[asterisk-users] Help configuring Audiocodes MP-104 FXO
Daniel - Asterisk
earohuanca at gmail.com
Fri Jan 29 14:02:58 CST 2010
It was a pending draft I forgot to send.. sorry.
On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins <mcollins at ccdservice.net>wrote:
> 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
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by
> > http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100129/19bb4bf9/attachment-0001.htm
More information about the asterisk-users
mailing list