outbound calls via Google Voice

I was able to get it working using Park() but I don't like that option because it's not "transparent" since it results in Asterisk reading you the slot number the call is parked on followed by a moment of MOH when you dial out. I don't understand AGI and Dialplans well enough to figure out how to use the Bridge() option.

NerdVittles has promised a "refined" version of the GV Dialout solution soon. This has been a great learning experience for me so far... I just wish I could find someone to explain how to get it working using Bridge() instead of Park() or a MeetMe conference since I get the impression that an understanding of that would lead to a better understanding of dialplans in general.

bridge() joins the specified channel to the current channel.

This little code example would dial SIP extension 200 and then bridge Skype 42424242424

exten => s,1,Wait(2)
exten => s,n,Dial(sip/200)
exten => s,n,Wait(2)
exten => s,n,Bridge(Skype/42242424242)
exten => s,n,Hangup

Have you checked out the eBook at www.asteriskdocs.org? It is a huge help in understanding Asterisk extension programming.

...is a bloody good book, I managed to pick up a hard copy from eBay for a reasonable price.

I always find a physical book more satisfying although it is easier to find things in the electronic version. But then again, you can have the book next to the screen which you can't do with the electronic one...

Anyway, good book, worth getting yourself a copy.

I have a Sony Reader which is nice for eBooks like this...

I see that "Asterisk: The Future of Telephony 2nd Edition" covers Asterisk 1.4. If I understand things correctly, Bridge() is new to Asterisk 1.6. I'll still go through the book to learn what I can and maybe it'll give me enough information to fill in the blanks so I can get a more complete picture.

NOTE: the forum software is mangling the URL in the wget line in the script below. Be wary of that when you do your cut-n-paste.
And that was exactly what I needed in order to figure it out.

Below are instructions for getting Google Voice Dialout working under TB CE 2.8. Please keep in mind, I installed trixbox on June 20th. Before that, I had never touched an Asterisk system before. These steps worked for me but it's quite possible (okay, it's LIKELY) that this is NOT the best way to do this. It is, at least, a better option for Asterisk 1.6 users than the MeetMe conference option or the Park() option. I'm doing this with a Gizmo5 trunk.

1) Set up a Gizmo5 trunk and set its context as "context=gv-inbound"
2) Create a Custom Trunk with the following Custom Dial String:
local/$OUTNUM$@gv-outbound
3) Set up an outbound route to use that Custom Trunk.
4) Save the following script to a file and then run it:

#!/bin/bash

clear
echo "This script installs Google Voice on your Asterisk 1.6 system."
echo "WARNING: No error checking is provided."
echo "By using this script, you agree to assume ALL RISK."
echo "NO WARRANTY, EXPRESS OR IMPLIED, OF ANY KIND IS PROVIDED."
echo " "
echo "If you make a typo while entering values below, press Ctrl-C and start over."
echo " "
echo "Your Google Voice entries are stored in /etc/asterisk/extensions_custom.conf."
echo "Edit that file and reload your Asterisk dialplan if you make future changes."
echo " "
echo -n "Google Voice email address: "
read acctname
echo "ACCTNAME: $acctname"
echo -n "Google Voice password: "
read acctpass
echo "ACCTPASS: $acctpass"
echo -n "10-digit Google Voice Phone Number (e.g. 6127654321): "
read gvnum
echo "GVNUMBER: $gvnum"
echo -n "11-digit Ring Back DID (e.g. 16781234567): "
read ringback
echo "RINGBACK: $ringback"
echo " "
echo "We're now ready to begin the installation."
echo "Check your entries above and then press Enter key to proceed."
echo "Or press Ctrl-C to abort and try again."
read readysetgo
clear
echo "Installing Google Voice support for Asterisk 1.6. One moment please..."
echo " "

echo " " >> /etc/asterisk/extensions_custom.conf
echo "[gv-inbound]" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,1,Set(DID_EXTEN=\${SIP_HEADER(To):5})" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,n,Set(DID_EXTEN=\${CUT(DID_EXTEN,@,1)})" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,n,GotoIf(\$[\${CALLERID(num)} = $gvnum]?:normalcall)" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,n,NoCDR()" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,n,Bridge(\${DB_DELETE(gv_dialout/channel)})" >> /etc/asterisk/extensions_custom.conf
echo "exten => s,n(normalcall),Goto(from-trunk,\${DID_EXTEN},1)" >> /etc/asterisk/extensions_custom.conf
echo " " >> /etc/asterisk/extensions_custom.conf
echo "[gv-outbound]" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,1,Wait(1)" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Set(ACCTNAME=$acctname)" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Set(ACCTPASS=$acctpass)" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Set(RINGBACK=$ringback)" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,System(gvoice -e \${ACCTNAME} -p \${ACCTPASS} call \${EXTEN} \${RINGBACK})" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Set(DB(gv_dialout/channel)=\${CHANNEL})" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Wait(5)" >> /etc/asterisk/extensions_custom.conf
echo "exten => _X.,n,Hangup" >> /etc/asterisk/extensions_custom.conf
echo " " >> /etc/asterisk/extensions_custom.conf

echo "Reloading Asterisk dialplan..."
asterisk -rx "dialplan reload"
echo " "

echo "Installing Python add-ons..."
cd /root
yum -y install python-setuptools
easy_install simplejson

echo " "
echo "Installing pygooglevoice..."

wget <a href="http://pygooglevoice.googlecode.com/files/pygooglevoice-0.2.tar.gz" title="http://pygooglevoice.googlecode.com/files/pygooglevoice-0.2.tar.gz">http://pygooglevoice.googlecode.com/files/pygooglevoice-0.2.tar.g...</a>
tar zxvf pygooglevoice-0.2.tar.gz

cd pygooglevoice-0.2
python setup.py install

sed -i 's|    help(|#    help(|' /usr/bin/gvoice

echo " "
echo "Installation script is finished. Do NOT run it again on this system!"

Most of this script was taken from the NerdVittles script.

Since I first posted this script, I have been able to test and debug it. I have edited this post to accommodate the changes that were required to make it work properly. It works fine with Gizmo5. I tried to get it working with sipgate but was unsuccessful (I didn't put in much time figuring out why).

I like using Gizmo5 for this because of their "direct" connection to Google Voice for the inbound calls. Using sipgate or IPKall means you're going from your Asterisk box to the provider via SIP, then out over the PSTN to Google where it becomes SIP again and then back out over the PSTN to the destination. My understanding of the Google/Gizmo5 connection is that the call remains SIP from Google to Gizmo to Asterisk. Does this make any real difference? Eh. Maybe. I just know that it's a lot easier to configure. Heh.

What I like best about using Bridge() instead of Park() or MeetMe is that the experience is transparent. The only difference is that it takes an extra two seconds or so before you hear the call ringing. There's no MOH or Parking Lot number announcement. It behaves like any other outbound call.

I'll tinker some more this weekend to see if I can figure out why it wasn't working with my sipgate connection. I'll look into IPKall as well.

Okay, if you set up IPKall the way he describes in his instructions, you've configured a Custom Destination that sends calls to "custom-park,s,1". Change that to "gv-inbound,s,1".

Then add this to extensions_custom.conf (replacing NXXNXXXXXX with your 10-digit Google Voice number):

[gv-inbound]
exten => s,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,GotoIf($[${CALLERID(num)} = NXXNXXXXXX]?:normalcall)
exten => s,n,NoCDR()
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})
exten => s,n(normalcall),Goto(from-trunk,${DID_EXTEN},1)

Next, find the [custom-gv] context and comment out (or just delete) these lines:

exten => _X.,n,Playback(pls-wait-connect-call)
exten => _X.,n,Set(PARKINGEXTEN=\${CALLPARK})
exten => _X.,n,Park()

Then add these lines to the end of the [custom-gv] context:

exten => _X.,n,Set(DB(gv_dialout/channel)=${CHANNEL})
exten => _X.,n,Wait(5)
exten => _X.,n,Hangup

I think that will work. Let me know how it goes.

wtf is up with the board trying to put a wrapper around the wget url in my [code ] tags

both tags have the problem, but it does not occur if the php or code opening or closing tag is on the same line as the link.
php tag

http://www.google.com
<a href="http://www.google.com" title="http://www.google.com">http://www.google.com</a>
http://www.google.com

code tag

http://www.google.com
<a href="http://www.google.com" title="http://www.google.com">http://www.google.com</a>
http://www.google.com

Something that put me off about the IPKall stuff is the need to add more custom stuff to config files for the incoming SIP connection. But this thread seems to imply you can avoid that by using IAX instead of SIP for IPKall since you can use authentication. Might be worth a try.

I am getting an "all circuits ar busy at this time" recording when trying to dial out 48|mynumber. I've done all the steps listed above in the final version... when trying to paste sed -i 's| help(|# help(|' /usr/bin/gvoice into my SSH it gave me all kinda gibberish, however. New to RedHat Linux stuff...

ghost, you need to be careful on the spacing of the sed command. sed is a find/replace command, if you have the spacing wrong, you will be find/replacing the wrong information.

The information in my post was pulled directly from the Nerdvittles.com install script. I simply did not want to run a script, preferring to change everything manually.

For whom want to used GoogleVoice with DID of IPKALL number on Trixbox 2.8.0.1.

at sip_custum.conf
Add
context=from-trunk

make sure you can call google number via ipkall to your trixbox with exten that you put on ipkall acount.

and follow all step that "sorvani" post on this page
only need to change the RINGBACK phone number from google voice number to ipkall number as :

from
exten => _X.,n,Set(RINGBACK=1NXXNXXXXXXX) ; to
exten => _X.,n,Set(RINGBACK=14252222222) ;

It will work.

enjoy freedom of free

ps: put google voice number at Inbound Caller ID: NxxNxxNxxx at add custum_Trunk step.(for incoming call)

You can't get it work on your tbce version 2.6.2.3
because tb 2.6 work under asterisk 1.4 without support Bridge function.
you have 2 choice.
1. Download tbce 2.8.0.1 clean install.
2. or remain use tbce 2.6.2.3 and follow with this link provide by Nerd Vittles.
http://nerdvittles.com/?m=20090805

SED command on this forum only put "#" sign on front of 2nd last statment of file ( /usr/bin/gvoice).

try your best to understand linux command line function.

Instead of screening incoming calls for callerid to determine whether to bridge the call or not - you could set a global variable in gv-outbound and use this as a flag in gv-inbound to either bridge or not bridge the inbound call. That way it makes it easier to use multiple GV trunks, and allows normal incoming calls on GV (where you have the GV callerid set to the GV number).

I modified my script as follows:

[gv-inbound]
exten => s,1,Set(DID_EXTEN=\${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=\${CUT(DID_EXTEN,@,1)})
exten => s,n,GotoIf($["${BRIDGE}"="YES"]?:normalcall)
exten => s,n,GotoIf(\$[\${CALLERID(num)} = $gvnum]?:normalcall)
exten => s,n,NoCDR()
exten => s,n,Set(GLOBAL(BRIDGE)=NO) ; RESET global variable BRIDGE just before bridge occurs
exten => s,n,Bridge(\${DB_DELETE(gv_dialout/channel)})
exten => s,n,hangup
exten => s,n(normalcall),Goto(from-trunk,\${DID_EXTEN},1)

[gv-outbound]" >> /etc/asterisk/extensions_custom.conf
exten => _X.,1,Wait(1)
exten => _X.,n,Set(ACCTNAME=$acctname)
exten => _X.,n,Set(ACCTPASS=$acctpass)
exten => _X.,n,Set(RINGBACK=$ringback)
exten => _X.,n,System(gvoice -e \${ACCTNAME} -p \${ACCTPASS} call \${EXTEN} \${RINGBACK})
exten => _X.,n,Set(DB(gv_dialout/channel)=\${CHANNEL})
exten => _X.,n,Set(GLOBAL(BRIDGE)=YES
exten => _X.,n,Wait(10) ; Changed this from 5 to 10 to allow more time while waiting for incoming gv call
exten => _X.,n,Set(GLOBAL(BRIDGE)=NO ; RESET global variable BRIDGE in the event of timeout
exten => _X.,n,Hangup
exten => _X.,h,Set(GLOBAL(BRIDGE)=NO ; RESET global variable BRIDGE in the event of hangup

From my previous post, now the Google voice dial-out works, finally.... I end up installing the trixbox 2.8, did NO updates whatsoever (only for FreePBX) and then just install the pygooglevoice and its needs.

I end up using the method that "Ozborg" described on 08/07/2009 and end up using Gizmo5 as RINGBACK, then redirecting from my GV # to to my IPKall, since the Gizmo5 CID is set in the incoming route to use the gv-inbound. I tried what is on "dubkin" post on 10/14/2009t but it did not work for me...

the other methods for the "extensions_custom.conf" worked as well but in some times with issues (calls get disconnected while dialing-out) I notice that this methods place the call, hang-up then ring back to the extension, so is not like a live session...

I have just one (1) question, when I dial-out using google from one of the extensions I can't not dial-out from other extension at the same time, I get the "all circuits are busy", does this means that using this lines for "extensions_custom.conf" I can place/dial-out only one (1) call at the same time?, so if someone already dial-out then others can't since the "line" is busy?

Thanks.

Hi again,
It seems that the problem was that pygooglevoice needs to deploy a Python .egg package - simplejson and by default, in my centos 5.2 based PIAF it tried to do it in /root/.python-eggs/simplejson-2.0.9-py2.4-linux-i686.egg-tmp/simplejson/_speedups.so
Of course, /root/.python-eggs is not writable by the user asterisk - which is the user who the asterisk program is running as.

Adding this line after the gvoice System call in [gv-outbound] helped me find this issue:

exten => _X.,n,System(gvoice -b -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK})
exten => _X.,n,Verbose(0,$[SYSTEMSTATUS_is_${SYSTEMSTATUS}])

(it shows this line in /var/log/asterisk/full:
[2009-11-09 12:21:17] VERBOSE[8299] app_verbose.c: SYSTEMSTATUS_is_APPERROR

Also, stopping the asterisk and safe_asterisk processes and capturing the main console with:

killall safe_asterisk
killall asterisk
/usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c

allows you to capture the problem - when you try to originate a call from a local extension by dialing 48YYYYYYYYYY, you see this:

  File "/usr/lib/python2.4/site-packages/pkg_resources.py", line 846, in extraction_error
    raise err
pkg_resources.ExtractionError: Can't extract file(s) to egg cache

The following error occurred while trying to extract file(s) to the Python egg
cache:

  [Errno 13] Permission denied: '/root/.python-eggs'

The Python egg cache directory is currently set to:

  /root/.python-eggs

Perhaps your account does not have write access to this directory?  You can
change the cache directory by setting the PYTHON_EGG_CACHE environment
variable to point to an accessible directory.

