<action application="bridge" data="{sip_invite_tel_params=rn=12135550000;npdi=yes,sip_invite_params=user=phone}sofia/gateway/gateway_name/12135551212"/>
orSee https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
<action application="bridge" data="sip:12135551212;rn=12135550000;npdi=yes@1.2.3.4:5060^12135551212>
Hi Richard
Is there any way I can do this without touching the asterisk? One of the links below indeed nearly matches with your description.
I was of the impression that someone had already thought of using Freeswitch as a UA. How can Yealink phone do that
out of the box?
Thanks
From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Richard Genthner
Sent: Thursday, September 19, 2013 12:02 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch register as UA
Set it up as a gateway like so
<include>
<gateway name="office">
<param name="apply-inbound-acl" value="asterisk_box"/>
<param name="username" value="mykillerusername" />
<param name="realm" value="asterisk" />
<param name="proxy" value="<iptohost>" />
<param name="password" value="mybadasspassword" />
</gateway>
</include>
Stuff that in your public sip profiles reload sofia and you should be able to access everything from asterisk thats on freeswitch.
You will need to add the following to asterisk:
; Enable dialing to Moose directly
exten => 10002,1,Verbose(2,Dialing Moose at ${EXTEN})
exten => 10002,n,Playback(silence/1&transfer)
exten => 10002,n,Dial(SIP/MOOSE/5000,20,rt)
exten => 10002,n,Hangup()
--
Thanks,
Richard Genthner
System Administrator
Symplicity
tel 703.351.0200 x 8051
web www.symplicity.com
On Sep 19, 2013, at 1:04 PM, "Murthy Gandikota" <mgandikota@nts.net> wrote:
Hi All
I want Freeswitch to register with Asterisk as a user agent.
To illustrate with Yealink phone I have the following configs:
a) username: cs3
b) SIP server: fs-murthy.nts.net
c) port: 5060
d) the ip of YeaLink phone itself (xx.xx.85.130)
Now I want Freeswitch to register with Asterisk with similar data. Since I don't know as much
about Asterisk as I know about Freeswitch, my goal is to bounce a call from Asterisk to
Freeswitch and let it play an IVR.
Here are some links I already checked:
Any help gratefully appreciated.
Murthy
_________________________________________________________________________
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
_________________________________________________________________________
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