Prerequisites
Asterisk communicates with Google Voice and Google Talk using the chan_gtalk Channel Driver and the res_jabber Resource module. Before proceeding, please ensure that both are compiled and part of your installation. Compilation of res_jabber and chan_gtalk for use with Google Talk / Voice are dependant on the iksemel library files as well as the OpenSSL development libraries presence on your system.
Calling using Google Voice or via the Google Talk web client requires the use of Asterisk 1.8.1.1 or greater. The 1.6.x versions of Asterisk only support calls made using the legacy GoogleTalk external client.
For basic calling between Google Talk web clients, you need a Google Mail account.
For calling to and from the PSTN, you will need a Google Voice account.
In your Google account, you'll want to change the Chat setting from the default of "Automatically allow people that I communicate with often to chat with me and see when I'm online" to the second option of "Only allow people that I've explicitly approved to chat with me and see when I'm online."
IPv6 is currently not supported. Use of IPv4 is required.
Google Voice can now be used with Google Apps accounts.
Gtalk configuration
The chan_gtalk Channel Driver is configured with the gtalk.conf configuration file, typically located in /etc/asterisk. What follows is the minimum required configuration for successful operation.
Minimum Gtalk Configuration
[general] context=local allowguest=yes bindaddr=0.0.0.0 externip=216.208.246.1 [guest] disallow=all allow=ulaw context=local connection=asterisk
This general section of this configuration specifies several items.
- That calls will terminate to or originate from the local context; context=local
- That guest calls are allowd; allowguest=yes
- That RTP sessions will be bound to a local address (an IPv4 address must be present); bindaddr=0.0.0.0
- (optional) That your external (the one outside of your NAT) IP address is defined; externip=216.208.246.1
The guest section of this configuration specifies several items.
- That no codecs are allowed; disallow=all
- That the ulaw codec is explicitly allowed; allow=ulaw
- That calls received by the guest user will be terminated into the local context; context=local
- That the Jabber connection used for guest calls is called "asterisk;" connection=asterisk
Jabber Configuration
The res_jabber Resource is configured with the jabber.conf configuration file, typically located in /etc/asterisk. What follows is the minimum required configuration for successful operation.
Minimum Jabber Configuration
[general] autoregister=yes [asterisk] type=client serverhost=talk.google.com username=your_google_username@gmail.com/Talk secret=your_google_password port=5222 usetls=yes usesasl=yes statusmessage="I am an Asterisk Server" timeout=100
The general section of this configuration specifies several items.
- Debug mode is enabled, so that XMPP messages can be seen on the Asterisk CLI; debug=yes
- Automated buddy pruning is disabled, otherwise buddies will be automatically removed from your list; autoprune=no
- Automated registration of users from the buddy list is enabled; autoregister=yes
The asterisk section of this configuration specifies several items.
- The type is set to client, as we're connecting to Google as a service; type=client
- The serverhost is Google's talk server; serverhost=talk.google.com
- Our username is configured as your_google_username@gmail.com/resource, where our resource is "Talk;" username=your_google_username@gmail.com/Talk
- Our password is configured using the secret option; secret=your_google_password
- Google's talk service operates on port 5222; port=5222
- TLS encryption is required by Google; usetls=yes
- Simple Authentication and Security Layer (SASL) is used by Google; usesasl=yes
- We set a status message so other Google chat users can see that we're an Asterisk server; statusmessage="I am an Asterisk Server"
- We set a timeout for receiving message from Google that allows for plenty of time in the event of network delay; timeout=100
Phone configuration
Now, let's place a phone into the same context as the Google calls. The configuration of a SIP device for this purpose would, in sip.conf, typically located in /etc/asterisk, look something like:
[malcolm] type=peer secret=my_secure_password host=dynamic context=local
Dialplan configuration
Incoming calls
Next, let's configure our dialplan to receive an incoming call from Google and route it to the SIP phone we created. To do this, our dialplan, extensions.conf, typically located in /etc/asterisk, would look like:
exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,SendDTMF(1) exten => s,n,Dial(SIP/malcolm,20)
This example uses the "s" unmatched extension, because Google does not forward any DID when it sends the call to Asterisk.
In this example, we're Answering the call, Waiting 2 seconds, sending the DTMF "1" back to Google, and then dialing the call.
We do this, because inbound calls from Google enable, even if it's disabled in your Google Voice control panel, call screening.
Without this SendDTMF event, you'll have to confirm with Google whether or not you want to answer the call.
Outgoing calls
Outgoing calls to Google Talk users take the form of:
exten => 100,1,Dial(gtalk/asterisk/mybuddy@gmail.com)
Where the technology is "gtalk," the dialing peer is "asterisk" as defined in jabber.conf, and the dial string is the Google account name.
Outgoing calls made to Google Voice take the form of:
exten => _1XXXXXXXXXX,1,Dial(gtalk/asterisk/+${EXTEN}@voice.google.com)
Where the technology is "gtalk," the dialing peer is "asterisk" as defined in jabber.conf, and the dial string is a full E.164 number (plus character followed by country code, followed by the rest of the digits).
Interactive Voice with Text Response (IVTR)
Because the Google Talk web client provides both audio and text interface, you can use it to provide a text-based way of traversing Interactive Voice Response (IVR) menus. This is necessary since the client does not have any DTMF inputs.
In the following dialplan example, we will answer the call, wait a bit, send some text that will show up in the caller's Google Talk client, play back a prompt, capture the caller's text-based response, and then dial the appropriate SIP device.
exten => s,1,Answer() exten => s,n,SendText("If you know the extension of the party you wish to reach, dial it now.") exten => s,n,Background(if-u-know-ext-dial) exten => s,n,Set(OPTION=${JABBER_RECEIVE(asterisk,${CALLERID(name)::15},5)}) exten => s,n,Dial(SIP/${OPTION},20)
Note that with the JABBER_RECEIVE function, we're receiving the text from asterisk which we defined earlier in this page as our connection to Google. We're also specifying with ${CALLERID(name)::15} that we want to strip off the last 15 characters from the CallerID name string - which is the number of characters that Google is appending, as of this writing, to represent an internal call ID number, and that we want to wait 5 seconds for a response.
Webinar
A Webinar was conducted on Tuesday, March 24, 2011 detailing Asterisk configuration for calling using Google as well as several usage cases. A copy of the slides, in PDF format, is available here - Google Calling Webinar - Public.pdf