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Author Topic: How to install Asterisk with GV dialout in 20 minutes  (Read 7068 times)
twinclouds
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« on: October 16, 2010, 08:45:49 PM »

Since I installed FreePBX on the Dockstar for home use a few weeks ago, it has been working flawlessly to date.  I mainly use it for Google-Voice dialout for US and Canada calls.  I feel it is very desirable for home PBX installed on a dedicated computer and Dockstar is a perfect platform for such a purpose due to its low power consumption.  Connected to it are an SPA3102 voip adapter with a cordless phone system with three handsets and a couple of soft phones.

I had used it for voip services such as voip.ms.  However, right now, I found it is more convenient to use the SPA3102 to connect to multiple voip services (up to 5 total include my FreePBX server).  Thus, I am using the FreePBX exclusively for GV dialout.  

Considering the limited RAM available in Dockstar and the functionalities that I required, I feel a full blown FreePBX maybe an overkill.  Thus I started to find out what is the minimum installation of Asterisk will serve the purposes that I need.

Since I am not really familiar with Asterisk programming, I relied on the information that I can find on the Web.  After a few weeks of frustration, I eventually succeeded.  Considering the troubles that I got into, I thought it might be useful for others to share my findings.  

Below, I provide the steps to implement such s minimum Asterisk implementation for GV-dialout.  The procedures are quite straightforward and can be completed in about 20 minutes.  Even though it is straightforward, I feel some knowledge about how to install debian Linux on Dockstar will be desirable.  It is also necessary to know how to set up SIP softphone and/or voip adapters and to register to your Asterisk PBX.

My set-up uses Google Voice (GV) for dial out and Sipgate for calling back.  I assume you have already signed for GV and Sipgate services.  The GV should be set up to use Sipgate as call destination.  If not, you should be able to find the information from the Web, such as the links cited in this how-to.  Once these are done, you are ready to roll on.

Below are steps:

1. Go to Jeff Doozan's website:
http://jeff.doozan.com/debian/install_lenny.htm.

2. Install Debian Lenny according to the instructions there.
(Note: I choose Jenny instead of Squeeze because it works fine for me and the installation process is much faster.)
The major steps are:

2.1 log into Dockstar through ssh using
Code:
Username: root
Password: stxadmin
2.2 plug in USB drive of at least 1GB and partion it:
Code:
fdisk /dev/sda
# Configure partition 1 as Linux (I recommend at least 700MB)
# Configure partition 2 as Linux Swap (256MB recommended)
2.3 Install Debian Lenny, as follows:
Code:
cd /tmp
wget http://jeff.doozan.com/debian/dockstar.debian-lenny.sh
chmod +x dockstar.debian-lenny.sh
./dockstar.debian-lenny.sh
2.4 Reboot into installed Debian Linux
2.5 login using:
Code:
username: root
password: root

3. Install asterisk as follows:
Code:
# cd /tmp
# apt-get update
# apt-get install asterisk

This will install asterisk1.4.  To test asterisk typing
Code:
# asterisk -rvvv
It should show the prompt:
Code:
debian*CLI>
type quit to exist.

4. Install python scripts for GV-dialout, such as:
Code:
# apt-get install python-setuptools
# easy_install simplejson
# wget http://bestof.nerdvittles.com/applications/gv/pygooglevoice.tgz
# tar zxvf pygooglevoice.tgz
# cd pygooglevoice
# python setup.py install
5. Set config files
Only modifications of three files are needed for call between extensions and call out through Google Voice/Sipgate, i.e., sip.conf, extensions.conf, and features.conf existed in directory /etc/asterisk/ after asterisk being installed.  The simplest form I compiled based on the information from Googling are given as follows.

