http://www.ipkall.com/
http://www.kallfree.com/
Free personal Washington State (USA) PSTN number that forwards to any SIP or IAX destination you specify, including your own Asterisk server.
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. ...
http://www.kallfree.com/
Free personal Washington State (USA) PSTN number that forwards to any SIP or IAX destination you specify, including your own Asterisk server.
Support
Instructions on setting up a direct incoming IPKall phone number under Asterisk, bypassing FWD
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.
[inbound]
exten => 508,1,Goto(your-main-menu|s|1)
exten => 508,1,Goto(your-main-menu|s|1)
Where 508 is the SIP phone number you specified when setting up IPKall.
Now, put the following information in your sip.conf file:
[508] ;IPKall
type=peer
dtmfmode=rfc2833
context=inbound
insecure=very
host=voiper.ipkall.com
nat=no
type=peer
dtmfmode=rfc2833
context=inbound
insecure=very
host=voiper.ipkall.com
nat=no
How it works
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.
Codecs Supported
G.711
GSM
iLIBC
G.729 (since Feb 2006)
Troubleshooting
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. ...