Even though Asterisk 1.8 was released, I didn't see any developers jumping on creating OpenWRT packages, so I decided to do so myself. I've produced packages for both ar71xx devices (like the one I detailed in earlier posts) as well as brcm-2.4 devices that the market is flooded with.
These packages can be found at http://www.arctangent.net/~superm1/openwrt/.
I've also submitted the packaging changes to a ticket for inclusion in a future OpenWRT release at https://dev.openwrt.org/ticket/8132.
So for an ar71xx device, you would add this line to your /etc/opkg.conf:
src/gz asterisk1.8 http://www.arctangent.net/~superm1/openwrt/ar71xx
If you've got asterisk 1.6 installed, you'll need to remove all of the associated packages for it before proceeding.
opkg remove asterisk16-sounds asterisk16-res-musiconhold asterisk16-res-agi asterisk16-func-db asterisk16-format-g726 asterisk16-form
at-g729 asterisk16-codec-g726 asterisk16-codec-a-mu asterisk16-app-system asterisk16
For Asterisk 1.8, I think these are the only packages you'll need:
opkg install asterisk18 asterisk18-func-db asterisk18-res-musiconhold asterisk18-chan-gtalk asterisk18-res-rtp-asterisk
Configuration Files
Once installed, you need to add your google information to /etc/asterisk/gtalk.conf. This configuration will dump my incoming gtalk calls directly to a context that will be defined as 'google-in'.
[general]
context=google-in ; Context to dump call into
[guest] ; special account for options on guest account
disallow=all
allow=ulaw
;context=guest
[superm1-gtalk]
username=superm1@gmail.com
disallow=all
allow=ulaw
context=google-in
connection=superm1
Next configure jabber information in /etc/asterisk/jabber.conf. This setup will cause you to always be signed into jabber, so make sure to set yourself away and give an informative message to know that it's a computer signed in, not a human.
[superm1]
type=client
serverhost=talk.google.com
username=superm1@gmail.com
secret=(your password)
statusmessage=I am asterisk [*]
status=xaway
Lastly configure your /etc/asterisk/extensions.conf dialplan for the new inbound and outbound features. This is built upon my previous dialplan. It will support incoming sip or google talk/voice as well as outgoing SIP, toll free, and GV.
; --
; Inbound Calls
; --
; * This extension is where any external SIP calls should route to
[default]
exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
exten => s,n,Dial(SIP/101,10)
exten => s,n, Hangup
exten => 101, 1, Dial(SIP/101, 10)
; * From Google (superm1)
[google-in]
exten => superm1@gmail.com, 1, Dial(SIP/101, 180, D(:1))
; --
; Outbound Calls
; --
; * Default starting context for internal SIP devices
[mario-default]
include => local-devices
include => tollfree
include => gv-outbound
include => dial-uri
; * These are for any local extensions we should be supporting
[local-devices]
exten => _1, 1, Dial(SIP/101,10)
; * Toll free numbers (don't use GV for these)
[tollfree]
exten => _411, 1, Dial(SIP/18004664411@proxy.ideasip.com,60)
exten => _1800NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1888NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1877NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
exten => _1866NXXXXXX,1,Dial(SIP/${EXTEN}@proxy.ideasip.com,60)
; * Route the call using the google voice bridge
[gv-outbound]
;append an area code if necessary
exten => _NXXXXXX,1,Set(CALLERID(dnid)=1512${CALLERID(dnid)})
exten => _NXXXXXX,n,Goto(1512${EXTEN},1)
;append a 1 if necessary
exten => _NXXNXXXXXX,1,Set(CALLERID(dnid)=1${CALLERID(dnid)})
exten => _NXXNXXXXXX,n,Goto(1${EXTEN},1)
;do our real dialing
exten => _1NXXNXXXXXX,1,Dial(Gtalk/superm1/${EXTEN}@voice.google.com)
exten => _+1NXXNXXXXXX,1,Dial(Gtalk/superm1/${EXTEN}@voice.google.com)
; * Dialing by SIP URL eg foo@domain.com
[dial-uri]
exten => _[a-z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _[A-Z].,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
exten => _X.,1,Dial(SIP/${EXTEN}@${SIPDOMAIN},120,tr)
Advantages / Disadvantages
Advantages
- This configuration is far quicker and more like a real phone. When you dial a number, it will be connected within seconds rather than 30's of seconds.
- The quality seems better than the old way (and in theory should be less hops, and less latency)
Disadvantages
- You have to always be signed into Google Talk/Chat with this solution. If people IM you and you aren't on a computer, then the IM is lost in your gmail history.
- If you are signed into Gmail Chat and the computer supports video (has the plugin installed) then your SIP phone won't ring.
Thanks
Thanks to Paul Jennings for helping to provide a quite functional configuration to get started here for me, as well as AST/Calling+using+Google for detailing all the options actually available to configure.Updates
11/23:
This post seems to be the post that people are linking to and reading mostly when setting up Asterisk 1.8. Just want to make sure I mention, there have been improvements since this post (as detailed in other future blog postings). To avoid having to relive the learning experience I did with my other improvements, my (scrubbed for passwords) configs are available at http://www.arctangent.net/~superm1/gv_configs/