Sip.conf:
Code:
; sip.conf

[general]
register => *sipgateID:*sipgatePW@sipgate.com/*sipgateID
; e.g.: *sipgateID = 9999999e0, *sipgatePW = AAAAA0

[sipgate]
type=peer
host=sipgate.com
outboundproxy=sipgate.com
insecure=invite
qualify=yes
dtmfmode=rfc2833
username=*sipgateID
defaultuser=*sipgateID
fromuser=*sipgateID
outboundproxy=proxy.live.sipgate.com
secret= *sipgatePW
context=sipgate_in
disallow=all
allow=ulaw

[101]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
defaultuser=101
secret=*secret1
callerid="*user1" <101>
mailbox=101

[102]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
defaultuser=102
secret=*secret2
callerid="*user2" <102>
mailbox=102

[103]
type=friend
host=dynamic
nat=yes
qualify=yes
context=sip
defaultuser=103
secret=*secret3
callerid="*user3" <103>
mailbox=103

Extensions.conf:
Code:
; extensions.conf

[general]

[globals]

[sip]
exten => _1XX,1,NoOp(Dialling phone ${EXTEN})
exten => _1XX,n,Dial(SIP/${EXTEN},,rotwh)
exten => _1XX,n,Hangup

exten => _X.,1,Log(NOTICE, Dialing out from ${CALLERID(all)} to ${EXTEN:1} through Google Voice)
exten => _XXXX.,n,Set(ACCTNAME=*mygv@gmail.com)
exten => _XXXX.,n,Set(ACCTPASS=*mygvPW)
exten => _XXXX.,n,Set(RINGBACK=*sipgatephone#)
exten => _XXXX.,n,Set(CALLPARK=75)
exten => _XXXX.,n,Playback(pls-wait-connect-call)
exten => _XXXX.,n,System(gvoice -e ${ACCTNAME} -p ${ACCTPASS} call ${EXTEN} ${RINGBACK}& )
exten => _XXXX.,n,Set(PARKINGEXTEN=${CALLPARK})
exten => _XXXX.,n,Park()

[sipgate_in]
exten => sipgateID,1,Log(NOTICE, Incoming call from ${CALLERID(all)})
exten => sipgateID,2,Wait(4)
exten => sipgateID,3,Set(CALLPARK=75)
exten => sipgateID,4,Set(GVNUM=*mygvphone#)
exten => sipgateID,5,Set(GVNUM1=*1mygvphone#) ; In some cases, a "1" could be added in front of the caller ID (your gv number)
exten => sipgateID,6,Log(NOTICE, CALLERID: ${CALLERID(number)})
exten => sipgateID,7,GotoIf($[$["${CALLERID(number)}"="${GVNUM}"]|$["${CALLERID(number)}"="${GVNUM1}"]]?10)
exten => sipgateID,8,Dial(SIP/102,20)
exten => sipgateID,9,Hangup()
exten => sipgateID,10,ParkedCall(${CALLPARK})


Features.conf:
Code:
; features.conf
[general]
parkext => 70
parkpos => 70-75
context => parkedcalls
findslot => first

Since these files are quite short, you can simply open existing corresponding files, delete the content and copy the content into them using your favorite editor, such as vi.  To be safe, you might want to make backup copies of the files came with Asterisk.

You will need to fill in your own Sipgate, GV and extensions specific information (those variables start with a *).

6. Get music on hold (moh) and other sound files (optional) from:
http://downloads.asterisk.org/pub/telephony/sounds/
I used the files in asterisk-moh-opsound-gsm-current.tar.gz by decompressing it and then put the files into debian:  /var/lib/asterisk/moh/ and the file pls-wait-connect-call.gsm from asterisk-extra-sounds-en-gsm-current.tar.gz into /usr/share/asterisk/sounds/ (I used a fat formated USB drive as the transfer media.)

7. Restart of Asterisk by type:
Code:
# /etc/init.d/asterisk restart
8. Setup your sip phone and/or your voip adapters, e.g., PAP2T or SPA3102, and register them with your asterisk.

9.  You are done now.  
You can call from one extension (I use 101, 102 and 103) to another and that extension will ring.  You can also dial a pstn phone from any of the extension.  If everything working correctly, you should hear the announcement of numbers "7" "5" and then the music (if you have the music on hold files installed).  After about 10-15 seconds, the phone on the other side will ring.

Disclaimer:

The procedures and codes provided are just for experiment.  Even though I have verified it works for the purpose, I cannot predict any possible outcomes.  In particular, no steps have been take to make the system secure.  Thus, I would like emphasize that you will be fully responsible for any consequences of its installation and utilization.

