Google Voice Install instructions

[klingon]

This should work on both the PlugPBX and Dockstar since both run Debian. I set mine to use IPKALL as my GV ring back provider, so instructions below are tailored to IPKALL.  SIPGATE and others shld work as well but I have not tried it.  Pls post back your success/failure once you've tried it.  Have fun.....

NOTE: Following only works for Asterisk 1.6 as this uses the Bridge() function

Code:

INSTALL Python GV
=================
>> cd /root
>> apt-get -y install python-setuptools
>> apt-get -y install python-simplejson
>> wget http://bestof.nerdvittles.com/applications/gv/pygooglevoice.tgz
>> tar zxvf pygooglevoice.tgz
>> cd pygooglevoice
>> python setup.py install

Go to GV and Set up GV to ring your IPKALL or SIPGATE number

** Using example of IPKALL fwd to SIP address  - 123456@mywbsite.com
Copy the following code to /etc/asterisk/extensions_custom.conf

[gv-outbound]
exten => _X.,1,Wait(1)
exten => _X.,n,Set(ACCTNAME=yourGVaccountname)
exten => _X.,n,Set(ACCTPASS=yourGVpassword)
exten => _X.,n,Set(RINGBACK=your IPKALL or Sipgate ringback number) ; exactly as you see it including the "1" if needed
exten => _X.,n,System(gvoice -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK})
exten => _X.,n,Set(GROUP()=gvout)  
exten => _X.,n,Set(DB(gv_dialout/channel)=${CHANNEL})
exten => _X.,n,Wait(5)
exten => _X.,n,Hangup

[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($[ ${GROUP_COUNT(gvout)} = 1 ]?:normalcall)  
exten => s,1,NoCDR()
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})
exten => s,n(normalcall),Goto(from-trunk,${DID_EXTEN},1)  


Copy the following code to /etc/asterisk/extensions_override_freepbx.conf.
This is because "ALLOW_SIP_ANON" is set to No in std installs
(replace 123456 with your IPKALL fwd to SIP address (front part before the @))  Not needed for SIPGATE

[from-sip-external]
exten => 123456,1,Goto(from-trunk,${DID},1)
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
exten => s,1,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?checklang:noanonymous)
exten => s,n(checklang),GotoIf($["${SIPLANG}"!=""]?setlanguage:from-trunk,${DID},1)
exten => s,n(setlanguage),Set(CHANNEL(language)=${SIPLANG})
exten => s,n,Goto(from-trunk,${DID},1)
exten => s,n(noanonymous),Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup


Login to Freepbx
Create Custom Destinations
Custom Destination: gv-inbound,s,1
Description : gv-inbound

Create Custom Trunk
Trunk Description : GV-Outbound
Dial Rules : 1+NXXNXXXXXX
Custom Dial String : local/$OUTNUM$@gv-outbound

Create Inbound Route
Description : GV-Inbouond
DID number : 123456   (replace with your IPKALL fwd to SIP address (front part before the @) OR your SIPGATE DID)
Custom Destinations : gv-inbound
(NOTE : If you use SIPGATE, then create a regular SIP trunk to register to SIPGATE as well)

Create Outbound Route (I use 48 to signify GV calls)
Dial patterns :
48|1NXXNXXXXXX
48|NXXNXXXXXX
Trunk Sequence : choose local/$OUTNUM$@gv-outbound

Reload Freepbx and you're good to go!  To use, dial 48 followed by your regular number.  Takes around 5-10 secs to connect call.

[AL]

Hi,

Copy the following code to /etc/extensions_override_freepbx.conf

Is that supposed to be /etc/asterisk/extensions_override_freepbx.conf?

TIA

[klingon]

Oops. Yes. I'll make the change.

[AL]

Thank you very much, klingon.

I now have a "semi"-working setup with GV.  By "semi" I mean that I can dial through the phone attached to my Linksys PAP2 and have GV ring back on my (non-VoIP) landline #.

