http://www.ipkall.com/
http://www.kallfree.com/

Free personal Washington State (USA) PSTN number that forwards to your Free World Dialup number. Should also work with any other free SIP service as both the user and proxy can be configured!

• VoIP Service Providers

Support

• IPKall Support Forum

First, register an IPKall number at either of the web addresses provided above.
IPKall will ask for the following information:

SIP phone number
SIP proxy
Email Address
Password (4 digit PIN)
Voicemail preferences

When specifying the SIP phone number, you can use any number, but you should try to avoid using
a number that's already an extension in your extensions.conf file.
Put the IP address (or hostname) of your Asterisk server into the SIP proxy field.
If your SIP proxy is not running on the default port 5060, then you will need to specify the port.
For example, if your IP is 123.254.254.1 and your SIP proxy is running on port 7777,
then you would enter 123.254.254.1:7777 as the SIP proxy. Be sure to enter a valid e-mail address.

Here is a sample IPKall configuration for sip.conf and extensions.conf.
Place this under [inbound] in extensions.conf. (note, if [inbound] already exists, don't add it again):
important note: your incoming context may be, and probably is different than [inbound].
Replace [inbound] with the appropriate context in both extensions.conf and the "context=" line below.


Where 508 is the SIP phone number you specified when setting up IPKall.
Now, put the following information in your sip.conf file:




1. When you dial your IPKall DID, IPKall sends a request (i.e. 508@123.254.254.1) to your SIP proxy
2. Asterisk accepts the request, since the [508] context in sip.conf tells Asterisk to accept incoming calls with the number "508"
3. Asterisk searches extensions.conf for the "inbound" context (since we specified context=inbound in the [508] context)
4. Asterisk then matches extension "508" in the [inbound] context, jumping to "your-main-menu" in this example.



G.711
GSM
iLIBC
G.729 (since Feb 2006)


If you just get a busy signal when calling your IPKall number, try the following:

1. From the Asterisk CLI (asterisk -r) turn on sip debugging by typing "sip debug", try calling your IPKall number again, when you get the busy signal, turn off sip debugging by typing "sip no debug". Then read through the debugging information for clues.

2. Make sure host=voiper.ipkall.com is specified in the IPKall context (in the above example, [508]), without this option you will likely get "SIP/2.0 404 Not Found" or "Found no matching peer or user for" sip debug errors. Since we are specifying a hostname, make sure DNS resolution works (nslookup voiper.ipkall.com)

3. If any part of the network between your Asterisk server and the Internet uses NAT, specify externip= and localnet= in sip.conf. Set externip= to your external IP address and localnet= to your local subnet

For example:
externip=123.254.254.1
localnet=192.168.0.0/255.255.255.0

4. If you have a firewall restricting access to your Asterisk server, make sure UDP ports 5060 and 10000 through 20000 are properly forwarded.

---

Settings for add Trunk in freepbx 2.3.0.2

1. click add sip trunk
2. leave everything blank except the following items

Outgoing Settings

Trunk Name: ipkall
Peer Details

context=from-pstn
host=voiper.ipkall.com
type=peer

Incoming Settings

USER Context from-trunk
USER Details:

allow=g729
context=from-trunk
disallow=all
dtmfmode=rfc2833
host=voiper.ipkall.com
insecure=very
type=peer

Note

If your SIP devices are behind a NAT and your Asterisk

; server isn't, try adding "nat=1" to each peer definition to
; solve translation problems.

general

1. include sip_general_additional.conf

externip = xx.xx.xx.xx ;your external IP
localnet = 10.80.0.0/255.255.255.0 ;your local IP most people are 192.168.1.x/255.255.255.0
progressinband=yes
bindport = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
qualify=yes
nat=yes
disallow=all
allow=ulaw
allow=alaw
; If you need to answer unauthenticated calls, you should change this
; next line to 'from-trunk', rather than 'from-sip-external'.
; You'll know this is happening if when you call in you get a message
; saying "The number you have dialed is not in service. Please check the
; number and try again."
context = from-trunk ; Send unknown SIP callers to this context