Asterisk Server with SIP Account Routing to Cell Phones

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?

From serverfault John

• 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()
  

  From Pablo Alsina

• 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
  

  From John