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I'm having trouble finding the exact documentation to do the following:

I have a SIP Account. I want my Asterisk Server on a VPS somewhere in the United States to accept the credentials of the SIP Account. When people call into my SIP account phone number at 111-222-3333 extension 55, it should re-route the call to my cell phone which is located somewhere in Canada.

Can anyone tell me how to do this? Or refer to me the relevant documentation?

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2 Answers 2

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Alright, I got things to work. This is EXACTLY what my sip.conf and extensions.conf looks like, I left all the other configuration files untouched

sip.conf - a) replace [username], [password], [host] and [port] with the appropriate values

b) because my SIP provider is very finicky, I had to try various values for [host], and sometimes the [host] in the register => line was a different value from [host] in the host= line (but this may not be a problem for others)

[general]
register => [username]:[password]@[host]:[port]
context=default

[mysipprovider]
type=friend
secret=pass
username=[username]
host=host
port=5070
fromuser=[username]
canreinvite=no
;insecure=very
qualify=2000
dtmfmode=inband
nat=yes

extensions.conf

[default]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(vm-extension)
exten => s,n,WaitExten()


exten => 55,1,Dial(SIP/mysipacc/3332221111) ; extension 55 calls phone 3332221111
exten => 55,n,Hangup

exten => 66,1,Dial(SIP/mysipacc/1112225555) ; extension 66 calls phone 1112225555
exten => 66,n,Hangup
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You should do basically 2 things:

  1. Setup Asterisk server to allow proper registration of your SIP account. This is done configuring the SIP credentials at /etc/asterisk/sip.conf
  2. Configure Asterisk dialplan to map extension 55 as a dialout to your cellphone. This is done at /etc/asterisk/extension.conf

You will find extensive documentation about how to do this at the voip-info.org site: sip.conf and extension.conf). This other link seems to be a good example of what you need.

Heres a small example on how it could look like:

sip.conf

[mysipprovider] 
type=peer 
secret=password 
username=2345 
host=sipserver.mysipprovider.com 
fromuser=2345 
canreinvite=no 
insecure=very 
qualify=yes 
nat=yes 
context=from-mysipprovider ; this section will be defined in extensions.conf 

and at the extension.conf:

[from-mysipprovider]
exten => 55,1,Verbose(1|Echo test application)
exten => 55,n,Dial(SIP/mysipprovider/5551234); Here is the outbound call, the exact dialstring depends on outgoing provider and channeltype
exten => 55,n,Hangup()
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