It was a pending draft I forgot to send.. sorry.<br><br><div class="gmail_quote">On Fri, Jan 29, 2010 at 1:23 PM, Matt Collins <span dir="ltr"><<a href="mailto:mcollins@ccdservice.net">mcollins@ccdservice.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Damn, where were you 6 months ago? ;)<br>
<div><div></div><div class="h5"><br>
Daniel - Asterisk wrote:<br>
> Just if it is helps someone, based on information at the blog:<br>
> <a href="http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html" target="_blank">http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html</a><br>
> I've summarized the following steps:<br>
><br>
> *Step 1:*<br>
> Configure audiocodes to have registration account with asterisk, this<br>
> can be done easily with "Protocol Management -> Protocol Definition -><br>
> Proxy&Registration", fill on "Proxy IP Address", "Enable Registration<br>
> : Yes", "Username", "Password", and "Authentication Mode : Per Endpoint".<br>
><br>
> *Step 2:*<br>
> Configuring "Protocol Management -> Endpoint Phone Number", this is<br>
> important part for make each FXO port on audiocodes registered with<br>
> asterisk, in here, under "Channel", you can fill with either 1, 1-2,<br>
> 1-8, 3-4, or whatever you want to have, this means that port 1, or<br>
> port 1-2, etc will registered on astersik with userid/username filled<br>
> on "Phone Number", yes, that is correct, "Phone Number" on this<br>
> configuration page is AlphaNumeric, the password is using global<br>
> "Password" on First step.<br>
><br>
> next, on same page configure "Hunt Group ID", this is another<br>
> important configuration which make audiocodes forward incoming call<br>
> from asterisk to any available FXO. Hunt Group ID is number from 0 to<br>
> any, I put 1.<br>
><br>
> *Step 3:*<br>
> to make audiocodes forward call from FXO to asterisk, configure<br>
> "Endpoint Settings -> Automatic Dialing", I have 777 number on<br>
> asterisk to handle all incoming call, so I put "Destination Phone<br>
> Number" as 777 so every incoming call on FXO will be forwarded to 777<br>
> on my Astersik.<br>
><br>
> *Step 4:*<br>
> this is the last configuration that everyone need, forward call from<br>
> asterisk to any available FXO. in "Routing Tables -> IP to Hunt Group<br>
> Routing Table" configure under "Dest. Phone Prefix" with "*" (or any<br>
> prefix that you might have), "Source Phone Prefix" with "*", "Source<br>
> IP Address" with "*", "Hunt Group ID" with any number you configure on<br>
> Step 2, in my case, 1.<br>
><br>
> /I add here addiiotnal steps needed for me to get ready/*:<br>
> Step 5:*<br>
> Add port by port authentication at Protocol Management -> Endoint<br>
> Settings -> Authentication<br>
><br>
> *Step 6:*<br>
> Choosing Channel Selection Mode: Protocol Management -> Hunt Group<br>
> Settings, choose the hunt group number and the way you prefer.<br>
><br>
> *Step 7:*<br>
> Choosing Dialing Mode: Protocol Management -> FXO Settings, I select<br>
> One Stage.<br>
><br>
> Hope it helps.<br>
><br>
> Elder Daniel<br>
><br>
><br>
><br>
> On Wed, Dec 2, 2009 at 2:08 PM, Daniel - Asterisk<br>
</div></div><div class="im">> <<a href="mailto:earohuanca@gmail.com">earohuanca@gmail.com</a> <mailto:<a href="mailto:earohuanca@gmail.com">earohuanca@gmail.com</a>>> wrote:<br>
><br>
> I've set at Protocol Management >> FXO Settings >> Dialing Mode<br>
> ==> One Stage and everything is fine now<br>
><br>
> Thank you very much John,<br>
><br>
> EDA<br>
><br>
> On Wed, Dec 2, 2009 at 1:43 PM, John Balogh <<a href="mailto:JDB@psu.edu">JDB@psu.edu</a><br>
</div><div><div></div><div class="h5">> <mailto:<a href="mailto:JDB@psu.edu">JDB@psu.edu</a>>> wrote:<br>
><br>
> > I want to do single-stage dialing. I've just realized I<br>
><br>
> > have the two-stage running now (I get dial tone and then,<br>
><br>
> > when i introduce the number, the call get through).<br>
