[Freeswitch-users] Freeswitch register as UA
Murthy Gandikota
mgandikota at nts.net
Fri Sep 27 19:40:53 MSD 2013
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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Saugort Dario Garcia Tovar
Sent: Thursday, September 26, 2013 11:10 AM
To: freeswitch-users at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 1.2.3.4:5060^12135551212>
See https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge
2013/9/19 Murthy Gandikota <mgandikota at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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
_________________________________________________________________________
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_________________________________________________________________________
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Mit freundlichen Grüßen
*Karsten Horsmann*
_________________________________________________________________________
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http://www.freeswitchsolutions.com
Official FreeSWITCH Sites
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--
Atentamente,
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Consultor.
CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
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Cel: +58 412 2221515
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