[Freeswitch-users] Asterisk equivalent Bridge() function implementation

[Michael Collins]

BTW,

This actually makes sense now that you spelled it out. Most likely you just
need uuid_bridge. If you know the uuid's of the two channels then you can do
whatever you want...

-MC

On Tue, Jul 27, 2010 at 7:46 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

>
>
> mercutioviz wrote:
> > > What Asterisk's Bridge() function does
> >> is to bridge the incoming calls from Google Voice with the existing
> >> (on-going) outbound calls to Google Voice by means of the Python scripts
> >> file.
> >
> > This is where you lost us. What existing call is being bridged to? I
> count
> > only two calls:
> > from GV back to the user (via Gizmo or whatever contact number the user
> > has)
> > from GV to PSTN (to reach the called party)
> >
> > I'm assuming that the ultimate result of the above is that the user will
> > be
> > talking to the destination:
> > GV user <--> Called party
> >
> > Where is the outgoing call to GV? Is there a third call that we're not
> > aware
> > of? If so, what is its purpose?
> You are right. Let's see if I can manage to explain it better.
>
> When I pick up a phone (connected to a Linksys PAP2v2 unit configured as an
> extension to my asterisk device) and dial a US PSTN number, one of the
> dialplans intercepts to execute the Google Voice Dialer scripts file using
> an AGI function. AGI locally creates (on my asterisk system) a SIP channel
> ID (SIP/1005-00000049) as shown below while the scripts perform two
> functions, i.e. to emulate a login process to Google Voice account and to
> interact with the Google Voice's click2call function to place an outbound
> call to a US PSTN network (dialed number). When Google Voice receives the
> click2call action, it performs two calls, i.e. an outbound call to the
> number listed on the forwarding to folder (in this case, my Gizmo5 number)
> and then another outbound call to the dialed number (a US PSTN number). The
> later part will be executed once the former outbound call to my Gizmo5
> number has been established between Google Voice and the telephone unit
> connected to my Linksys PAP2v2 unit (configured as an extension to my
> asterisk device). When my asterisk system receives the incoming call on its
> Gizmo5 trunk, the asterisk Bridge() function assosociates and assigns this
> incoming call to the existing call with a SIP channel ID
> (SIP/1005-00000049). At this time, I will hear the calling tone (or even
> busy signals). When the callee party picks up his/her phone, then the
> conversation starts between the caller and callee (like business as usual).
>
> I am sure FreeSWITCH can do this better than Asterisk. The problem with me
> being a newbie to FreeSWITCH is I don't know how to do this. I am hoping
> someone who has done this will be able to help. After all, the Google Voice
> DialOut Python scripts file can be freely downloaded/read from
> http://www.pmarks.net/posted_links/google-voice-dialout.agi here .
>
>    -- Launched AGI Script /usr/lib/asterisk/agi-bin/google-voice-dialer.agi
> <SIP/1005-00000049>AGI Tx >> agi_request: google-voice-dialer.agi
> <SIP/1005-00000049>AGI Tx >> agi_channel: SIP/1005-00000049
> <SIP/1005-00000049>AGI Tx >> agi_language: en
> <SIP/1005-00000049>AGI Tx >> agi_type: SIP
> <SIP/1005-00000049>AGI Tx >> agi_uniqueid: 1280239277.95
> <SIP/1005-00000049>AGI Tx >> agi_version: 1.6.2.9
> <SIP/1005-00000049>AGI Tx >> agi_callerid: 1005
> <SIP/1005-00000049>AGI Tx >> agi_calleridname: 11
> <SIP/1005-00000049>AGI Tx >> agi_callingpres: 0
> <SIP/1005-00000049>AGI Tx >> agi_callingani2: 0
> <SIP/1005-00000049>AGI Tx >> agi_callington: 0
> <SIP/1005-00000049>AGI Tx >> agi_callingtns: 0
> <SIP/1005-00000049>AGI Tx >> agi_dnid: 1404XXXXXXX
> <SIP/1005-00000049>AGI Tx >> agi_rdnis: unknown
> <SIP/1005-00000049>AGI Tx >> agi_context: phones
> <SIP/1005-00000049>AGI Tx >> agi_extension: 1404XXXXXXX
> <SIP/1005-00000049>AGI Tx >> agi_priority: 13
> <SIP/1005-00000049>AGI Tx >> agi_enhanced: 0.0
> <SIP/1005-00000049>AGI Tx >> agi_accountcode:
> <SIP/1005-00000049>AGI Tx >> agi_threadid: 65546
> <SIP/1005-00000049>AGI Tx >> agi_arg_1: <USERNAME>
> <SIP/1005-00000049>AGI Tx >> agi_arg_2: <PASSWORD>
> <SIP/1005-00000049>AGI Tx >> agi_arg_3: 747XXXXXXX
> <SIP/1005-00000049>AGI Tx >> agi_arg_4: 404XXXXXXX
> <SIP/1005-00000049>AGI Tx >>
>
>
> -----
> don't and stop are the ONLY two 4-letter words considered offensive to men,
> but not when used together.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-function-implementation-tp5303089p5342450.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20100728/fec40cc7/attachment-0001.html