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. Murthy ________________________________________________________________________ _ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org> http://wiki.freeswitch.org <http://wiki.freeswitch.org> http://www.cluecon.com <http://www.cluecon.com> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <http://www.freeswitch.org>

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. Murthy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org> http://wiki.freeswitch.org <http://wiki.freeswitch.org> http://www.cluecon.com <http://www.cluecon.com> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <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 -- Mit freundlichen Grüßen *Karsten Horsmann*

I have a partial solution. Here's how I did it: a) connected Freeswitch to Asterisk server (https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk) b) dialed Freeswitch extension in Asterisk as exten => 1100,1,Dial(SIP/freeswitch_1/${EXTEN},30,M(testme)) Where the macro testme is: [macro-testme] exten => s,1,NoOp() exten => s,n,Set(TIMEOUT(digit)=100) exten => s,n,Verbose(3,in macro testme) exten => s,n,Read(DIGIT_PRESSED,,1,,100) exten => s,n,Set(GLOBAL(DIGIT_PRESSED)=${DIGIT_PRESSED}) exten => s,n,Verbose(DIGIT PRESSED=${DIGIT_PRESSED}) exten => s,n,Set(MACRO_RESULT=CONTINUE) c) Then transferred the call same => n,GotoIf($[${DIALSTATUS}=ANSWER]?answered:noAnswer) same => n(answered),Verbose(3, digit pressed=${DIGIT_PRESSED}) same => n,GotoIf("${DIGIT_PRESSED}" = "3" ?call-agi:noDigit) I could've created an AGI to do the same. But I wanted to learn a bit about both the systems. There you have it. ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Saugort Dario Garcia Tovar Sent: Thursday, September 26, 2013 11:10 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch register as UA Ok.... If I understand what are you trying to do is make Freeswtich act as a softphone/SIP extension/SIP UA/etc for your asterisk pbx. I think this links will help you: http://wiki.freeswitch.org/wiki/Freeswitch_softphone and http://wiki.freeswitch.org/wiki/Traditional_Gateway_connecting On 09/26/2013 11:24 AM, Murthy Gandikota wrote: 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. Murthy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com> FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com <http://www.cudatel.com> Official FreeSWITCH Sites http://www.freeswitch.org <http://www.freeswitch.org> http://wiki.freeswitch.org <http://wiki.freeswitch.org> http://www.cluecon.com <http://www.cluecon.com> FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> http://www.freeswitch.org <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 -- Mit freundlichen Grüßen *Karsten Horsmann* _________________________________________________________________________ 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 -- Atentamente, Dario García Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve