[WM: Absolutely.]

I ran apt-get to update my python packages and install the pyinstaller. pygooglevoice installed without issues over that. I confirmed that it was working by running from command-line.

Then I added the following to my extensions.ael.
in globals{}:
GV8_PARK=720;
GV8_NUM=888xxxxxxx;
GV8_ACCT=jay@xxx.net;
GV8_PASS=xxxx;
GV8_RING=1555xxxxxx;

New context:
context googlevoice8 {
_1NXXNXXXXXX =>
{
Ringing();
System(gvoice -e \${GV8_ACCT} -p \${GV8_PASS} call \${EXTEN} $
Set(PARKINGEXTEN=\${GV8_PARK});
Park();
}

and in my inbound-context:
555xxxxxxx =>
{ if("${CALLERID(number)}"="${GV8_NUM}") {
ParkedCall(${GV8_PARK});
} else {
&exec-incoming(us,2:,${EXT_ALL},4000,,yes);
}
}

However, I did have to make a tweaks to the script for stability (asterisk would crash and re-start when I tried the custom dialplan) and wanted to share with others who might have same problem. This could be a non issue on faster PCs (I’m using a slow laptop).

In the [custom-park] script, I had to change the first line to Wait(5) instead of Wait(2). I don’t know what the parameter does.. Does it increase the delay time?

[WM: It does increase the delay. We already had changed it to 4 to address another issue on some systems, but thanks.]

For 2 or more concurrent calls, we can use MeetMeCount to check if the conference is available. If not, go to next available room.

Bao Nguyen.

I also read other comments about Meetme not hangup, and how to make 2 concurrent calls or more. Here is my solution:

Note: My below instruction will allow you to use 2 concurrent calls through GV. Hangup issue with Meeetme also be resolved.

Requiremetns: 2 GV accounts with 2 callback Sipgate (SG) numbers.

0. Run the setup script install-gv

1. Create each inbound 1-DID rule for each pair of GV-SG.

2. Create 2 Conference Rooms, say 48001 & 48002 via FreePBX

3. Edit extensions_override_freepbx.conf, and add these lines:

;——————–start
[ext-meetme]
exten => 48001,1,Macro(user-callerid,)
exten => 48001,n,Set(MEETME_ROOMNUM=48001)
exten => 48001,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?USER)
exten => 48001,n,Answer
exten => 48001,n,Wait(1)
exten => 48001,n(USER),Set(MEETME_OPTS=qx)
exten => 48001,n,Goto(STARTMEETME,1)
exten => 48001,hint,MeetMe:48001

exten => 48002,1,Macro(user-callerid,)
exten => 48002,n,Set(MEETME_ROOMNUM=48002)
exten => 48002,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?USER)
exten => 48002,n,Answer
exten => 48002,n,Wait(1)
exten => 48002,n(USER),Set(MEETME_OPTS=qx)
exten => 48002,n,Goto(STARTMEETME,1)
exten => 48002,hint,MeetMe:48002
;———————end

Note: The reason for this step is to add an ‘x’ option to MEETME_OPTS.

4. Edit extensions_custom.conf, section [custom-gv]. Delete/comment out ALL the existing lines in this context, then add:

;————–start
exten => _X.,1,Wait(1)
exten => _X.,n,Playback(pls-wait-connect-call)

;room 1, GV acc 1
exten => _X.,n,Set(ACCTNAME=GVacc1)
exten => _X.,n,Set(ACCTPASS=****)
exten => _X.,n,Set(RINGBACK=SGNo1)
;exten => _X.,n,Set(GVCONFNO=48001)
exten => _X.,n,Set(GVCONFNO=48001)
;—————–

;—–Select the available room
exten => _X.,n,MeetMeCount(48001,mmCount)
exten => _X.,n,GotoIf($["${mmCount}" = "0″]?ROOMREADY)

;—————–room 2, GV acc 2
exten => _X.,n,Set(ACCTNAME=GVacc2)
exten => _X.,n,Set(ACCTPASS=****)
exten => _X.,n,Set(RINGBACK=SGNo2)
exten => _X.,n,Set(GVCONFNO=48002)
;—————–

exten => _X.,n(ROOMREADY),System(gvoice -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK})
exten => _X.,n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?USER)
exten => _X.,n,Answer
exten => _X.,n,Wait(1)
exten => _X.,n(USER),Set(MEETME_OPTS=Aq)
;start Meetme
exten => _X.,n,MeetMe(${GVCONFNO},${MEETME_OPTS},${PIN})
exten => _X.,n,Hangup

Then reload your asterisk server, and there you go.