Acknowledgments:

The procedure of install Debian on Dockstar are based on what I found from: http://jeff.doozan.com/debian/install_lenny.htm and http://forums.plugpbx.org/index.php?board=10.0

The configuration files are based on the information that I found from: http://gebl.landq.org/23950793, http://kenny.barnt.us/?p=102, http://www.rolandli.com/2010/01/14/comprehensive-guide-asterisk-freepbx-google-voice-on-dns-323, and http://www.orderlyq.com/asteriskqueues.html.  A tutorial about using GV-dialout with FreePBX can be found in http://nerdvittles.com/?p=635  

All of the credits are due to the respective authors.  What I did was compiling them into a working version.

Hope you will enjoy this posting.  Please let me know if you have any comments and/or any issues encountered.
« Last Edit: October 21, 2010, 10:19:51 PM by twinclouds » Logged
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Posts: 426



« Reply #1 on: October 17, 2010, 08:10:29 AM »

Cool. I used to run Asterisk 1.2 on my Buffalo NAS (hacked with debian) and initially ran it without FreePBX to save on Memory usage.

That being said, I ended up scripting alot of functionality directly in Asterisk Dial-plans. So I dug up the source and pasted it here. You guys might find some of the calling features handy if you want to put a slimmer PBX in your DockStars. I used this with Unlimitel.ca, so if you are a customer, you can just populate the variables at the top Wink

Notice as well I had this set with both my wifes Cell phone, and My Cell #, so you can utilize the call escalation feature also scripted here. Perhaps this will inspire some DockStar Asterisk guys. It worked pretty well.

NOTE: This is using an IAX connection to my provider, not SIP ....But again, easy to modify!

Enjoy!

Code:

[general]

static=yes
writeprotect=no
autofallthrough=no ; Screws up my menus if its set to yes
clearglobalvars=no
priorityjumping=no


[globals]

;Unlimitel.ca Phone Service Account Variables
;********************************************
UNLNUM=XXXXXXXXX; Account Number with Unlimitel.ca Account
UNLPASS=YYYYYYY;   Password for account

;Cell Phone Numbers
;******************
GREGCELL=AAAAAAAA
CRYSTALCELL=BBBBBBBBB


[unlimitel-out]

exten => _NXXNXXXXXX,1,SetAccount(${UNLNUM})
exten => _NXXNXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _NXXNXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN})
exten => _NXXNXXXXXX,4,Congestion

exten => _1800NXXXXXX,1,SetAccount(${UNLNUM})
exten => _1800NXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _1800NXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN})
exten => _1800NXXXXXX,4,Congestion

exten => _1888NXXXXXX,1,SetAccount(${UNLNUM})
exten => _1888NXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _1888NXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN})
exten => _1888NXXXXXX,4,Congestion  
  
exten => _1877NXXXXXX,1,SetAccount(${UNLNUM})
exten => _1877NXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _1877NXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN})
exten => _1877NXXXXXX,4,Congestion
    
exten => _1866NXXXXXX,1,SetAccount(${UNLNUM})
exten => _1866NXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _1866NXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN})
exten => _1866NXXXXXX,4,Congestion
    
exten => _1NXXNXXXXXX,1,SetAccount(${UNLNUM})
exten => _1NXXNXXXXXX,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _1NXXNXXXXXX,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN:1})
exten => _1NXXNXXXXXX,4,Congestion

exten => _911,1,SetAccount(${UNLNUM})
exten => _911,2,Set(CALLERID(all)=Your CallerID Name Here <${UNLNUM}>)
exten => _911,3,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${EXTEN},30,r)
exten => _911,4,Playback(cannot-complete-network-error)


[default]
;Default Dialing Stuff Here


[local_phone]

;Internal Phones in our house use this Dial Plan...

include => parkedcalls
include => unlimitel-out


; **** Internal Features Below ****

; Dialling 666 on the handset connects to the voicemail box
exten => 666,1,VoiceMailMain(s3000)

; Test VoiceMail internally - Leave message
exten => 667,1,VoiceMail(u3000)

