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

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
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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




--
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