Here's my setup:<br><br>conf/sip_profiles/outbound/lesnet.xml<br><br><include><br> <gateway name="lesnet"><br> <param name="username" value="1659525395"/><br> <param name="password" value="XXXXXX"/><br>
<param name="realm" value="<a href="http://did.voip.les.net">did.voip.les.net</a>"/><br> <param name="proxy" value="<a href="http://did.voip.les.net">did.voip.les.net</a>"/><br>
<param name="register" value="true"/><br> </gateway><br></include><br><br>Then:<br><br>sessionx.originate(session, "sofia/gateway/lesnet/1502XXXXXXX", 30);<br><br>HTH,<br>
<br>Nick<br><br><br><div class="gmail_quote">On Wed, Jun 4, 2008 at 6:30 PM, Klaus Teller <<a href="mailto:klaus.teller@gmx.net">klaus.teller@gmx.net</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Hi Brian,<br>
<br>
I created a LES profile located at sip_profiles/external/les.xml with following content:<br>
<br>
<include><br>
<gateway name="<a href="http://did.voip.les.net" target="_blank">did.voip.les.net</a>"><br>
<param name="username" value="1490236124"/><br>
<param name="realm" value="<a href="http://did.voip.les.net" target="_blank">did.voip.les.net</a>"/><br>
<param name="password" value="mash.n2rown4"/><br>
</gateway><br>
</include><br>
<br>
<br>
My dialing is actually done in a Javascript file in the following way:<br>
<br>
session.execute("bridge","sofia/gateway/<a href="http://did.voip.les.net/14156113200" target="_blank">did.voip.les.net/14156113200</a>");<br>
<br>
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.<br>
<br>
<br>
In the <a href="http://les.net" target="_blank">les.net</a> management console I createa a Peer/Trunk. To do this follow the following steps:<br>
<br>
1) Login at <a href="http://les.net" target="_blank">les.net</a><br>
2) On the left menu click on "Peers / Trunk"<br>
3) then click on "create peer"<br>
4) Next click on "Edit" to edit the settings for this trunk.<br>
5) Specify the IP Address and the password of the trunk.<br>
<br>
<br>
Hope this solves your problem.<br>
<br>
Klaus.<br>
<br>
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-------- Original-Nachricht --------<br>
> Datum: Wed, 4 Jun 2008 12:37:51 -0700<br>
> Von: "Brian B" <<a href="mailto:freeswitch@brian-burt.com">freeswitch@brian-burt.com</a>><br>
> An: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>
> Betreff: [Freeswitch-users] FS installed, but no external calls<br>
<div><div></div><div class="Wj3C7c"><br>
> FS is definitely up and running on my Lylix VPS, which is exciting.<br>
> X-Lite<br>
> is dialed in and can talk to the IVR etc. But no inbound or outbound<br>
> calls.<br>
><br>
> I tried configuring <a href="http://les.net" target="_blank">les.net</a> (which I got working in * on the same box) but<br>
> couldn't get that working, and sofia status is NOREG.<br>
><br>
> I got gizmo to where status is REGED but from XLite when I dial a 10 digit<br>
> number the console shows<br>
> [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing()<br>
> No<br>
> Route, Aborting<br>
><br>
><br>
> Any tips ...or is there an affordable contractor out there who can get a<br>
> "hello world" call going?<br>
><br>
><br>
><br>
> sofia status<br>
> API CALL [sofia(status)] output:<br>
> Name Type<br>
> Data State<br>
> =================================================================================================<br>
> internal profile<br>
> <a href="http://sip:mod_sofia@67.228.218.111:5060" target="_blank">sip:mod_sofia@67.228.218.111:5060</a><br>
> RUNNING (0)<br>
> nat profile<br>
> <a href="http://sip:mod_sofia@67.228.218.111:5070" target="_blank">sip:mod_sofia@67.228.218.111:5070</a><br>
> RUNNING (0)<br>
> default alias<br>
> internal ALIASED<br>
> external profile<br>
> <a href="http://sip:mod_sofia@67.228.218.111:5080" target="_blank">sip:mod_sofia@67.228.218.111:5080</a><br>
> RUNNING (0)<br>
> gizmo gateway<br>
</div></div>> <a href="mailto:sip%3A17476481805@proxy01.sipphone.com">sip:17476481805@proxy01.sipphone.com</a><<a href="mailto:sip%253A17476481805@proxy01.sipphone.com">sip%3A17476481805@proxy01.sipphone.com</a>><br>
> REGED<br>
> lesnet gateway<br>
> <a href="mailto:sip%3A16643@did.voip.les.net">sip:16643@did.voip.les.net</a><<a href="mailto:sip%253A16643@did.voip.les.net">sip%3A16643@did.voip.les.net</a>><br>
<div class="Ih2E3d">> NOREG<br>
> <a href="http://67.228.218.111" target="_blank">67.228.218.111</a> alias<br>
> internal ALIASED<br>
> outbound alias<br>
> external ALIASED<br>
> =================================================================================================<br>
><br>
><br>
> Thanks for any tips or suggestions!<br>
><br>
> -Brian B<br>
<br>
</div><font color="#888888">--<br>
Psssst! Schon vom neuen GMX MultiMessenger gehört?<br>
Der kann`s mit allen: <a href="http://www.gmx.net/de/go/multimessenger" target="_blank">http://www.gmx.net/de/go/multimessenger</a><br>
</font><div><div></div><div class="Wj3C7c"><br>
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