<div dir="ltr">You have to create it as a trunk, not a user. I have a SPA3000 (earlier version of the 3102 I think) working this way. For host URL enter "dynamic" this tells Asterisk not to register with the service provider, but rather that the service provider (the SPA3102) will register with Asterisk. enter a userid/password that the SPA will use to Authenticate with Asterisk. My context in Users.conf ended up looking like this... (this was built with the v1 GUI, but I manually added some lines like qualify. v2 should build a similar entry for you)<br>
<br><span class="Apple-style-span" style="border-collapse: separate; color: rgb(0, 0, 0); font-family: 'Lucida Grande'; font-size: 12px; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-indent: 0px; text-transform: none; white-space: pre-wrap; widows: 2; word-spacing: 0px;">[localPSTN]<br>
allow=ulaw,alaw,gsm,g726<br>context=DID_localPSTN<br>dialformat=${EXTEN:1}<br>canreinvite=no<br>hasexten=no<br>hasiax=no<br>hassip=yes<br>host=dynamic<br>type=friend<br>qualify=yes<br>registeriax=no<br>registersip=no<br>secret=password<br>
trunkname=Custom - localPSTN<br>trunkstyle=customvoip<br>username=localPSTN<br>insecure=very<br>dtmfmode=auto</span><br><br>You also need to configure the PSTN Line of the SPA to enable the PSTN to VoIP Gateway and set a PSTN answer delay of about 3 seconds (to give time for the CallerID to come through) and set the PSTN Caller Default DP to one of the dial plans (I used "1"). You need to set up this Dial Plan to tell the SPA to call through to Asterisk on incoming calls. In my case I set "(S0<:102>)" which means zero seconds delay, try and call extension 102. But the extension number is actually not used by asterisk, as it hits the incoming call rules logic rather than go to the extension. <br>
<br>Hope this helps.<br><br>David<br><br><div class="gmail_quote">On Wed, Sep 17, 2008 at 4:42 PM, Fadi Almasalha <span dir="ltr"><<a href="mailto:falmas2@uic.edu">falmas2@uic.edu</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;">
You can't Create an analog trunk without any hardware FXO PCI Cards<br>
<div><div></div><div class="Wj3C7c"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-gui-bounces@lists.digium.com">asterisk-gui-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-gui-bounces@lists.digium.com">asterisk-gui-bounces@lists.digium.com</a>] On Behalf Of Pari Nannapaneni<br>
Sent: Wednesday, September 17, 2008 3:39 PM<br>
To: Asterisk GUI project discussion<br>
Subject: Re: [asterisk-gui] Linksys ATA SPA3102 Using The GUI<br>
<br>
> I have created a regular user account and linked it to the SPA3102<br>
<br>
you should probably create an analog trunk and connect the SPA3102<br>
like just any other PSTN connection.<br>
<br>
-Pari<br>
<br>
----- Original Message -----<br>
From: "Fadi Almasalha" <<a href="mailto:falmas2@uic.edu">falmas2@uic.edu</a>><br>
To: <a href="mailto:asterisk-gui@lists.digium.com">asterisk-gui@lists.digium.com</a><br>
Sent: Wednesday, September 17, 2008 3:23:33 PM GMT -06:00 US/Canada Central<br>
Subject: [asterisk-gui] Linksys ATA SPA3102 Using The GUI<br>
<br>
Hi ,<br>
<br>
I am not able to create a sip trunk using the asterisk GUI 2.0 for Linksys SPA 3102 . Please any ideas how to create a sip trunk on the GUI .<br>
<br>
I have created a regular user account and linked it to the SPA3102 , It receive calls and make calls with no problem . but the only problem that I can't configure the incoming call rules , as it only support the trunk method .<br>
<br>
Yours Fadi<br>
<br>
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