kenny.barnt.us

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Testing

Now that we’ve got everything configured, all we need to do is test our setup.

1. Dial your Google Voice number from a land-line or cell phone. This should ring your SIP phone. Try this a few times and:
   • Answer the call
   • Let the call ring through to voicemail (unavailable greeting)
   • Reject the call (busy greeting)
2. Dial 1002 from your SIP phone. This should give you an error message.
3. Dial 1009 from your SIP phone. This should connect you to the voicemail system. Enter 1000 for the mailbox number, and 1234 for the password. Try recording greetings and playing any messages you’d left earlier.
4. Call a land-line or cell phone from your SIP phone.
5. Dial 1010 from your SIP phone, this should ring the phone number you specified in extensions.conf
6. Dial 3000 from your SIP phone, this should playback a test message.
7. Dial 911 from your SIP phone, this should playback an error message.

Conclusion

Now that you’ve got started with Asterisk, you can customize your setup to do any number of things, all it takes is a little creativity and ingenuity.

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Now the final, most involved part of the process: Configuring the dialplan. We’ll take this bit by bit, but if you’re not sure about what a particular line does, check out the Wiki at Voip-Info.org.

Dialplan Basics
First we’ll get the basics of our dialplan configuration in place.

1. Create /etc/asterisk/extensions.conf with your favorite text editor.
2. Add the following block to the new extensions file
   [general]

   [globals]

   This gives us some general structure for the extensions configuration.

3. Next, we will add a subroutine for handling calls to internal extensions. This will simplify the dialplan later on.
   [internal-call]
   exten => s,1,Verbose(====> Processing Internal Call)
   exten => s,n,Set(DIALEXT=${ARG1})
   exten => s,n,Verbose(====> Call is from ${CALLERID(all)} to ${DIALEXT})
   exten => s,n,Dial(SIP/${DIALEXT},15,tT)
   exten => s,n,GotoIf($["${SIPPEER(${DIALEXT},status)}" = ""]?badext:vmsw)
   exten => s,n(vmsw),GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
   exten => s,n(unavail),Verbose(====> Extension ${DIALEXT} is unavailable)
   exten => s,n,Voicemail(${ARG1}@default,u)
   exten => s,n,Hangup()
   exten => s,n(busy),Verbose(====> Extension ${DIALEXT} is busy)
   exten => s,n,Voicemail(${ARG1}@default,b)
   exten => s,n,Hangup()
   exten => s,n(badext),Verbose(====> Extension dialed does not go to a valid SIP channel)
   exten => s,n,Playback(pbx-invalid)
   exten => s,n,Hangup()
   exten => s,n,Return

   exten => h,1,Verbose(====> Ending Internal Call Processing - call was hung up)
exten => h,n,Return
   This subroutine take an extension as an argument and dials it. If the extension doesn’t exist, an error message is played. If the extension is busy or unavailable, the call is transferred to Voicemail, which plays the appropriate greeting. s and h are special extensions in Asterisk. s acts as a placeholder for calls entering the context, while h defines actions that should be taken when a call in the context is hung up.

4. Now we’ll add a subroutine to handle making the outgoing calls using Google Voice
   [gv-out]
   exten => s,1,Verbose(====> Exexution passed to Google Voice Handler)
   exten => s,n,Set(CALLING=${ARG1})
   exten => s,n,Set(GVUID=googlevoiceusername)
   exten => s,n,Set(GVPASS=googlevoicepassword)
   exten => s,n,Set(SGNUM=5555550000)
   exten => s,n,Set(PARKEXT=7100)
   exten => s,n,Playback(pls-wait-connect-call)
   exten => s,n,Set(PARKINGEXTEN=${PARKEXT})
   exten => s,n,Park()

   exten => h,1,Verbose(====> Call Parked - Initiating Google Voice Call)
exten => h,n,System(gvoice -e ${GVUID} -p ${GVPASS} call ${CALLING} ${SGNUM})
exten => h,n,Verbose(====> GV Call initated)
   This subroutine takes a single argument: the 11-digit number to call out to. You’ll need to change GVUID, GVPASS, and SGNUM to be your Google Voice username, Google Voice password, and sipgate phone number.

