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Knowledgebase: SIP Trunking > Device Configuration
Asterisk Distribution Configuration Guide
Posted by , Last modified by Albert Diaz on 27 August 2018 02:08 PM

For Asterisk setups the following needs to be done.

  • Find the Asterisk version
  • Modify extensions_custom.conf
  • Add SIP Trunk

For this example we will be using Elastix. 

 Finding Asterisk Verison

Use Putty or use the Asteris-CLI

asterisk -r
core show version

Or

asterisk -r
show version

 

Modify extensions_custom.conf 

  1. Log into the PBX and go to the PBX tab
  2. Go to Tools
  3. From the left menu options click on Asterisk File Editor and find extensions_custom.conf
  4. Add the following text to extensions_custom.conf (depending on the Asterisk version)

Asterisk versions 1.6 or older

[from-outside-redir]
;Dialed Number routing according to SIP To: header
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(TO),@,1),:,2):1},1)

Asterisk versions 1.8 or newer

[from-trunk]
;Dialed Number routing according to SIP To: header
exten => s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(TO),@,1),:,2):1},1)

  1. Click Save
  2. Reload Asterisk

 

 

Add SIP Trunk

  1. Go to PBX Configuration
  2. From the left menu options click on Trunks
  3. From the right menu options click on Add Trunk
  4. Click on Add SIP Trunk

  1. Add the Trunk Name
  2. Add the Outbound Trunk Name
  3. Add the PEER Details - Change the ACCOUNT-ID and SIP-PASSWORD based on your acctual account information. Based on the Asterisk version use either from-outside-redir or from-trunk

Asterisk versions 1.6 or older

type=peer
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-outside-redir
username=ACCOUNT-ID
trustrpid=yes
sendrpid=yes
secret=SIP-PASSWORD
allow=all
canreinvite=no

Asterisk versions 1.8 or newer

type=peer
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-trunk
username=ACCOUNT-ID
trustrpid=yes
sendrpid=yes
secret=SIP-PASSWORD
allow=all
canreinvite=no

  1. Add the USER Context
  2. Add the USER Details

Asterisk versions 1.6 or older

type=user
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-outside-redir
allow=all

Asterisk versions 1.8 or newer

type=user
insecure=invite,port
dtmfmode=auto
host=sbc.ringlogix.com
context=from-trunk
allow=all

  1. Add the Registration String

ACCCOUNT-ID:SIP-PASSWORD@sbc.ringlogix.com:5060

  1. Click Submit Changes

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