Hi Karsten Since posting the original message, I read a bit about Asterisk. So I know a little bit about both Freeswitch and Asterisk. A little knowledge being dangerous, I confess I need more details to poke around these two systems. I studied the URL you have given and it doesn't answer my question. What I would like to accomplish is: o Register Freeswitch extension as a UA with Asterisk (send a REGISTER command from Freeswitch to Asterisk just like a VoIP phone does) o Use Asterisk Dial action to call the Freeswitch o On Freeswitch play IVR (which I know how to do in Javascript) and send DTMF o Asterisk should receive the DTMF and hangup I would be grateful if you shed some more light on this issue. Thanks Murthy ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Karsten Horsmann Sent: Tuesday, September 24, 2013 1:18 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch register as UA Hi Murthy, IMHO you could play around with the bridge parameter like this: <action application="bridge" data="{sip_invite_tel_params=rn=12135550000;npdi=yes,sip_invite_params=user=phone}sofia/gateway/gateway_name/12135551212"/> or <action application="bridge" data="sip:12135551212;rn=12135550000;npdi=yes@1.2.3.4:5060^12135551212> See https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge 2013/9/19 Murthy Gandikota <mgandikota@nts.net> 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 <http://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: https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk <https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk> wiki.freeswitch.org/wiki/Freeswitch_softphone <http://wiki.freeswitch.org/wiki/Freeswitch_softphone> Any help gratefully appreciated. 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