Being woken up several times throughout the night from anonymous calls is not fun.  Here is a screencast (shot with my shiny new MacBook) that explains how to delimit these annoying calls while still being able to route incoming SIP calls from Gizmo and IPKall to their appropriate destinations.

Here is the code I used to allow IPKall incoming SIP connections :


[ipkall]
disallow=all
host=66.54.140.46
context=from-trunk
insecure=port,invite
qualify=yes
type=peer
dtmfmode=rfc2833
allow=ulaw
nat=no

[ipkall2]
disallow=all
host=66.54.140.47
context=from-trunk
insecure=port,invite
qualify=yes
type=peer
dtmfmode=rfc2833
allow=ulaw
nat=no


Well I just received an invite to Google Voice and I must say I was quite upset to discover that it is not yet available in Canada… nonetheless I had a socks box in Cali so I tunnelled my registration and was ready to rock.

For those of you unfamilliar with Socks – it basically pushes all requests through a secure connection on a remote system and is quite simple to do – take a look at my “How to Watch American TV in Canada” post from a while back.

Once I was in I ran into another little snag, you can not set your call-in number to an international DID (in my case a Canadian DID).  So I just pointed them to my free IPKall number and was really cooking with gas (you too can get a free IPKall DID at ipkall.com).

For those of you anticipating an invite yuo can expect the usual Google UI (thank god (I adore simplicity)), you can also look forward to transcribing your voicemail messages to text – and have them emailed.  This functionality has been lacking in Asterisk which is why I believe this is going to drop a nuke on the Asterisk devs to get this module out – so support your * developers will ya!

I’ve really only done minimal “tire-kicking” but I expect this could have many telcos shaking in their boots.  Not only is it super feature rich, sociable and ridiculously easy to use – it is backed by a company who has provided extremely reliable services since day one.

It is interesting to see the callout function implementation.  There have been plenty of callout scripts for Asterisk (which pretty much work exactly the same – Asteristickies is one of ‘em) but once again Google has made it that much friendlier.  Callout is a dialing function when you input a number to call, it connects to you first – then dials the remote party.

SMS is a joy! I was lucky enough to have my sister visiting this week with her unlimited text messaging plan.  I popped open the SMS box and we were sending messages with ease instantaneously.  I especially love the contact manager.  Being able to maintain a list of contacts with click to call functionality is a very powerful feature.

Anyway, I am very excited to play with my new toy – feel free to give me a ring at

(361)-GEEK-HUT

Heh… or use this thing

Follow these easy steps to point a free Washington area code phone number to your Gizmo account.  This is a great solution for those overseas who wish to have a North American DID (for free / cheap calls.)

I was having some one way audio issues with IPKall – here is the solution (examine bottom two entries) :

The key here is mapping incoming 5060 to internal 5068 and 10000 to internal 20000.  Voila!

You also want to make sure you have :

externip=<extenal ip address>

externhost=<fqdn, unless you have externip set>

localnet=10.0.0.0/255.255.255.0

srvlookup=yes

nat=yes

in the /etc/asterisk/sip_custom.conf file.