Specific VoIP providers¶
Create an interconnection¶
There are three types of interconnections :
SIP interconnections are used to connect to a SIP provider to to another PBX that is part of your telecom infrastructure.
General SIP configurations are available inand trunk configurations are in
Environment with NAT¶
There are some configuration steps that are required when connecting to a SIP provider from a NAT environment.
Inset your External IP address and your Local network.
- External IP address: This is your public IP address
- Local network: Your internal network range
Inset the NAT option to Yes and the Monitoring option to Yes.
Customized interconnections are mainly used for interconnections using DAHDI or Local channels:
Name : it is the name which will appear in the outcall interconnections list,
Interface : this is the channel name (for DAHDI see DAHDI interconnections)
Interface suffix (optional) : a suffix added after the dialed number (in fact the Dial command will dial:
Context : currently not relevant
To use your DAHDI links you must create a customized interconnection.
Name : the name of the interconnection like e1_span1 or bri_port1
Interface : must be of the form
dahdi/[group order][group number] where :
group orderis one of :
g: pick the first available channel in group, searching from lowest to highest,
G: pick the first available channel in group, searching from highest to lowest,
r: pick the first available channel in group, going in round-robin fashion (and remembering where it last left off), searching from lowest to highest,
R: pick the first available channel in group, going in round-robin fashion (and remembering where it last left off), searching from highest to lowest.
group numberis the group number to which belongs the span as defined in the /etc/asterisk/dahdi-channels.conf.
if you use a BRI card you MUST use per-port dahdi groups. You should not use a group like g0 which spans over several spans.
For example, add an interconnection to the menu
Name : interconnection name Interface : dahdi/g0
Interesting Asterisk commands:
sip show peers sip show registry sip set debug on
When setting up an interconnection with the public network or another PBX, it is possible to set a caller ID in different places. Each way to configure a caller ID has it’s own use case.
The format for a caller ID is the following
"My Name" <9999> If you don’t set the number part of
the caller ID, the dialplan’s number will be used instead. This might not be a good option in most
Outgoing call caller ID¶
When you create an outgoing call, it’s possible to set the it to internal, using the check box in the outgoing call configuration menu. When this option is activated, the caller’s caller ID will be forwarded to the trunk. This option is use full when the other side of the trunk can reach the user with it’s caller ID number.
When the caller’s caller ID is not usable to the called party, the outgoing call’s caller id can be fixed to a given value that is more use full to the outside world. Giving the public number here might be a good idea.
A user can also have a forced caller ID for outgoing calls. This can be use full for someone who has his own public number. This option can be set in the user’s configuration page. The Outgoing Caller ID id option must be set to Customize. The user can also set his outgoing caller ID to anonymous.
The order of precedence when setting the caller ID in multiple place is the following.
- User’s outgoing caller ID
- Outgoing call
- Default caller ID