2:*
> Configuring "Protocol Management -> Endpoint Phone Number", this is
> important part for make each FXO port on audiocodes registered with
> asterisk, in here, under "Channel", you can fill with either 1, 1-2,
> 1-8, 3-4, or whatever you want to have, this means that port 1, or
> port 1-2, etc will registered on astersik with userid/username filled
> on "Phone Number", yes, that is correct, "Phone Number" on this
> configuration page is AlphaNumeric, the password is using global
> "Password" on First step.
>
> next, on same page configure "Hunt Group ID", this is another
> important configuration which make audiocodes forward incoming call
> from asterisk to any available FXO. Hunt Group ID is number from 0 to
> any, I put 1.
>
> *Step 3:*
> to make audiocodes forward call from FXO to asterisk, configure
> "Endpoint Settings -> Automatic Dialing", I have 777 number on
> asterisk to handle all incoming call, so I put "Destination Phone
> Number" as 777 so every incoming call on FXO will be forwarded to 777
> on my Astersik.
>
> *Step 4:*
> this is the last configuration that everyone need, forward call from
> asterisk to any available FXO. in "Routing Tables -> IP to Hunt Group
> Routing Table" configure under "Dest. Phone Prefix" with "*" (or any
> prefix that you might have), "Source Phone Prefix" with "*", "Source
> IP Address" with "*", "Hunt Group ID" with any number you configure on
> Step 2, in my case, 1.
>
> /I add here addiiotnal steps needed for me to get ready/*:
> Step 5:*
> Add port by port authentication at Protocol Management -> Endoint
> Settings -> Authentication
>
> *Step 6:*
> Choosing Channel Selection Mode: Protocol Management -> Hunt Group
> Settings, choose the hunt group number and the way you prefer.
>
> *Step 7:*
> Choosing Dialing Mode: Protocol Management -> FXO Settings, I select
> One Stage.
>
> Hope it helps.
>
> Elder Daniel
>
>
>
> On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk
> <earohuanca at gmail.com <mailto:earohuanca at gmail.com>> wrote:
>
> 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
> <mailto: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>
> [mailto: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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