; Playback Monkeys screaming test
exten => 6665397,1,Playback(tt-monkeys)
exten => 6665397,2,Hangup

; Test MOH
exten => 111,1,Answer
exten => 111,2,SetMusicOnHold(default)
exten => 111,3,WaitMusicOnHold(240)
exten => 111,4,Hangup

;Random Voice Prompts
exten => 112,1,Answer
exten => 112,2,SetMusicOnHold(apeshit)
exten => 112,3,WaitMusicOnHold(240)
exten => 112,4,Hangup

; Test Call Block Message
exten => 5150,1,Answer
exten => 5150,2,Goto(incoming,black,1)

;Dial GXP-2000 Handset
exten => 3001,1,Dial(SIP/3001,20,rth)
exten => 3001,2,Hangup()

;Dial Cordless Phones
exten => 3000,1,Dial(SIP/3000,20,rth)
exten => 3000,2,Hangup()

;Dial Gregs Cell phone as an Extension
exten => 3002,1,SetCallerID(${UNLNUM})
exten => 3002,2,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${GREGCELL},30,rth)
exten => 3002,3,Hangup()

;Dial Crystals Cell phone as an Extension
exten => 3003,1,SetCallerID(${UNLNUM})
exten => 3003,2,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${CRYSTALCELL},30,rth)
exten => 3003,3,Hangup()

;Dial The Automated Operator - Have to write this code yet
exten => 0,1,Answer
exten => 0,2,Playback(unavailable)
exten => 0,3,Playback(goodbye)
exten => 0,4,Hangup()


;Dial All Phones

exten => 999,1,SetCallerID(${UNLNUM})
exten => 999,2,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${CRYSTALCELL}&IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${GREGCELL}&SIP/3000&SIP/3001,30,rth)
exten => 999,3,Hangup()

;Simulate Incoming call(s)
exten => 1234,1,Goto(ringphones,9,6)

;Caller ID / Caller Blacklist / Blocker
exten => *69,1,Answer
exten => *69,2,Wait(1)
exten => *69,3,Set(number=${DB(lastcallerid/number)})
exten => *69,4,GotoIf($["${number}"=""]?104,1)
exten => *69,5,Playback(privacy-last-caller-was)
exten => *69,6,SayDigits(${number})
exten => *69,7,Playback(press-5)
exten => *69,8,Playback(privacy-to-blacklist-last-caller)
exten => *69,9,Playback(or)
exten => *69,10,Playback(please-try-again-later)
exten => 5,1,DBput(blacklist/${number}=1)
exten => 5,2,Playback(privacy-blacklisted)
exten => 5,3,Wait,1
exten => 5,4,PlayBack(goodbye)
exten => 5,5,Hangup
exten => *69,104,Playback(unidentified-no-callback)
exten => *69,106,Hangup
exten => t,1,Playback(goodbye)                          ; No selection made in 30 seconds, hangup caller
exten => t,2,Hangup



[incoming]

;Route All calling sources into Main Routine

exten => 5551236666,1,Goto(s,1) ; Route Unlimitel.ca Calls into system, screening for spam  (your DID # here)


; All outside callers Come in via this dialplan. We send them to the internal phones after a short fake
; ringing delay. If someone presses the * key during that time, they get the backdoor menu to get voicemail
; remotely, as well as internal dialtone and other handy-dandy features.

exten => s,1,Answer                             ; Pickup New call
exten => s,2,LookupBlacklist()                  ; Delay 1 Second to allow for stuff
exten => s,3,GotoIf($["${LOOKUPBLSTATUS}" = "FOUND"]?black,1)  ;Caller is blacklisted, send 'em to hell
exten => s,4,GotoIf($["${CALLERIDNUM:0:9}" = "anonymous"]?8)   ;Anon Caller, Send to voicemail
exten => s,5,GotoIf($["${CALLERIDNUM:0:9}" = ""]?8)            ;Empty Caller-ID, Send to Voicemail
exten => s,6,Set(DB(lastcallerid/number)=${CALLERIDNUM})  ;Note calling number for usage with *69

