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****
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Set it up as a gateway like so****
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<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>****
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Stuff that in your public sip profiles reload sofia and you should be able to access everything from asterisk thats on freeswitch. ****
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You will need to add the following to asterisk:****
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; 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()****
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-- Thanks,
Richard Genthner System Administrator Symplicity tel 703.351.0200 x 8051 web www.symplicity.com****
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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:********
****
https://wiki.freeswitch.org/wiki/Connecting_Freeswitch_and_Asterisk******* *
****
wiki.freeswitch.org/wiki/Freeswitch_softphone********
****
Any help gratefully appreciated.********
Murthy********
****
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** **
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