I've also tried to specify my IPKall # in both /etc/asterisk/extensions_custom.conf and /etc/asterisk/extensions_override_freepbx.conf instead of my (non-VoIP) landline #.  With that setting, whenever I dialed out, it'd pause for a while before it then hangs up (I think that's what it did, but not sure).  So, my out-dialing attempts didn't work with the IPKall # as the ringback #.

Are there other configurations or steps I may have missed (whether to specify the IPKall # elsewhere or anything else)?

Also (maybe helpful for my debugging), is the mechanism using "Bridge()" different than what I've read elsewhere about using "parking lots"?  I also recall reading about the "parking lots" mechanism being preferred even though there was a more recent method (don't remember if it's using "Bridge()" or something else).  I don't remember if I read those in Nerd Vittles articles or not.  Sorry, I've been reading around quite a bit lately that the associations with what I read on which sites had become very faded memories. 

[klingon]

AL - May be a stupid question but did you set up IPKALL as the fwd to number in GV?  Also look at the asterisk logs for clues on whats failing - /var/log/asterisk/full.  In order for this to work, I find that you need to make sure your firewall accepts SIP calls, i.e. calls which are not registered in Asterisk as a SIP trunk.  I tend to fwd ports 5060-5061 and whatever RTP ports you have under /etc/asterisk/rtp.conf.  Try this and see if it works.

I chose Bridge() instead of Park as I find its a better feature.  With Park, it always makes the park extension announcement for every call. Either way works.  I believe Nerdvittles chose Park because Bridge is not avail in Asterisk 1.4.

[AL]

Thank you for your help, klingon.

I did have the IPKall as one of the numbers to ring on GV's settings.  I disabled the the other numbers so that the IPKall number is the only one that'll get rung for now.

Port forwarding is allowed for 5060-5061 and 10000-20000 for both TCP and UDP.

I didn't describe the end result completely correctly last night.  A few seconds after dialing out through my soft-phone (192.168.3.65) I'd hear "the number is not answering" and then a busy signal before it hangs up.  I tried with both my soft-phone and my PAP2 device.

I am guessing the "SIP/2.0 603 Declined" that's found in part of my "asterisk/full" log below might be where the problem is.

Here's the portion of the log file with asterisk started up with "/usr/sbin/asterisk -vvvvvv -g  -dddddd -c" and "sip set debug on" executed.  The DockStar's IP is 192.168.3.64 and I've masked out my GV # with 8005551212 there.    The "Generic SIP Device" extension I created is "222" and I've tried both "peer" and "friend" types.  "my-domain.us" is the placeholder for the pointer to my IP.  And, I do have my IPKall ID be set to "123456" for now.

Code:

[Aug  5 23:15:49] DEBUG[2396] devicestate.c: Changing state for SIP/66.54.140.46 - state 2 (In use)                                                                                                       
[Aug  5 23:15:49] DEBUG[2396] devicestate.c: device 'SIP/66.54.140.46' state '2'                                                                                                                         
[Aug  5 23:15:49] DEBUG[2401] app_queue.c: Device 'SIP/66.54.140.46' changed to state '2' (In use) but we don't care because they're not a member of any queue.                                           
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'Goto'                                                                                                                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-sip-external:1] Goto("SIP/66.54.140.46-00000008", "from-trunk,,1") in new stack                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Goto (from-trunk,123456,1)                                                                                                                                 
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'Set'                                                                                                                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-trunk:1] Set("SIP/66.54.140.46-00000008", "__FROM_DID=123456") in new stack                                                         
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Function result is 'GOOG V'                                                                                                                                         
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Expression result is '0'                                                                                                                                             
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Function result is '8005551212'                                                                                                                                     
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'ExecIf'                                                                                                                                                   
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-trunk:2] ExecIf("SIP/66.54.140.46-00000008", "0 ?Set(CALLERID(name)=8005551212)") in new stack                                       
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Function result is 'allowed_not_screened'                                                                                                                           
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'Set'                                                                                                                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-trunk:3] Set("SIP/66.54.140.46-00000008", "__CALLINGPRES_SV=allowed_not_screened") in new stack                                     
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'Set'                                                                                                                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-trunk:4] Set("SIP/66.54.140.46-00000008", "CALLERPRES()=allowed_not_screened") in new stack                                         
[Aug  5 23:15:49] DEBUG[2503] pbx.c: Launching 'Goto'                                                                                                                                                     
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Executing [123456@from-trunk:5] Goto("SIP/66.54.140.46-00000008", "gv-inbound,s,1") in new stack                                                           
[Aug  5 23:15:49] VERBOSE[2503] pbx.c:     -- Goto (gv-inbound,s,1)                                                                                                                                       
[Aug  5 23:15:51] VERBOSE[2414] chan_sip.c:                                                                                                                                                               
<--- SIP read from UDP:192.168.3.65:53593 --->                                                                                                                                                           
                                                                                                                                                                                                         
