After completely rewriting ENUMPlus from the ground up (2 days total) we have come much closer to a universal ENUM lookup source.  The new module can be downloaded here so install it now! Here’s why :

• The module forms a HTTP request to our API which queries many other sources and returns the most accurate address.
• There is no longer a need to add multiple lookup sources to the old enumlookup AGI.
• It audibly notifies you when the call is made over ENUM.
• You can track your calls in the panel.
• White pages / search functionality (including avatars )
• Call validation is much more stable.

If you don’t have an account, get one – registration is quick and simple and you will be a happy geek.

Check out the project page / wiki / site / sourceforge / FreePBX / voip-info / piaf for more info.

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


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.

Spent about an hour researching how to do this and once again I am blogging it for future reference, I hope you will find it useful as well.

IPKall lets you (for free) register a Washington DID (phone number) that can be passed to an Asterisk inbound route.  Do not be confused – it is not a sip proxy, you do not get free VoIP.  It just gives you a routable* inbound number which can be passed to any service on your Asterisk system.

To set it up simply:

• Register a free account at ipkall.com.
• Point SIP server to your asterisk IP (use everydns or dyndns)
• Specify your SIP port (default : 5060)
• Create a random unused extension number – so asterisk can identify and route.

Next all you have to do is login to your Trixbox panel and click Inbound Routes, create a new inbound route and put in the “DID” field the unused extension you specified above.  Then select a destination for this number and voila – you have a new phone number.

The biggest benefit to this as far as I can tell would be if either :

a) You have relatives in Washington or,

b) You need an extra DID to receive faxes (not recommended with this service).

Anyway, I have mine playing Soma FM – give it a call 425-606-3524.