><br>
><br>
><br>
> Right.<br>
><br>
><br>
><br>
> According to the SIP User's Manual<br>
><br>
> LTRT-65405 MediaPack SIP User's Manual Ver 4.6.pdf<br>
><br>
> page 67/294<br>
><br>
><br>
><br>
> "<br>
><br>
> Enable Digit Delivery to Tel [EnableDigitDelivery]<br>
><br>
> Disable [0] = Disabled (default).<br>
><br>
> Enable [1] = Enable Digit Delivery feature for MediaPack/FXO<br>
> & FXS.<br>
><br>
> The digit delivery feature enables sending of DTMF digits to<br>
> the gateway’s port after the line is offhooked (FXS) or seized<br>
> (FXO). For IP->Tel calls, after the line is offhooked /<br>
> seized, the MediaPack plays the DTMF digits (of the called<br>
> number) towards the phone line.<br>
><br>
> [...]<br>
><br>
> To use this feature with FXO gateways, configure the gateway<br>
> to work in one<br>
><br>
> stage dialing mode.<br>
><br>
> "<br>
><br>
><br>
><br>
> You probably need to set the above.<br>
><br>
><br>
><br>
> The FXO parameter (from page 107/294):<br>
><br>
><br>
><br>
> "<br>
><br>
> Dialing Mode [IsTwoStageDial]<br>
><br>
> One Stage [0] = One-stage dialing.<br>
><br>
> Two Stage [1] = Two-stage dialing (default).<br>
><br>
> Used for IP->FXO gateways calls.<br>
><br>
><br>
><br>
> If two-stage dialing is enabled, then the FXO gateway seizes<br>
> one of the PSTN/PBX lines without performing any dial, the<br>
> remote user is connected over IP to PSTN/PBX, and all further<br>
> signaling (dialing and Call Progress Tones) is performed<br>
> directly with the PBX without the gateway’s intervention.<br>
><br>
><br>
><br>
> If one-stage dialing is enabled, then the FXO gateway seizes<br>
> one of the available lines (according to Channel Select Mode<br>
> parameter), and dials the destination phone number received in<br>
> INVITE message. Use the ‘Waiting For Dial Tone’ parameter to<br>
> specify whether the dialing should come after detection of<br>
> dial tone, or immediately after seizing of the line.<br>
><br>
> "<br>
><br>
><br>
><br>
> So you probably need to clear that parameter (it is not<br>
> configured in your .INI file now, so you need to add it, or<br>
> change the web interface drop-down control).<br>
><br>
><br>
><br>
> Let us know if this helps.<br>
><br>
><br>
><br>
> JDB<br>
><br>
><br>
><br>
> *From:* <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>><br>
> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
> <mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>>] *On Behalf<br>
> Of *Daniel - Asterisk<br>
><br>
> *Sent:* Wednesday, December 02, 2009 12:33 PM<br>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion<br>
> *Subject:* [asterisk-users] Help configuring Audiocodes MP-104 FXO<br>
><br>
><br>
><br>
> Hi list,<br>
><br>
><br>
><br>
> I'm trying to get ready the MP-104 FXO to use qith my box, but<br>
> when I send calls I hear only dial tone and after a few<br>
> seconds I get busy signal.<br>
><br>
> I very appreciate your advices.<br>
><br>
> Command line results and SIPconfigurations follows:<br>
><br>
> *CLI>*<br>
> -- Executing [7991696900@total:1]<br>
> Playback("SIP/101-09dd8918", "beep") in new stack<br>
> -- <SIP/101-09dd8918> Playing 'beep' (language 'es')<br>
> -- Executing [7991696900@total:4] Dial("SIP/101-09dd8918",<br>
> "SIP/201/991696900") in new stack<br>
> -- Called 201/991696900<br>
> -- SIP/201-09ddc890 answered SIP/101-09dd8918<br>
><br>
><br>
> *sip.conf*<br>
> [201]<br>
> secret = ****<br>
> callerid = Mobile_01 <201><br>
> type = friend<br>
> host = dynamic<br>
> context = total<br>
> dtmfmode=rfc2833<br>
> qualify = yes<br>
> call-limit=5<br>
> disallow = all<br>
> allow = gsm<br>
> allow = ulaw<br>
> allow = alaw<br>
> allow = g729<br>
><br>
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><br>
><br>
><br>
<br>
<br>
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</div></div></blockquote></div><br>