asterisk configuration guide using accessline … configuration guide using accessline sip trunks 2...
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1 April 2013 Version 1.0
Asterisk Configuration Guide
Using AccessLine SIP Trunks
2 April 2013 Version 1.0
Setting up the Firewall
1) From the Back Office Panel, go to Security and then Define Rules.
2) Set the SIP ports to 5060-6060.
3) Change RTP ports to 30000-50000.
Setting up the trunks
1) Select Add Trunk.
2) Select Add Sip Trunk.
3) Set Outbound Caller Id to the preferred number.
4) Set Caller ID Options to Allow Any CID.
5) Change Maximum Channels to how many SIP lines the customer ordered.
6) Set Peer Details as follows:
host=usbc.accessline.com
username=dgwsidxxxxx
secret=xxxxxxxx
type=peer
canreinvite=no
nat=yes
qualify=yes
insecure=port,invite
NOTE: For Username and Secret use SIP Trunk ID & Password provided by Accessline.
7) Leave User Details blank.
8) Set the Register String as follows:
dgwsidxxxxx:[email protected]/MainNumber
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Setting up Outbound Routes
1) Caller ID can be set here. Extension and Trunk Caller ID will override this.
2) A route password can be set to ensure that international, long distance, etc. cannot be dialed
without a password.
3) Use the Dial Patterns Wizards to add dial plans for Long Distance, Emergency, Local,
International.
4) Under Trunk Sequence for Matched Routes select the trunk you would like to dial out on.
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Setting up Inbound Routes
1) Select Add Incoming Route.
2) Add a name for the Description.
3) Put in the DID in the DID Number section.
4) Set the destination for the DID at the bottom.
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Setting Caller ID by Extension
1) Select the extension you would like to change the caller ID for.
2) Under Extension Options change the Outbound CID to the CID you would like displayed.
3) Press Submit to save changes.
Changing Sip Settings
1) Make ulaw the primary codec and the only one enabled.
2) Set Video Support to Disabled.