                                                                                                                                                                                                         
<------------->                                                                                                                                                                                           
[Aug  5 23:15:51] DEBUG[2414] chan_sip.c:  Header  0 [  0]:                                                                                                                                               
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Function result is '<sip:123456@my-domain.us>'                                                                                                                       
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Launching 'Set'                                                                                                                                                     
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Executing [s@gv-inbound:1] Set("SIP/66.54.140.46-00000008", "DID_EXTEN=123456@my-domain.us>") in new stack                                                 
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Function result is '123456'                                                                                                                                         
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Launching 'Set'                                                                                                                                                     
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Executing [s@gv-inbound:2] Set("SIP/66.54.140.46-00000008", "DID_EXTEN=123456") in new stack                                                               
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Function result is '1'                                                                                                                                               
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Expression result is '1'                                                                                                                                             
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Launching 'GotoIf'                                                                                                                                                   
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Executing [s@gv-inbound:3] GotoIf("SIP/66.54.140.46-00000008", "1?:normalcall") in new stack                                                               
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Not taking any branch                                                                                                                                               
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Auto fallthrough, channel 'SIP/66.54.140.46-00000008' status is 'UNKNOWN'                                                                                   
[Aug  5 23:15:51] DEBUG[2503] channel.c: Soft-Hanging up channel 'SIP/66.54.140.46-00000008'                                                                                                             
[Aug  5 23:15:51] DEBUG[2503] channel.c: Hanging up channel 'SIP/66.54.140.46-00000008'                                                                                                                   
[Aug  5 23:15:51] DEBUG[2503] chan_sip.c: Hangup call SIP/66.54.140.46-00000008, SIP callid 41bf790b23e80d2e7e45ce990a124a27@66.54.140.46                                                                 
[Aug  5 23:15:51] DEBUG[2503] chan_sip.c: Hanging up channel in state Ring (not UP)                                                                                                                       
[Aug  5 23:15:51] VERBOSE[2503] chan_sip.c: Scheduling destruction of SIP dialog '41bf790b23e80d2e7e45ce990a124a27@66.54.140.46' in 32000 ms (Method: INVITE)                                             
[Aug  5 23:15:51] VERBOSE[2503] chan_sip.c:                                                                                                                                                               
<--- Reliably Transmitting (NAT) to 66.54.140.46:5060 --->                                                                                                                                               
SIP/2.0 603 Declined                                                                                                                                                                                     
Via: SIP/2.0/UDP 66.54.140.46:5060;branch=z9hG4bK7fbfd562;received=66.54.140.46;rport=5060                                                                                                               
From: "GOOG V" <sip:8005551212@66.54.140.46>;tag=as63d8d11f                                                                                                                                               
To: <sip:123456@my-domain.us>;tag=as1be6e0b8                                                                                                                                                             
Call-ID: 41bf790b23e80d2e7e45ce990a124a27@66.54.140.46                                                                                                                                                   
CSeq: 102 INVITE                                                                                                                                                                                         
Server: Asterisk PBX 1.6.2.6                                                                                                                                                                             
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO                                                                                                                                 
Supported: replaces, timer                                                                                                                                                                               
Content-Length: 0

Earlier in the log, I also saw an instance of "SIP/2.0 401 Unauthorized" and am not sure if that's expected or not (as the handshaking seemed to continue).  Attached is the more complete log for the part when the soft-phone registers itself and then my attempt to dial 4818007055754.

