BTW,<br><br>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...<br><br>-MC<br><br><div class="gmail_quote">
On Tue, Jul 27, 2010 at 7:46 AM, mazilo <span dir="ltr"><<a href="mailto:Nabble@slickdeals.endjunk.com">Nabble@slickdeals.endjunk.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
<div class="im"><br>
<br>
mercutioviz wrote:<br>
> > What Asterisk's Bridge() function does<br>
>> is to bridge the incoming calls from Google Voice with the existing<br>
>> (on-going) outbound calls to Google Voice by means of the Python scripts<br>
>> file.<br>
><br>
> This is where you lost us. What existing call is being bridged to? I count<br>
> only two calls:<br>
> from GV back to the user (via Gizmo or whatever contact number the user<br>
> has)<br>
> from GV to PSTN (to reach the called party)<br>
><br>
> I'm assuming that the ultimate result of the above is that the user will<br>
> be<br>
> talking to the destination:<br>
> GV user <--> Called party<br>
><br>
> Where is the outgoing call to GV? Is there a third call that we're not<br>
> aware<br>
> of? If so, what is its purpose?<br>
</div>You are right. Let's see if I can manage to explain it better.<br>
<br>
When I pick up a phone (connected to a Linksys PAP2v2 unit configured as an<br>
extension to my asterisk device) and dial a US PSTN number, one of the<br>
dialplans intercepts to execute the Google Voice Dialer scripts file using<br>
an AGI function. AGI locally creates (on my asterisk system) a SIP channel<br>
ID (SIP/1005-00000049) as shown below while the scripts perform two<br>
functions, i.e. to emulate a login process to Google Voice account and to<br>
interact with the Google Voice's click2call function to place an outbound<br>
call to a US PSTN network (dialed number). When Google Voice receives the<br>
click2call action, it performs two calls, i.e. an outbound call to the<br>
number listed on the forwarding to folder (in this case, my Gizmo5 number)<br>
and then another outbound call to the dialed number (a US PSTN number). The<br>
later part will be executed once the former outbound call to my Gizmo5<br>
number has been established between Google Voice and the telephone unit<br>
connected to my Linksys PAP2v2 unit (configured as an extension to my<br>
asterisk device). When my asterisk system receives the incoming call on its<br>
Gizmo5 trunk, the asterisk Bridge() function assosociates and assigns this<br>
incoming call to the existing call with a SIP channel ID<br>
(SIP/1005-00000049). At this time, I will hear the calling tone (or even<br>
busy signals). When the callee party picks up his/her phone, then the<br>
conversation starts between the caller and callee (like business as usual).<br>
<br>
I am sure FreeSWITCH can do this better than Asterisk. The problem with me<br>
being a newbie to FreeSWITCH is I don't know how to do this. I am hoping<br>
someone who has done this will be able to help. After all, the Google Voice<br>
DialOut Python scripts file can be freely downloaded/read from<br>
<a href="http://www.pmarks.net/posted_links/google-voice-dialout.agi" target="_blank">http://www.pmarks.net/posted_links/google-voice-dialout.agi</a> here .<br>
<br>
-- Launched AGI Script /usr/lib/asterisk/agi-bin/google-voice-dialer.agi<br>
<SIP/1005-00000049>AGI Tx >> agi_request: google-voice-dialer.agi<br>
<SIP/1005-00000049>AGI Tx >> agi_channel: SIP/1005-00000049<br>
<SIP/1005-00000049>AGI Tx >> agi_language: en<br>
<SIP/1005-00000049>AGI Tx >> agi_type: SIP<br>
<SIP/1005-00000049>AGI Tx >> agi_uniqueid: 1280239277.95<br>
<SIP/1005-00000049>AGI Tx >> agi_version: 1.6.2.9<br>
<SIP/1005-00000049>AGI Tx >> agi_callerid: 1005<br>
<SIP/1005-00000049>AGI Tx >> agi_calleridname: 11<br>
<SIP/1005-00000049>AGI Tx >> agi_callingpres: 0<br>
<SIP/1005-00000049>AGI Tx >> agi_callingani2: 0<br>
<SIP/1005-00000049>AGI Tx >> agi_callington: 0<br>
<SIP/1005-00000049>AGI Tx >> agi_callingtns: 0<br>
<SIP/1005-00000049>AGI Tx >> agi_dnid: 1404XXXXXXX<br>
<SIP/1005-00000049>AGI Tx >> agi_rdnis: unknown<br>
<SIP/1005-00000049>AGI Tx >> agi_context: phones<br>
<SIP/1005-00000049>AGI Tx >> agi_extension: 1404XXXXXXX<br>
<SIP/1005-00000049>AGI Tx >> agi_priority: 13<br>
<SIP/1005-00000049>AGI Tx >> agi_enhanced: 0.0<br>
<SIP/1005-00000049>AGI Tx >> agi_accountcode:<br>
<SIP/1005-00000049>AGI Tx >> agi_threadid: 65546<br>
<SIP/1005-00000049>AGI Tx >> agi_arg_1: <USERNAME><br>
<SIP/1005-00000049>AGI Tx >> agi_arg_2: <PASSWORD><br>
<SIP/1005-00000049>AGI Tx >> agi_arg_3: 747XXXXXXX<br>
<SIP/1005-00000049>AGI Tx >> agi_arg_4: 404XXXXXXX<br>
<SIP/1005-00000049>AGI Tx >><br>
<div class="im"><br>
<br>
-----<br>
don't and stop are the ONLY two 4-letter words considered offensive to men,<br>
but not when used together.<br>
--<br>
</div>View this message in context: <a href="http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-function-implementation-tp5303089p5342450.html" target="_blank">http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-function-implementation-tp5303089p5342450.html</a><br>
<div><div></div><div class="h5">Sent from the freeswitch-users mailing list archive at Nabble.com.<br>
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