<div>Hello,</div><div><br></div><div>Thanks for the reply.</div><div><br></div><div>Not what I'm looking for though.</div><div>I register an asterisk with with, say with username 1001. I receive a call from user 2 and want to terminate what 1002 sends with the gateway (not a gateway per se, but a registered user).</div>
<div><br></div><div>in the dialplan I have:</div><div><br></div><div><div> <action application="bridge" data="sofia/internal/1001%domain^NUMBER_TO_DIAL"/></div></div><div><br></div><div>
<br></div><div>But all asterisk gets is a call to extension "s" in the INVITE, even though the "To" header is OK.... how do I set it correctly??</div><div><br></div><div><div>INVITE <a href="mailto:sip%3As@5.6.7.8">sip:s@5.6.7.8</a> SIP/2.0</div>
<div>Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgmBy3gy8BSKZB</div><div>Max-Forwards: 69</div><div>From: "David Villasmil" <<a href="mailto:sip%3A1002@1.2.3.4">sip:1002@1.2.3.4</a>>;tag=1ep74eBrB2Fcj</div>
<div>To: <<a href="mailto:sip%3A912074419@1.2.3.4">sip:912074419@1.2.3.4</a>></div><div>Call-ID: 3393725d-e8d7-122f-6395-0013725ca38a</div><div>CSeq: 25549284 INVITE</div><div>Contact: <<a href="http://sip:mod_sofia@1.2.3.4:5060">sip:mod_sofia@1.2.3.4:5060</a>></div>
<div>User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-</div><div>Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE</div><div>Supported: timer, precondition, path, replaces</div>
<div>Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer</div><div>Content-Type: application/sdp</div><div>Content-Disposition: session</div>
<div>Content-Length: 482</div><div>X-FS-Support: update_display</div><div>Remote-Party-ID: "David Villasmil" <<a href="mailto:sip%3A1002@1.2.3.4">sip:1002@1.2.3.4</a>>;party=calling;screen=yes;privacy=off</div>
</div><div><br></div><br><div class="gmail_quote">On Thu, Mar 15, 2012 at 1:23 AM, Gabriel Gunderson <span dir="ltr"><<a href="mailto:gabe@gundy.org">gabe@gundy.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="im">On Wed, Mar 14, 2012 at 5:58 PM, David Villasmil<br>
<<a href="mailto:david.villasmil.work@gmail.com">david.villasmil.work@gmail.com</a>> wrote:<br>
> I seem to be unable to send calls through a registered gateway... is there<br>
> any sample dialplan available?<br>
<br>
</div><a href="http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways" target="_blank">http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways</a><br>
<br>
Best,<br>
Gabe<br>
<br>
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