Maybe my port forwarding isn't done correctly or there are other missing steps/configurations, still, in my setup?  I've spent quite a bit of time on this and not sure if the end of the tunnel is near or not.  I'd very much appreciate pointers on what else I should look for as well as other debugging techniques.

[klingon]

I believe your firewall is correctly set up as I see gv-inbound routine being called.  But just to make sure, try calling your IPKALL number and see if your phone rings. 

Your call seems to be failing at this point :
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Executing [s@gv-inbound:3] GotoIf("SIP/66.54.140.46-00000008", "1?:normalcall") in new stack
[Aug  5 23:15:51] DEBUG[2503] pbx.c: Not taking any branch
[Aug  5 23:15:51] VERBOSE[2503] pbx.c:     -- Auto fallthrough, channel 'SIP/66.54.140.46-00000008' status is 'UNKNOWN'
[Aug  5 23:15:51] DEBUG[2503] channel.c: Soft-Hanging up channel 'SIP/66.54.140.46-00000008'
[Aug  5 23:15:51] DEBUG[2503] channel.c: Hanging up channel 'SIP/66.54.140.46-00000008'

Try this - comment out the first 3 lines in gv-inbound and set the 4th-5th lines as such.  This will force the call to be bridged and see if it works.

exten => s,1,NoCDR()
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})

[AL]

Thank you again for your help, klingon.

I did as you suggested, to test the incoming handling first and found that, somehow, the following from gv-inbound would cause an infinite recursion for me.

Code:

exten => s,n(normalcall),Goto(from-trunk,${DID_EXTEN},1)

I also saw occurrences of "No such label 'normalcall' in extension 's' in context 'gv-inbound'" and "Priority 'normalcall' must be a number > 0, or valid label" in the log.

I looked around at other examples on the net and found one where I needed to create another "Inbound Route" which I gave the "DID Number" of "696969" and "Set Destination" to my "222" Asterisk extension.  I'm not sure if it's necessary, but I also modified the priority for the "NoCDR" line.  So, the resulting modified "gv-inbound" looks like this now:

Code:

[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($[ ${GROUP_COUNT(gvout)} = 1 ]?:normalcall)
exten => s,n,NoCDR() ; I changed the priority of "1" to "n" here.
exten => s,n,Bridge(${DB_DELETE(gv_dialout/channel)})
exten => s,n(normalcall),Goto(from-trunk,696969,1) ; I replaced the occurrence of "${DID_EXTEN}" with "696969" here to associate with the second "Inbound Route" I mentioned above.

With the above, my tests to call both my IPKall number and my GV number (with IPKall as the only number to forward/ring) finally came through fine.

I am, however, still having problem with dialing out.  As before, I tested with both a soft-phone and my PAP2 device.  As a test, I'd call my landline number by dialing out through the soft-phone/PAP2 and can see Asterisk dialing out just fine.  My landline phone would then ring but I would just hear nothing/silence when I picked up my landline phone.  There was also no ring or any other sound coming out of the soft-phone/PAP2 (not even any error message) before Asterisk would eventually hang up the call.

I'd appreciate other suggestions to try to figure this out.  I feel like it's getting closer, but really don't know for sure.

If you can share the relevant part(s) of the Asterisk "full" log file that shows the debugging info for the bridging process, I can try to compare with what I'm seeing as that may help me to figure out what else I may be mis-configuring.  Please also let me know any type of verbosity and/or debug settings used when the log file was generated.