The solution that worked for me was to change the /usr/bin/gvoice script, by adding 2 lines (import os and os.environ):

#!/usr/bin/python
# EASY-INSTALL-SCRIPT: 'pygooglevoice==0.4','gvoice'
__requires__ = 'pygooglevoice==0.4'
import os
os.environ['PYTHON_EGG_CACHE'] = '/var/lib/asterisk/.python-eggs'
import pkg_resources
pkg_resources.run_script('pygooglevoice==0.4', 'gvoice')

Hope this will help others that try to set up gvoice - Asterisk integration.
Raz

I am running Trixbox 2.8.0 CE and I had conference rooms setup fine. when a user accesses the conference bridge, it asks for the password and when i enter it hangs up.

Thanks and Regards
Ravi

Ok, I have mine setup with a IAX2 trunk from ipkall. The annoying thing is, all unknown numbers seem to come from 2062040232, which messes with the way my script was working. (I just started getting that from my Google voice number today too.... So I rewrote the gv-inbound and gv-outbound. I only use the ipkall did for outgoing, as the Caller ID has been crazy lately. all Google voice numbers seem to loose their caller ID. my cell comes in fine... Anyways, I have my ipkall go strait to hold as long as the 2 caller id's of either my Google Voice number, or 2062040232. So here is what I did. I setup the stuff like the guide, but instead of directing the DID of ipkall to the gv-inbound, I did the two caller id numbers with the DID too (Otherwise the DID of 9696 wont work, and you will have a loop on your hands.... Since there are 2 different ones, It creates a global variable that directs whether or not I am placing a call. If I didn't place the call, then it sends the call out as being from trunk 9696, then you set how you want to route that. I basically put it the same place as my ipkall DID. I call it with another google voice number, and it routes right where it is supposed to.

Here is my modified code. By the way, obeliks, thank you soo much for the "bash -c 'HOME=~asterisk && gvoice..." I was pulling my hair out trying to get it to work right without that. If you could share how you modified /var/lib/asterisk/.gvoice, that would be awesome!

[gv-outbound]
exten => _X.,1,Set(DB(gv_dialout/channel)=${CHANNEL})
exten => _X.,n,Set(GLOBAL(GOOGDIALED)=1)
exten => _X.,n,System(bash -c 'HOME=~asterisk && gvoice -e YOUR_GOOGLE _VOICE_EMAIL -p YOUR_PASSWORD call ${EXTEN} YOUR_RINGBACK')
exten => _X.,n,Wait(10)
exten => _X.,n,Set(GLOBAL(GOOGDIALED)=0)
exten => _X.,n,Hangup

[gv-inbound]
exten => s,1,GotoIf($[${GOOGDIALED} = 1]?:normalcall)"
exten => s,n,Set(GLOBAL(GOOGDIALED)=0)
exten => s,n,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,NoCDR()
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})
exten => s,n(normalcall),Goto(from-trunk,9696,1)

As you can see, I have it set the GOOGDIALED to 1 if it calls out, and if it fails to connect, then it sets it back to 0. I then have my CallWithUs account as the backup trunk. At .99 Cents per minute, it is a good deal. But free is even better :) Then on the inbound, if it isn't 1, it jumps to the normal call line, and that changed the trunk to 9696, and it routes the call from there. if it is one, the next line changes it to 0, and then it bridges the call.

A side note, I have tried to do more that one call on Google voice at once, and it fails. The Google voice doesn't initiate another call to my ipkall. So it isn't a replacement for a outgoing sip account, or sip trunk. Although, I just called it with all six of my lines, and they are all on hold now. Then I used my iSip on my iphone, and called the AT&T number going through google voice, and it worked. So I guess this means that you can get a cool DID from google voice, and have it go to your ipkall DID, and be set.... Although callerid from Google voice has been 2062040232 for all of these calls since around noonish eastern today. since all of these extensions are using ulaw, the processor use is on 2.3% on a P4 2.4 with only 512 megs of ram.

Besides the caller id issue, Ipkall is awesome!. I called it with all 6 of my lines at my office, and my cellphone. I wonder what the mas simultaneous lines is. Also, I know you aren't supposed to send faxes through voip, but I have been experimenting with it, and getting very good results. I am even on the east coast.

