Send calls through registered gateway
Hello guys, I seem to be unable to send calls through a registered gateway... is there any sample dialplan available? thanks David
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
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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On Wed, Mar 14, 2012 at 6:32 PM, David Villasmil <david.villasmil.work@gmail.com> wrote:
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).
I think you may have it backwards. If you want your Asterisk server to act as the GW to FreeSWITCH (you send calls from FS to Asterisk and it hit's its own dialplan), then you would setup a gateway in FreeSWITCH (see the sofia profie). You don't *need* FreeSWITCH to register to its GWs, but you can if you like. Then from FS's perspective, you would dial it like so: sofia/gateway/NAME_OF_GW/NUMBER_TO_DIAL I'm sorry if I misunderstood what you're doing, but I hope this helps. Best, Gabe
HEllo, Thanks for the reply. Tried that but doesn't work. I register asterisk on FS. FS sends an "s@whatever" in the invite, with the actual number on the "Contact" header. but I set this on asterisk dialplan and now it works: exten => s,1,Goto(CONTEXT,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) thanks! On Thu, Mar 15, 2012 at 3:22 AM, Gabriel Gunderson <gabe@gundy.org> wrote:
On Wed, Mar 14, 2012 at 6:32 PM, David Villasmil <david.villasmil.work@gmail.com> wrote:
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).
I think you may have it backwards. If you want your Asterisk server to act as the GW to FreeSWITCH (you send calls from FS to Asterisk and it hit's its own dialplan), then you would setup a gateway in FreeSWITCH (see the sofia profie). You don't *need* FreeSWITCH to register to its GWs, but you can if you like.
Then from FS's perspective, you would dial it like so: sofia/gateway/NAME_OF_GW/NUMBER_TO_DIAL
I'm sorry if I misunderstood what you're doing, but I hope this helps.
Best, Gabe
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (2)
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David Villasmil
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Gabriel Gunderson