This is the example of 2 concurrent calls, you can edit to have n concurrent calls with n pairs of GV-SG

(*) One of my Google Voice has problem with Sipgate account. GV can reach SG, but the call does not pass to Asterisk. Tried with other SG accounts with the same problems. I have to use another GV acc and it works properly

Best regards,

Bao Nguyen.
nttranbao

http://muffinworld.net/freepbx/auto_end_conference_instructions.txt

I’m curious to see how you solved this, Ward. Don’t make us wait until tomorrow!

[WM: See the updated article. Much simpler.]

After some research I discovered that you can create a conference with an "x" option which will end the conference when there are no "special" users left. I created two different extensions in extensions_custom.conf. The first extension is where the incoming callback from your GV number is sent. This joins the conference with options Mqx. M for music on hold enabled, q for quiet mode, and x to end the conference with no special users are left.
Then I created a second extension that my home phone extension is sent two after the ringback is initiated. This dumps the home phone into the same conference, but with the "A" flag which sets you as special user. When you hang up, the other person is ended. It even works with call waiting. I was able to flash and answer another call, and the waiting caller received music on hold until I came back. When I hung up, the conference ended. I will post back with some specific code.

[WM: Thanks. We’re incorporating that into the "refinement" article as well.]

[WM: That is a shortcoming with the MeetMe conference approach unless you check your Active Channels. A refinement is underway. Stay tuned!]

exten => ${SIPGATE},1,MeetMeCount(4356,mmCount)
exten => ${SIPGATE},n,GotoIf($["${mmCount}" != "1″]?incoming,s,1)
exten => ${SIPGATE},n,GotoIf($["${CALLERID(num)}" != "${MYGVNUMBER}"]?incoming,s,1)
exten => ${SIPGATE},n,MeetMe(4356,1wFqx)
exten => ${SIPGATE},n,Hangup()

[WM: That is an issue as are answering machines. Stay tuned. Refinements are in the works.]

"As this is written, the only restrictions on use to make free U.S. calls are those set forth in the Google Voice Terms of Service." Doesn’t it also depend on the free service provided by either IPKall or SIPgate? Don’t they provide the gateway functionality between PSTN and SIP?

In the Gizmo model, Google Voice is acting as gateway to bridge PSTN and SIP domain. With this new design, the DID numbers entered in Google Voice are regular PSTN numbers hence the inbound call made by Google is through PSTN, then switched into SIP by IPKall or SIPgate.

[WM: You’re correct. IPkall, at least, has been at this a very long time and is also an ILEC which means they get revenue from the caller every time a call is placed to one of their DIDs.]

Good news for anyone who hasn’t yet submitted an invite request — looks like the turn-around time has continued to shrink!

I’ve noticed the logical issue with dialing someone and getting their voicemail – is there a way to use meetme’s ‘wait for admin’ or similar feature to allow yourself to hang up the far end when you’re done leaving a message? As it currently stands, the conference stays up as long as the other user doesn’t hang up – in this example, AT&T or Verizon’s voicemail system.

Thoughts?

Obviously not a deal-killer. Really great work everyone!

Ward – I’m still ‘working on’ (read: thinking about working on) that OpenVPN tutorial for you…

1. Add dialplan for outgoing calls to call the AGI script and then Park()
2. Set your parking lot to use extention 8 (this is for brevity)
3. Set the incoming dialplan for (your_line/gvoice_num) to Wait(3) and then ParkedCall(8)

my extensions_custom looks like this:

[into_parked]
exten => 1747XXXXXXX/801XXXXXXX,1,Wait(2)
exten => 1747XXXXXXX/801XXXXXXX,2,ParkedCall(8)

[call_gvoice]
exten => _X.,1,AGI(gvoice/agiGVoice.php|${EXTEN})
exten => _X.,2,Park(8)

When you call out, you’ll get a dealy while it logs on and tells gvoice to make the call, and then you’ll hear "8″ (parking in spot 8), and then it will start ringing as the incoming call is connected to that parking lot.

Has been working great for me for awhile; you can make multiple outgoing calls just fine, though you need multiple google voice numbers and you can’t make them at the same time.

So in the inbound route, if the CID on that DID match GV#, point that to your unique conference, otherwise, point it to your inbound extension.

It depends whether the RingBack calls display the number of the person being called or the GV main number. If it’s the latter, it would work. But I suspect it’s going to show the number of the person being called. Try it and let us know.]