I'm attaching the log file in case you'd easily spot what might be the culprit.  I've looked at it over and over and don't even know where things start going wrong.  The one thing that caught my attention was the "ZOMBIE" occurrences but maybe that's normal/expected.

I'm also attaching the configuration files in case you see something I missed in following your configuration instructions (I hope not).  I've masked out the real credentials and IPKall number by replacing them with their corresponding logical names.

TIA

[klingon]

My call log :

Code:

[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Goto (macro-dialout-trunk,s,22)
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:22] Set("SIP/200-00000000", "pre_num=AMP:local/") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:23] Set("SIP/200-00000000", "the_num=OUTNUM") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:24] Set("SIP/200-00000000", "post_num=@gv-outbound") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/200-00000000", "1?outnum:skipoutnum") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Goto (macro-dialout-trunk,s,26)
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:26] Set("SIP/200-00000000", "the_num=19085551212") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] pbx.c:     -- Executing [s@macro-dialout-trunk:27] Dial("SIP/200-00000000", "local/19085551212@gv-outbound,300,TW") in new stack
[2010-08-06 08:57:23] VERBOSE[9509] app_dial.c:     -- Called 19085551212@gv-outbound
[2010-08-06 08:57:23] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:1] Wait("Local/19085551212@gv-outbound-9601;2", "1") in new stack
[2010-08-06 08:57:24] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:2] Set("Local/19085551212@gv-outbound-9601;2", "ACCTNAME=abc@abc.com") in new stack
[2010-08-06 08:57:24] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:3] Set("Local/19085551212@gv-outbound-9601;2", "ACCTPASS=yourpassword") in new stack
[2010-08-06 08:57:24] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:4] Set("Local/19085551212@gv-outbound-9601;2", "RINGBACK=youripkalltelnbr") in new stack
[2010-08-06 08:57:24] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:5] System("Local/19085551212@gv-outbound-9601;2", "gvoice -e abc@abc.com -p yourpassword call 19085551212 youripkalltelnbr") in new stack
[2010-08-06 08:57:30] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:6] Set("Local/19085551212@gv-outbound-9601;2", "DB(gv_dialout/channel)=Local/19085551212@gv-outbound-9601;2") in new stack
[2010-08-06 08:57:30] VERBOSE[9511] pbx.c:     -- Executing [19085551212@gv-outbound:7] Wait("Local/19085551212@gv-outbound-9601;2", "5") in new stack
[2010-08-06 08:57:31] VERBOSE[7169] netsock.c:   == Using SIP RTP TOS bits 184
[2010-08-06 08:57:31] VERBOSE[7169] netsock.c:   == Using SIP RTP CoS mark 5
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-sip-external:1] Goto("SIP/66.54.140.50-00000001", "from-trunk,,1") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Goto (from-trunk,ipkallnbr,1)
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:1] Set("SIP/66.54.140.50-00000001", "__FROM_DID=ipkallnbr") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:2] Gosub("SIP/66.54.140.50-00000001", "app-blacklist-check,s,1") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [s@app-blacklist-check:1] GotoIf("SIP/66.54.140.50-00000001", "0?blacklisted") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [s@app-blacklist-check:2] Set("SIP/66.54.140.50-00000001", "CALLED_BLACKLIST=1") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [s@app-blacklist-check:3] Return("SIP/66.54.140.50-00000001", "") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:3] ExecIf("SIP/66.54.140.50-00000001", "0 ?Set(CALLERID(name)=yourGVnbr)") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:4] Set("SIP/66.54.140.50-00000001", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:5] Set("SIP/66.54.140.50-00000001", "CALLERPRES()=allowed_not_screened") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [ipkallnbr@from-trunk:6] Goto("SIP/66.54.140.50-00000001", "gv-inbound,s,1") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Goto (gv-inbound,s,1)
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [s@gv-inbound:1] NoCDR("SIP/66.54.140.50-00000001", "") in new stack
[2010-08-06 08:57:31] VERBOSE[9514] pbx.c:     -- Executing [s@gv-inbound:2] Bridge("SIP/66.54.140.50-00000001", "Local/19085551212@gv-outbound-9601;2") in new stack
[2010-08-06 08:57:31] VERBOSE[9509] app_dial.c:     -- Local/19085551212@gv-outbound-9601;1 answered SIP/200-00000000
[2010-08-06 08:57:32] VERBOSE[9511] pbx.c:   == Spawn extension (gv-outbound, 19085551212, 7) exited non-zero on 'Bridge/Local/19085551212@gv-outbound-9601;2<ZOMBIE>'
[2010-08-06 08:58:28] VERBOSE[9509] pbx.c:     -- Executing [h@macro-dialout-trunk:1] Macro("SIP/200-00000000", "hangupcall,") in new stack
[2010-08-06 08:58:28] VERBOSE[9509] pbx.c:     -- Executing [s@macro-hangupcall:1] GotoIf("SIP/200-00000000", "1?skiprg") in new stack
[2010-08-06 08:58:28] VERBOSE[9509] pbx.c:     -- Goto (macro-hangupcall,s,4)
[2010-08-06 08:58:28] VERBOSE[9509] pbx.c:     -- Executing [s@macro-hangupcall:4] GotoIf("SIP/200-00000000", "1?skipblkvm") in new stack

