Create an interconnection

There are two types of interconnections :

  • Customized

  • SIP

Customized interconnections

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:

    <Interface>/<EXTEN><Interface suffix>
  • Context : currently not relevant

SIP interconnections

  • General, Signaling and Advanced tabs create the SIP peer information

  • Register tab creates the registration chain


in XiVO PBX Web interface slash “/” character is not supported in the password field.

DAHDI interconnections

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 order is 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 number is 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 Services ‣ IPBX ‣ Trunk management ‣ Customized

Name : interconnection name
Interface : dahdi/g0


Interesting Asterisk commands:

sip show peers
sip show registry
sip set debug on

Caller ID

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 cases. If you only need to set a number as an outgoing caller ID, you just have to put the number in the caller ID field like 0123456789.

Outgoing call caller ID

There are several behavior for the outgoing caller ID.

Use outgoing caller ID

When the internal 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 useful 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 useful for a user who has his own public number (DID number). This option can be set in the user’s configuration page. For this, the Outgoing Caller ID option must be set to Customize.


The user can also set his outgoing caller ID to Anonymous.

If you use a SIP provider trunk, and if your provider supports the RFC3325 for Anonymous calls, you have to set the Send the Remote-Party-ID option of your SIP trunk to PAI:

  1. Services ‣ Trunk management ‣ SIP Protocol ‣ Edit ‣ tab Advanced

  2. set parameter Send the Remote-Party-ID to PAI

With this option anonymous calls will be sent to your SIP provider with the RFC 3325 standard. Note that in this case, the P-Asserted-Identity SIP Header will contain the Outgoing caller ID number if set. Otherwise it will use the user’s internal caller id, which not a good idea. So you should configure a default caller ID in the outgoing call.

Order of precedence

The order of precedence when setting the caller ID in multiple places is the following.

  1. Internal

  2. User’s outgoing caller ID

  3. Outgoing call caller ID

  4. Default caller ID