I even did this: I have a cheap HP all in one scanner-fax-inkjet printer, hooked up to a SPA-2102-SF hooked to my work network (Going through 2 100mbit switches) and connecting to my trixbox. I then have my trixbox setup with 2 ethernet cards. One is connected directly to my cablemodem (Time Warner businness 2mb up 7mb down with 5 static ip's) then I have my work router hooked to that, and a client network, and a guest wifi (WPA protected) hooked to that. Setup the trixbox because we are going to hook up our phone system we had at our other building that had 2 T1's run to it there, but here it would of cost too much to get them here, and we also scaled way down in the move too. Now we are going to need the capacity again, and to save lots of $$, I am going voip :) So I will be getting a Dual T1 card soon to play with.... Probably a month. When I do, I will also get a Netgear GS108T QOS switch, and set the trixbox as the highest priority, then my office net on normal, and the clients on low. I do have a gigabit card in my trizbox, but this switch isn't much more than the 100mb version.

Ok, Got sidetracked.... So after a grueling night lastnight, I got Asterfax setup to be able to send and receive faxes from any one of my SIP & IAx2 accounts. IT uses the fax detection pretty well. Granted I only tested it with my fax machine I have at my office that is connected to a POTS line from AT&T. It received faxes from my Google voice-ipkall combo nicely. It also sent just fine to my ipkall DID, and even our old companies number that is forwarded to me sipgate DID. (That just goes to a voicemail saying that the company is out of business, and where to sent mail, and to leave a message type thing.) Today I brought in a fresh SPA2102, and configured both lines. 1 - the all in one fax, and 2, I hooked up to our credit card machine. (Not supposed to work from what I hear) and it worked! Then I did the ultimate test. I sent a fax using google voice, through ipkall to call my AT&T DID that is forwarded to sipgate. and it worked!!!!!!! Got the fax delivered to my Email, and it looks perfect :)

So when our contract is up with AT&T for our business lines, we are going to cut them off. It is kind of sad to think of the 100 pair of copper telephone lines that come into this building, and only 2 will be used. (Someone renting space downstairs has one, and there is another that has a DSL that is the open WIFI for people that visit. the client WPA's one it the Plus network that the privileged clients get access too.)

So anyways, I have been shopping for a good SIP trunk that does IAX2. Some of these places don't seem too much cheaper that a real T1. I did find Unified Vox that looks real promising, but haven found really anything about them on Google. But $39.95 a month with unlimited inbound, unlimited concurrent calls, 2000 outbound, and $1 for ever additional DID sounds like a super deal to me. Granted after the 2000 minutes it is 2.9 cents per minute, that is where callwithus can come in at .99 cents per minute. The beauty of trixbox. You can mix up the trunks however you can imagine.

And to think it was only 1 week ago today that I discovered trixbox. I had been playing with sipsorcery for google voice before. but now it only goes down when I am messing with the trixbox :) My trixbox also seems more NAT friendly. With sipsorcery, I could only use my iphone through AT&T with gizmo5. Not sipgate, or ipcall, or even ipcomms. Anything in my trunks and enums workes perfectly through my iphone now. And to think, I am not even a Linux guy. Im a mac Guy. But hey, I am very familiar with the shell environment on it, and I have been using DD-wrt on my routers for years :)

I've been trying all night to get this to work, but with no dice. I keep getting an 'all circuits are busy' message when dialing out. Here's my extensions_custom.conf:

[gv-inbound]
exten => s,1,Set(DID_EXTEN=${SIP_HEADER(To):5})
exten => s,n,Set(DID_EXTEN=${CUT(DID_EXTEN,@,1)})
exten => s,n,GotoIf($[${CALLERID(num)} = 207956xxxx]?:normalcall)
exten => s,n,NoCDR()
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})
exten => s,n,(normalcall),Goto(from-trunk,123456,1)

[gv-outbound]
exten => _X.,1,Wait(1)
exten => _X.,n,Set(ACCTNAME=xxxxx)
exten => _X.,n,Set(ACCTPASS=xxxxx)
exten => _X.,n,Set(RINGBACK=747474xxxx)
exten => _X.,n,System(gvoice -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK})
exten => _X.,n,Set(DB(gv_dialout/channel)=${CHANNEL})
exten => _X.,n,Set(GLOBAL(BRIDGE)=YES)
exten => _X.,n,Wait(10)
exten => _X.,n,Set(GLOBAL(BRIDGE)=NO)
exten => _X.,n,Hangup
exten => _X.,n,Set(GLOBAL(BRIDGE)=NO)

Okay, so I was able run the gvoice command from the command line and I got a call (needed that 7!). However, now when calling from the PBX I get two 'all circuits are busy now' messages immediately.