5. Now we’ll add the [incoming] context. This is where all calls coming in from sipgate end up.
   [incoming]
   exten => 9999999e1,1,Verbose(====> Processing incoming sipgate call from ${CALLERID(num)})
   exten => 9999999e1,n,Set(GVCID=5555555555)
   exten => 9999999e1,n,Set(RINGEXT=1000)
   exten => 9999999e1,n,Set(PARKEXT=7100)
   exten => 9999999e1,n,GotoIf($[${CALLERID(num)}=${GVCID}]?gvringback:incoming)
   exten => 9999999e1,n(gvringback),Verbose(====> Call is a Google Voice Ringback)
   exten => 9999999e1,n,Answer()
   exten => 9999999e1,n,ParkedCall(${PARKEXT})
   exten => 9999999e1,n,Verbose(====> Ringback Connected)
   exten => 9999999e1,n,Return
   exten => 9999999e1,n(incoming),Verbose(====> Call is a regular incoming call)
   exten => 9999999e1,n,Gosub(internal-call,s,1(${RINGEXT}))
   exten => 9999999e1,n,Verbose(====> Execution returned to incoming)
   exten => 9999999e1,n,Return

   exten => h,1,Verbose(====> End processing incoming sipgate call - call was hung up)
exten => h,n,Return
   You’ll need to change 9999999e1 to your sipgate SIP-ID and GVCID to your 10-digit Google Voice number

6. Finally, we’ll add the [internal] extension, where calls from any of our phones end up.
   [internal]
   ; Internal Calls
   exten => _100[1-8],1,Verbose(====> Internal call)
   exten => _100[1-8],n,GoSub(internal-call,s,1(${EXTEN})
   exten => _100[1-8],n,Verbose(====> End internal call)

   exten => 1009,1,Verbose(====> Dialing voicemail)
   exten => 1009,n,VoiceMailMain(@default)
   exten => 1009,n,Verbose(====> End dialing voicemail)

   exten => 1010,1,Verbose(====> Speed dial 15555557777)
   exten => 1010,n,GoSub(gv-out,s,1(15555557777))
   exten => 1010,n,Verbose(====> End speed dial 15555557777)

   exten => 3000,1,Verbose(====> Play a test message)
   exten => 3000,n,Playback(tt-weasels)
   exten => 3000,n,Verbose(====> End playing test message)

   exten => 911,1,Verbose(====> 911 call)
   exten => 911,n,Playback(no-911-1)
   exten => 911,n,Hangup()
   exten => 911,n,Verbose(====> End 911 call)

   ; Outbound Calls
   exten => _1NXXNXXXXXX,1,Verbose(====> 11-digit outbound call to ${EXTEN})
   exten => _1NXXNXXXXXX,n,GoSub(gv-out,s,1(${EXTEN}))
   exten => _1NXXNXXXXXX,n,Verbose(====> End 11-digit outbound call to ${EXTEN})

   exten => _NXXNXXXXXX,1,Verbose(====> 10-digit outbound call to ${EXTEN})
   exten => _NXXNXXXXXX,n,GoSub(gv-out,s,1(1${EXTEN}))
   exten => _NXXNXXXXXX,n,Verbose(====> End 10-digit outbound call to ${EXTEN})

   exten => _NXXXXXX,1,Verbose(====> 7-digit outbound call to ${EXTEN})
exten => _NXXXXXX,n,Set(AREACODE=555)
exten => _NXXXXXX,n,GoSub(gv-out,s,1(1${AREACODE}${EXTEN}))
exten => _NXXXXXX,n,Verbose(====> End 7-digit outbound call to ${EXTEN})
   This is the real meat of the dialplan. You’ll want to change AREACODE to whatever area code you want to use for 7-digit dialing, as well as the number for the speed dial on extension 1010. As you can see, combining pattern matching with naming the SIP channels to match their associated extension number reduces the number of entries required in the dialplan. It also allows us to add more phones by just configuring sip.conf and voicemail.conf, with no changes to the dialplan.

7. Now, reload the dialplan from the Asterisk CLI
   asterisk*CLI> dialplan reload

Now that we’ve configured the dialplan, all that’s left to do is test… (continue)

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With Asterisk installed and sipgate and Google Voice configured, it’s time for the meat and potatoes of this whole thing: configuring Asterisk.

safe_asterisk
Before we get into the configuration, though, we need to make a quick security change. We’ll be using safe_asterisk to start asterisk as a daemon, because it will automatically restart Asterisk in case it crashes for some reason. The safe_asterisk script, however, has a fairly serious security flaw out of the box, and we need to patch it.