exten => s,7,Goto(11)   ; Caller ID Valid, Pass them off for Lookup
;
exten => s,8,Zapateller()                ; Zap Telemarketing Autodiallers with no valid callerid
exten => s,9,Goto(ringphones,_1,1) ; Send the bad person right to voicemail, FUCK'em
;
exten => s,10,Playtones(ring)                    ; Fake ringing during backdoor menu 'window' of time
;
;exten => s,11,AGI(callerid_shell.agi|${CALLERIDNUM})
;exten => s,12,NoOp(AGI Returned ${lookupname})
;exten => s,13,Set(CALLERID(name)=${lookupname})
exten => s,11,NoOp()
exten => s,12,NoOp()
exten => s,13,NoOp()
;
exten => s,14,ResponseTimeout(1)                 ; Allow delay for Backdoor selection
exten => s,15,DigitTimeout(1)                    ; Limit Keypress limit to 1 sec
;
exten => *,1,Goto(backdoor,*,1)                 ; Jump to Backdoor menu!
;
exten => t,1,Goto(ringphones,9,1)               ; Jump to the Call Ringing Routine, no selectio made treat it as normal call
;
exten => i,1,Goto(ringphonest,9,1)              ; Invalid keypress, bypass the delay, and send call to phones

exten => black,1,Zapateller() ; This caller is blacklisted. Fuck them and disconnect.
exten => black,n,Playback(go-away1)
exten => black,n,Playback(go-away2)
exten => black,n,Playback(privacy-you-are-blacklisted)
exten => black,n,Playback(privacy-stop-calling-not-welcome)
exten => black,n,Playback(thank-you-cooperation)
exten => black,n,Hangup()

[ringphones]

; Pass the call inside the house to the home phones
; If nobody answers, offer to send the call to voicemail or our cell phones
; If no call is answered via the cell phones, have it terminate to voicemail

exten => 9,1,NoOp()
exten => 9,2,NoOp()
exten => 9,3,StopPlaytones
exten => 9,4,Dial(SIP/3000&SIP/3001,20,rth)          ; Pass the call to phones in house
exten => 9,5,ResponseTimeout(5)
exten => 9,6,Background(press-1)
exten => 9,7,Background(T-to-leave-msg)
exten => 9,8,Background(press-2)
exten => 9,9,Background(for)
exten => 9,10,Background(emergencycallfwd)
exten => 9,11,Background(press-3)
exten => 9,12,Background(to)
exten => 9,13,Background(ringphonesagain)
;
exten => _1,1,VoiceMail(u3000) ; They asked to leave a message, send them to the voicemail routine
exten => _1,2,Playback(goodbye)                  ; Say goodbye
exten => _1,3,Hangup                             ; Close circuit, end call. Doom!
;
exten => _2,1,Playback(pls-wait-connect-call)   ; They have asked to try our cell phones, tell them to hang on
exten => _2,2,Setvar(NewCaller=${CALLERIDNUM}) ; We'll pass on their callerid info when we call both our phones
exten => _2,3,SetCIDNum(${CALLERIDNUM})         ; Grab their Number to spoof unto our cells
exten => _2,4,SetCIDName(${CALLERIDNAME}) ; Grab their 'name' from the phone company to spoof to our cells
exten => _2,5,Dial(IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${GREGCELL}&IAX2/${UNLNUM}:${UNLPASS}@iax02.unlimitel.ca/${CRYSTALCELL},30,rth)
exten => _2,6,SetCIDNum(${NewCaller})
exten => _2,7,Playback(sorry)
exten => _2,8,Playback(number-not-answering)
exten => _2,9,Goto(9,5)                         ; Give up and send them to the main options menu again, we didn't answer our cells
;
exten => _3,1,Goto(ringphones,9,4) ; Ring the Home Phones once more...
;
exten => *,1,Goto(backdoor,*,1) ; Provide Backdoor menu option here as well...
;
exten => t,1,Goto(_1,1) ; No selection made, send the caller to Voicemail - screw 'em
;
exten => i,1,Playback(wrong-try-again-smarty)
exten => i,2,Goto(9,7) ; They pressed an invalid key, send them back to the menu
;