I've adapted your earlier config a bit, which now looks like this:

[gv-inbound]
exten => s,n,GotoIf($[GROUP_COUNT(gvbridge) = 1]?:normalcall) ;
exten => s,n,GotoIf($["${CALLERID(num)}" = "+1207956xxxx"]?:normalcall) ; this is where you put your Google Voice number
exten => s,n,Set(GROUP()=gvbridge) ; GROUP_COUNT(bridge) should be 2 now
exten => s,n,NoCDR() ; remove callback CDR
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)}, p)
exten => s,n,System(bash -c 'HOME=~asterisk && gvoice --batch cancel') ; mostly in case local hanged up
exten => s,n,Hangup()
exten => s,n(normalcall),Goto(from-trunk,123456,1); will use an inbound route for this DID

[gv-outbound]
exten => _X.,n,Set(ACCTNAME=xxxx)
exten => _X.,n,Set(ACCTPASS=xxxx)
exten => _X.,n,Set(RINGBACK=747474xxxx)
exten => _X.,n,Set(DB(gv_dialout/channel)=${CHANNEL}) ; inbound needs this
exten => _X.,n,Set(GROUP()=gvbridge);
exten => _X.,n,System(bash -c 'HOME=~asterisk && gvoice --batch -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK} 7');
exten => _X.,n,Wait(5) ; 5 secs should be plenty or you need better trunks
exten => _X.,n,Hangup()

I am able to receive incoming calls sent to my GV number, so that part of the config does seem to be working properly.

@sjohnson

Does ls -la /var/lib/asterisk show any files owned by root ?
If so, chown -R asterisk /var/lib/asterisk

you should also less -R /var/log/asterisk/full for clues

One more problem in your dialplan, you need to start numbering priorities from 1, not from n.

Do dialplan show gv-outbound to make sure it starts from 1.

Btw, why, oh why, you simply did not copy my recipe ???

Hi everyone,

After parsing through the instructions above and following NerdVittles' guide I can't seem to get this working properly. I am using SIPGate and Google Voice on TB 2.8.0.1. I call a long distance number and get the message "Please wait while I connect your call." I then hear "75" and some music on hold. After that I get a fast busy signal no matter what number I dial. When I dial my Google Voice number from any land line or cell I have it forwarded to my SIPGate account which then forwards to an extension in TrixBox.

Here's what my logs say:

[Jun 2 10:55:49] VERBOSE[2946] logger.c: == Using SIP RTP TOS bits 184
[Jun 2 10:55:49] VERBOSE[2946] logger.c: == Using SIP RTP CoS mark 5
[Jun 2 10:55:49] VERBOSE[2946] logger.c: == Using SIP VRTP TOS bits 136
[Jun 2 10:55:49] VERBOSE[2946] logger.c: == Using SIP VRTP CoS mark 6
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [NumberI'mDialing@from-internal:1] Macro("SIP/599-b7cf1cf0", "user-callerid,SKIPTTL,") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:1] Set("SIP/599-b7cf1cf0", "AMPUSER=599") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:2] GotoIf("SIP/599-b7cf1cf0", "0?report") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:3] ExecIf("SIP/599-b7cf1cf0", "1?Set(REALCALLERIDNUM=599)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:4] Set("SIP/599-b7cf1cf0", "AMPUSER=599") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:5] Set("SIP/599-b7cf1cf0", "AMPUSERCIDNAME=House Phones") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:6] GotoIf("SIP/599-b7cf1cf0", "0?report") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/599-b7cf1cf0", "AMPUSERCID=599") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:8] Set("SIP/599-b7cf1cf0", "CALLERID(all)="House Phones" ") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:9] ExecIf("SIP/599-b7cf1cf0", "0?Set(CHANNEL(language)=)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:10] GotoIf("SIP/599-b7cf1cf0", "1?continue") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-user-callerid,s,19)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-user-callerid:19] NoOp("SIP/599-b7cf1cf0", "Using CallerID "House Phones" ") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [NumberI'mDialing@from-internal:2] Set("SIP/599-b7cf1cf0", "_NODEST=") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [NumberI'mDialing@from-internal:3] Macro("SIP/599-b7cf1cf0", "record-enable,599,OUT,") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/599-b7cf1cf0", "1?check") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-record-enable,s,4)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/599-b7cf1cf0", "recordingcheck,20100602-105549,1275494149.252") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Jun 2 10:55:49] VERBOSE[20174] logger.c: recordingcheck,20100602-105549,1275494149.252: Outbound recording enabled.
[Jun 2 10:55:49] VERBOSE[20174] logger.c: recordingcheck,20100602-105549,1275494149.252: CALLFILENAME=OUT599-20100602-105549-1275494149.252
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- AGI Script recordingcheck completed, returning 0
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-record-enable:999] MixMonitor("SIP/599-b7cf1cf0", "OUT599-20100602-105549-1275494149.252.wav,,") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [NumberI'mDialing@from-internal:4] Macro("SIP/599-b7cf1cf0", "dialout-trunk,4,NumberI'mDialing,,") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:1] Set("SIP/599-b7cf1cf0", "DIAL_TRUNK=4") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/599-b7cf1cf0", "0?sub-pincheck,s,1") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/599-b7cf1cf0", "0?disabletrunk,1") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:4] Set("SIP/599-b7cf1cf0", "DIAL_NUMBER=NumberI'mDialing") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:5] Set("SIP/599-b7cf1cf0", "DIAL_TRUNK_OPTIONS=trwW") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:6] Set("SIP/599-b7cf1cf0", "OUTBOUND_GROUP=OUT_4") in new stack
[Jun 2 10:55:49] VERBOSE[20176] logger.c: == Begin MixMonitor Recording SIP/599-b7cf1cf0
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/599-b7cf1cf0", "1?nomax") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-dialout-trunk,s,9)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/599-b7cf1cf0", "0?skipoutcid") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:10] Set("SIP/599-b7cf1cf0", "DIAL_TRUNK_OPTIONS=wW") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:11] Macro("SIP/599-b7cf1cf0", "outbound-callerid,4") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/599-b7cf1cf0", "0?Set(CALLERPRES()=)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/599-b7cf1cf0", "0?Set(REALCALLERIDNUM=599)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/599-b7cf1cf0", "1?normcid") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-outbound-callerid,s,6)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:6] Set("SIP/599-b7cf1cf0", "USEROUTCID=") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:7] Set("SIP/599-b7cf1cf0", "EMERGENCYCID=") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:8] Set("SIP/599-b7cf1cf0", "TRUNKOUTCID=GoogleVoicePhoneNumber") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/599-b7cf1cf0", "1?trunkcid") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-outbound-callerid,s,12)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/599-b7cf1cf0", "1?Set(CALLERID(all)=GoogleVoicePhoneNumber)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/599-b7cf1cf0", "0?Set(CALLERID(all)=)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/599-b7cf1cf0", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/599-b7cf1cf0", "0?AGI(fixlocalprefix)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:13] Set("SIP/599-b7cf1cf0", "OUTNUM=NumberI'mDialing") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:14] Set("SIP/599-b7cf1cf0", "custom=AMP") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/599-b7cf1cf0", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^)wW)") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:16] Macro("SIP/599-b7cf1cf0", "dialout-trunk-predial-hook,") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/599-b7cf1cf0", "") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/599-b7cf1cf0", "0?bypass,1") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/599-b7cf1cf0", "1?customtrunk") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-dialout-trunk,s,21)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:21] Set("SIP/599-b7cf1cf0", "pre_num=AMP:local/") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:22] Set("SIP/599-b7cf1cf0", "the_num=OUTNUM") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:23] Set("SIP/599-b7cf1cf0", "post_num=@custom-gv") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/599-b7cf1cf0", "1?outnum:skipoutnum") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Goto (macro-dialout-trunk,s,25)