1. Open /usr/sbin/safe_asterisk in your favorite text editor and comment out the TTY=9 line.
   # vim /usr/sbin/safe_asterisk…
   #!/bin/sh
# vim:textwidth=80:tabstop=4:shiftwidth=4:smartindent:autoindent
CLIARGS="$*" # Grab any args passed to safe_asterisk
TTY=9 # TTY (if you want one) for Asterisk to run on…becomes…
   #!/bin/sh
# vim:textwidth=80:tabstop=4:shiftwidth=4:smartindent:autoindent
CLIARGS="$*" # Grab any args passed to safe_asterisk
#TTY=9 # TTY (if you want one) for Asterisk to run on
2. Start Asterisk with safe_asterisk
   # safe_asterisk

Asterisk Configuration
Now that Asterisk is running, we need to get everything configured for our calls to work. If you’re not familiar with Asterisk Configuration, you may want to check out the wiki at Voip-Info.org (search for the file you’re working on) or the Asterisk Documentation Project for some good references.

1. Before we begin our configurations, we’ll make backup copies of the samples of the configuration files we’ll change
   # cd /etc/asterisk
# mv extensions.conf extensions.sample
# mv sip.conf sip.sample
# mv voicemail.conf voicemail.sample
# mv features.conf features.sample
2. We also need to a little bit of planning at this point. You’ll need to decide how you want to setup your dialplan. In this example, we’re going to use four digit internal extension numbers, with the names of SIP peers set to their extension numbers. You’re free to change this to suit your needs or wants, and I’ll try and point out all the things you will need to change from these examples.

SIP Configuration
The first thing we’ll configure is our SIP settings. We’ll need to configure Asterisk to connect to sipgate, and also configure Asterisk to accept connections from a phone. In this and any other examples 9999999e1 is your sipgate SIP-ID and AAAA1A is your sipgate SIP-Password.

1. Create the file /etc/asterisk/sip.conf in the text editor of your choice.
2. Add this to the beginning of the new file:
   [general]
register => 9999999e1:AAAA1A@sipgate.com/9999999e1
   The [general] section of sip.conf is where configuration options that apply to all SIP peers are set. It is also where register statements are placed. Register statements allow Asterisk to connect as a client to other SIP servers. We connect to sipgate with a register statement so that calls coming into our sipgate number come to Asterisk.
3. Add this block below the previous one:
   [sipgate]
type=peer
host=sipgate.com
outboundproxy=proxy.live.sipgate.com
insecure=invite
qualify=yes
dtmfmode=rfc2833
username=9999999e1
defaultuser=9999999e1
fromuser=9999999e1
fromdomain=sipgate.com
secret=AAAA1A
context=incoming
disallow=all
allow=ulaw
   This [sipgate] section of sip.conf configures the options to use for the sipgate connection. These options are all specified by sipgate.
4. Add this last block below the [sipgate] section:
   [1000]
type=friend
host=dynamic
nat=yes
qualify=yes
context=internal
defaultuser=1000
secret=1000
callerid="Your Name" <1000>
mailbox=1000
   This section defines the configuration for our SIP phone to connect to Asterisk at extension 1000.
5. Save the file as /etc/asterisk/sip.conf . Now, connect to the running Asterisk daemon to get to the Asterisk CLI
   # asterisk -rvvv
6. At the Asterisk command prompt, reload the SIP configuration. Most configuration changes in Asterisk don’t require restarting Asterisk, but simply reloading the configuration files that have been changed.
   asterisk*CLI> sip reload
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': == Found
== Parsing '/etc/asterisk/sip_notify.conf': == Found
asterisk*CLI>
7. Now we’ll setup our SIP phone to connect to Asterisk. Use 1000 in place of your sipgate SIP-ID and SIP-Password, and the hostname or IP of your server in place of sipgate.com in the phone’s configuration.
8. Let’s make sure our SIP configuration is working properly.
   asterisk*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipgate.com:5060 N 1207176e0 105 Registered Tue, 20 Apr 2010 19:16:08
1 SIP registrations.
asterisk*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
1000/1000 10.13.10.85 D N 5060 OK (50 ms)
sipgate-kenny/9999999e1 172.16.17.10 5060 OK (45 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]
asterisk*CLI>
   If you don’t have 1 registration and 2 peers showing up, there’s something amiss, and you’ll need to look into it.

Next we’ll configure voicemail.conf and features.conf… (continue)