[backdoor]

;Menu accesable from outside world to users with password - Provides Voicemail/DISA etc

exten => *,1,StopPlaytones
exten => *,2,Authenticate(1234,a) ; Ask for access password    (yes the password is 1234, ensure you change this)
exten => *,3,Playback(pin-number-accepted) ; We're in
exten => *,4,Playback(main-menu) ; MAIN MENU prompts
exten => *,5,ResponseTimeout(30)
exten => *,6,Background(press-1)
exten => *,7,Background(to-collect-voicemail)
exten => *,8,Background(press-2)
exten => *,9,Background(for)
exten => *,10,Background(system)
;
exten => _1,1,VoiceMailMain(s3000) ; Link caller to voicemail system box
exten => _1,2,Goto(*,6)
;
exten => _2,1,DigitTimeout(5) ; Provide remote system dialtone access / DISA dialing
exten => _2,2,Playback(all-your-base)
exten => _2,3,DISA(no-password,local_phone)
;
exten => t,1,Playback(goodbye) ; No selection made in 30 seconds, hangup caller
exten => t,2,Hangup
;
exten => i,1,Goto(*,6) ; Repeate menu on bad selection





« Last Edit: October 17, 2010, 08:19:01 AM by PlugPBX Admin » Logged

-Greg
jet101
Newbie
*
Posts: 15


Email
« Reply #2 on: October 19, 2010, 06:21:48 PM »

Great write up on getting this basic install pkg running. I like running it on a 1 Gig thumb drive. I was able to get through this fairly easily. I added music on hold files and have that working, however, the only thing I haven't bee able to get working right is the sound files. I got the sound files and added them in /var/lib/asterisk/sounds, but it will not play the one you have for instance in

exten => _XXXX.,n,Playback(pls-wait-connect-call)

If I add the path to make it (/var/lib/asterisk/sounds/pls-wait-connect-call) it plays just fine. Anyone have any idea what's going on there? Here's the error I get in the Asterisk Log:

[Oct 20 03:11:44] WARNING[1833]: file.c:602 ast_openstream_full: File pls-wait-connect-call does not exist in any format
[Oct 20 03:11:44] WARNING[1833]: file.c:912 ast_streamfile: Unable to open pls-wait-connect-call (format 0x4 (ulaw)): No such file or directory
[Oct 20 03:11:44] WARNING[1833]: app_playback.c:439 playback_exec: ast_streamfile failed on SIP/101-0018d990 for pls-wait-connect-call

Great job by the way. Thanks!
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twinclouds
Jr. Member
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Posts: 99


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« Reply #3 on: October 19, 2010, 06:31:33 PM »

I am glad you like it.
I didn't get the "pls ..." work either.  I am interested to know the solution also.
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jet101
Newbie
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Posts: 15


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« Reply #4 on: October 21, 2010, 10:22:45 AM »

It turns out that the default directory for sound files in Asterisk 1.4 is actually /usr/share/asterisk/sounds, not /var/lib/asterisk/sounds. The sounds loaded in that file at install are a bit different these days, and "pls-wait-connect-call" is not in there. When I added a copy of it in that folder, it worked just fine.
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twinclouds
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« Reply #5 on: October 21, 2010, 03:22:44 PM »

Thank you.  I tried to find out for sometime without success.  How did you find out?  I will try it tonight.  Once verified working, I will revised my how to.
« Last Edit: October 21, 2010, 04:14:44 PM by twinclouds » Logged
jet101
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« Reply #6 on: October 21, 2010, 03:47:47 PM »

I was setting up a record process in the dialplan. When I made a test call, I saw it gave me an error trying to find a file, and the error showed the path it was trying to use. I got to digging around on google and found out about the right path in Asterisk 1.4.
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twinclouds
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« Reply #7 on: October 21, 2010, 04:26:13 PM »

Thanks.
One personal preference.  I don't like the announcement of the parking slot number.  I modified the script based on http://michigantelephone.wordpress.com/2010/05/21/how-to-use-google-voice-for-free-outgoing-calls-on-an-asteriskfreepbx-system-the-easy-way/.  Then that was disabled.  I don't what it does but at least the number was no longer announced.  Just want to share.

