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

Fri Jan 29 12:12:33 CST 2010

Just if it is helps someone, based on information at the blog:
http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.htmlI'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>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> 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
>>
>> _______________________________________________
>> -- 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/20100129/8f39457a/attachment.htm 