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:25] Set("SIP/599-b7cf1cf0", "the_num=NumberI'mDialing") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Executing [s@macro-dialout-trunk:26] Dial("SIP/599-b7cf1cf0", "local/NumberI'mDialing@custom-gv,300,wW") in new stack
[Jun 2 10:55:49] VERBOSE[20174] logger.c: -- Called NumberI'mDialing@custom-gv
[Jun 2 10:55:49] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:1] Wait("Local/NumberI'mDialing@custom-gv-54a5;2", "1") in new stack
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:2] Set("Local/NumberI'mDialing@custom-gv-54a5;2", "ACCTNAME=GoogleVoiceEmailAddress") in new stack
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:3] Set("Local/NumberI'mDialing@custom-gv-54a5;2", "ACCTPASS=GoogleVoicePassword") in new stack
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:4] Set("Local/NumberI'mDialing@custom-gv-54a5;2", "RINGBACK=PSTNPhoneNumber") in new stack
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:5] Set("Local/NumberI'mDialing@custom-gv-54a5;2", "CALLPARK=75") in new stack
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:6] Playback("Local/NumberI'mDialing@custom-gv-54a5;2", "pls-wait-connect-call") in new stack
[Jun 2 10:55:50] VERBOSE[20174] logger.c: -- Local/NumberI'mDialing@custom-gv-54a5;1 answered SIP/599-b7cf1cf0
[Jun 2 10:55:50] VERBOSE[20177] logger.c: -- Playing 'pls-wait-connect-call.ulaw' (language 'en')
[Jun 2 10:55:52] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:7] System("Local/NumberI'mDialing@custom-gv-54a5;2", "gvoice -e GoogleVoiceEmailAddress -p GoogleVoicePassword call NumberI'mDialing PSTNPhoneNumber") in new stack
[Jun 2 10:55:55] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:8] Set("Local/NumberI'mDialing@custom-gv-54a5;2", "PARKINGEXTEN=75") in new stack
[Jun 2 10:55:55] VERBOSE[20177] logger.c: -- Executing [NumberI'mDialing@custom-gv:9] Park("Local/NumberI'mDialing@custom-gv-54a5;2", "") in new stack
[Jun 2 10:55:56] VERBOSE[20177] logger.c: == Parked Local/NumberI'mDialing@custom-gv-54a5;2 on 75@parkedcalls in lot 0. Will timeout back to extension [custom-gv] s, 1 in 30 seconds
[Jun 2 10:55:56] VERBOSE[20177] logger.c: -- Added extension '75' priority 1 to parkedcalls (0x9591f20)
[Jun 2 10:55:56] VERBOSE[20177] logger.c: -- Playing 'digits/7.gsm' (language 'en')
[Jun 2 10:55:56] VERBOSE[20179] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Jun 2 10:55:56] VERBOSE[20179] logger.c: == Manager 'admin' logged off from 127.0.0.1
[Jun 2 10:55:57] VERBOSE[20177] logger.c: -- Playing 'digits/5.gsm' (language 'en')
[Jun 2 10:55:58] VERBOSE[20177] logger.c: -- Started music on hold, class 'default', on Local/NumberI'mDialing@custom-gv-54a5;2
[Jun 2 10:55:58] VERBOSE[20177] logger.c: == Spawn extension (custom-gv, s, 1) exited non-zero on 'Parked/Local/NumberI'mDialing@custom-gv-54a5;2'
[Jun 2 10:56:26] VERBOSE[2930] logger.c: -- Stopped music on hold on Local/NumberI'mDialing@custom-gv-54a5;2
[Jun 2 10:56:26] VERBOSE[2930] logger.c: -- Added extension 'SIP0599' priority 1 to park-dial (0x9621a80)
[Jun 2 10:56:26] VERBOSE[2930] logger.c: == Timeout for Local/NumberI'mDialing@custom-gv-54a5;2 parked on 75. Returning to park-dial,SIP0599,1
[Jun 2 10:56:26] VERBOSE[20180] logger.c: -- Executing [SIP0599@park-dial:1] Dial("Local/NumberI'mDialing@custom-gv-54a5;2", "SIP/599|30|t") in new stack
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Using SIP RTP TOS bits 184
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Using SIP RTP CoS mark 5
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Using SIP VRTP TOS bits 136
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Using SIP VRTP CoS mark 6
[Jun 2 10:56:26] WARNING[20180] chan_sip.c: No such host: 599|30|t
[Jun 2 10:56:26] WARNING[20180] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Everyone is busy/congested at this time (1:0/0/1)
[Jun 2 10:56:26] VERBOSE[20180] logger.c: -- Executing [SIP0599@park-dial:2] Goto("Local/NumberI'mDialing@custom-gv-54a5;2", "app-blackhole,hangup,1") in new stack
[Jun 2 10:56:26] VERBOSE[20180] logger.c: -- Goto (app-blackhole,hangup,1)
[Jun 2 10:56:26] VERBOSE[20180] logger.c: -- Executing [hangup@app-blackhole:1] NoOp("Local/NumberI'mDialing@custom-gv-54a5;2", "Blackhole Dest: Hangup") in new stack
[Jun 2 10:56:26] VERBOSE[20180] logger.c: -- Executing [hangup@app-blackhole:2] Hangup("Local/NumberI'mDialing@custom-gv-54a5;2", "") in new stack
[Jun 2 10:56:26] VERBOSE[20180] logger.c: == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'Local/NumberI'mDialing@custom-gv-54a5;2'
[Jun 2 10:56:26] VERBOSE[20181] logger.c: == Manager 'admin' logged on from 127.0.0.1
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Executing [h@macro-dialout-trunk:1] Macro("SIP/599-b7cf1cf0", "hangupcall,") in new stack
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Executing [s@macro-hangupcall:1] GotoIf("SIP/599-b7cf1cf0", "1?skiprg") in new stack
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Goto (macro-hangupcall,s,4)
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Executing [s@macro-hangupcall:4] GotoIf("SIP/599-b7cf1cf0", "1?skipblkvm") in new stack
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Goto (macro-hangupcall,s,7)
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Executing [s@macro-hangupcall:7] GotoIf("SIP/599-b7cf1cf0", "1?theend") in new stack
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Goto (macro-hangupcall,s,9)
[Jun 2 10:56:26] VERBOSE[20174] logger.c: -- Executing [s@macro-hangupcall:9] Hangup("SIP/599-b7cf1cf0", "") in new stack
[Jun 2 10:56:26] VERBOSE[20174] logger.c: == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/599-b7cf1cf0' in macro 'hangupcall'
[Jun 2 10:56:26] VERBOSE[20174] logger.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/599-b7cf1cf0'
[Jun 2 10:56:26] VERBOSE[20174] logger.c: == Spawn extension (macro-dialout-trunk, s, 26) exited non-zero on 'SIP/599-b7cf1cf0' in macro 'dialout-trunk'
[Jun 2 10:56:26] VERBOSE[20174] logger.c: == Spawn extension (from-internal, NumberI'mDialing, 4) exited non-zero on 'SIP/599-b7cf1cf0'
[Jun 2 10:56:26] VERBOSE[20176] logger.c: == End MixMonitor Recording SIP/599-b7cf1cf0
[Jun 2 10:56:26] VERBOSE[20181] logger.c: == Manager 'admin' logged off from 127.0.0.1


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