The latest:  Yes, it works.  I updated my how-to.
« Last Edit: October 22, 2010, 03:34:09 PM by twinclouds » Logged
toolbox123
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« Reply #8 on: October 23, 2010, 12:04:46 PM »

Which file did you update to stop announcement of the parking slot number? Thanks.
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twinclouds
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« Reply #9 on: October 23, 2010, 02:24:10 PM »

The gv-out portion of the extensions.conf.  It essentially send the announcement to a device dsp that is not available.  Not sure it will affect the connectivity or not but you can try to find out.
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PlugPBX Admin
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« Reply #10 on: October 23, 2010, 07:39:52 PM »

FYI guys, Asterisk 1.8 directly supports Google Voice. I haven't got the details, just read that in my RSS feeds tonight.

You might want to look into that perhaps boys?

I'll have to see if Asterisk 1.8 is 'stable' enough to include in PlugPBX
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-Greg
twinclouds
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« Reply #11 on: October 24, 2010, 10:43:17 AM »

FYI guys, Asterisk 1.8 directly supports Google Voice. I haven't got the details, just read that in my RSS feeds tonight.

You might want to look into that perhaps boys?

I'll have to see if Asterisk 1.8 is 'stable' enough to include in PlugPBX

It looks like that way.  I need to find out more details.  Thanks for heads up.
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jet101
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« Reply #12 on: October 31, 2010, 09:18:43 AM »

The NerdVittles guys are working on an Asterisk 1.8, FreePBX 2.8 solution that uses Google Voice and a Jabber/Jingle combination to make instantaneous calls. They are supposed to release information tomorrow, Nov. 1. This would be a great DockStar application. No more please wait, 7,5, and MOH while waiting for the parked call functions to complete.
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twinclouds
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« Reply #13 on: October 31, 2010, 10:12:21 AM »

I installed Asterisk 1.8 with no problem.  However, I cannot make gtalk work yet.  Maybe I need more experiment and more examples.
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toolbox123
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« Reply #14 on: November 19, 2010, 12:26:58 PM »

Should these instructions work with asterisk 1.6.2.9? I was able to make one outgoing call with the setup. However, all subsequent calls ended in buys tone.

Part of the log:
Code:
    -- Executing [MySipGateID@sipgate_in:4] [1;36mSet[0m("[1;35mSIP/sipgate-0000000b[0m", "[1;35mGVNUM=MyGV#[0m") in new stack
    -- Executing [MySipGateID@sipgate_in:5] [1;36mSet[0m("[1;35mSIP/sipgate-0000000b[0m", "[1;35mGVNUM1=1MyGV#[0m") in new stack
    -- Executing [MySipGateID@sipgate_in:6] [1;36mLog[0m("[1;35mSIP/sipgate-0000000b[0m", "[1;35mNOTICE, CALLERID: MyGV#[0m") in new stack
[Nov 19 11:21:03] [1;33mNOTICE[0m[2734]: [1;37mExt. MySipGateID[0m:[1;37m6[0m [1;37m@ sipgate_in[0m:  CALLERID: MyGV#
    -- Executing [MySipGateID@sipgate_in:7] [1;36mGotoIf[0m("[1;35mSIP/sipgate-0000000b[0m", "[1;35m1?10[0m") in new stack
    -- Goto (sipgate_in,MySipGateID,10)
    -- Executing [MySipGateID@sipgate_in:10] [1;36mParkedCall[0m("[1;35mSIP/sipgate-0000000b[0m", "[1;35m75[0m") in new stack

[Klanai*CLI>
[0K    -- Channel SIP/sipgate-0000000b connected to parked call 75
    -- Native bridging SIP/sipgate-0000000b and SIP/101-0000000a

[Klanai*CLI>
[0K    -- Got SIP response 420 "Option Disabled" back from 204.155.28.10

[Klanai*CLI>
[0K  == Spawn extension (sipgate_in, MySipGateID, 10) exited non-zero on 'SIP/sipgate-0000000b'

[Klanai*CLI>
[0K
JABBER: superm1 INCOMING: 

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