[AL]

Thank you, yet again, klingon.

I'll compare and see if I can figure out what's wrong/missing from my setup and will report back if I find anything useful.

[klingon]

You're welcome.  You may want to just delete and redo your GV setup, incl the .conf files settings and see if that works. It happened to me once and I did nothing but just recreated the trunk and it worked.  Maybe something in Freepbx screwed up when you first set it up, who knows.  Just delete pygooglevoice from /root folder and start again and follow the instructions exactly, dont deviate for now. I know the steps work as I've done it more than once.  See if this may help.

[twinclouds]

Eventually, I made GV-Sipgate dial-out working on my Dockstar.  It looks very stable right now.
After install FreePBX, the steps I followed was that given in http://www.rolandli.com/2010/01/14/comprehensive-guide-asterisk-freepbx-google-voice-on-dns-323/5 by Roland Li (Part 5).  His procedure was adapted from elsewhere but I feel his descriptions were easier to follow.  The only change need to be made was one in his step 11.  Once you get install-gv-new, you need to edit it, search for yum and replace "yum" to "apt-get".
After completed the installation, it appears Asterisk was working on my Dockstar but had very long delay when making calls to make the calls unreliable.  The CPU loading was very low when this happened so that is not the cause.  I found that the problem was because dns searching took too long.    (One test you can try is to ping google.com (or any known external IP address).  What I found was that the ping delay itself was not too bad (~20ms).  However, it would take a few seconds between the pings.  If this is the case for you, you would have the same problem as I did.)  To fix this problem, you should change the ip address setting to static by editing /etc/network/interfaces and add one or two public DNS server addresses following word "nameserver" (e.g. "nameserver 208.67.222.222" and "nameserver 8.8.8.8".) at the beginning of the file /etc/resolv.conf.   You can repeat the pinging experiment again and should find that there will be very little delays between the pings.  After doing that, The GV-Sipgate dialing out on Dockstar works very reliably.  (Actually, I would suggest to do the IP and DNS setting changes immediately after you installed Debian.  This will make the whole installation processes much faster.)
I tried this on both Asterisk/FreePBX 1.4/2.6 and 1.6/2.7 installations.  Both works.  However, the 1.4/2.6 seemed consuming less memory than 1.6/2.7 and had fewer problems in installation, e.g., it does not have the high CPU loading problem.  I am using 1/4/2.6 installation right now.
Hope this helps.