Hello, Thanks for the reply. Not what I'm looking for though. 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). in the dialplan I have: <action application="bridge" data="sofia/internal/1001%domain^NUMBER_TO_DIAL"/> 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?? INVITE sip:s@5.6.7.8 SIP/2.0 Via: SIP/2.0/UDP 1.2.3.4;rport;branch=z9hG4bKgmBy3gy8BSKZB Max-Forwards: 69 From: "David Villasmil" <sip:1002@1.2.3.4>;tag=1ep74eBrB2Fcj To: <sip:912074419@1.2.3.4> Call-ID: 3393725d-e8d7-122f-6395-0013725ca38a CSeq: 25549284 INVITE Contact: <sip:mod_sofia@1.2.3.4:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 482 X-FS-Support: update_display Remote-Party-ID: "David Villasmil" <sip:1002@1.2.3.4
;party=calling;screen=yes;privacy=off
On Thu, Mar 15, 2012 at 1:23 AM, Gabriel Gunderson <gabe@gundy.org> wrote:
On Wed, Mar 14, 2012 at 5:58 PM, David Villasmil <david.villasmil.work@gmail.com> wrote:
I seem to be unable to send calls through a registered gateway... is there any sample dialplan available?
http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gate...
Best, Gabe
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