Outbound calls on VoIP lines

When placing an outgoing call the IP address or domain of the Switch/PBX which will be routing the call needs to be specified.

For example, when dialing another extension on the VoIP Switch/PBX the number dialed would be in the format: extension@ip_address_of_voip_switch

eg:

204@10.1.1.11

According to how the VoIP switch is set up, the "extension" could also be an external phone number.

when dialing through a VoIP provider, often the VoIP provider's SIP server domain/address needs to be specified. eg:

15551231234@callcentric.com

 

CallerID / "From:" field

When making an outgoing call on a VoIP line it is usually necessary to specify the CallerID to be used on the outgoing call. This advises the switch relaying the VoIP call as to which account/subscriber is making the call.

The CallerID is set in the SIP header's "From:" field.

The CallerID ("From:" field) on outgoing calls can be specified using the <CallerID> tag in the call's options, in this format:

<CallerID>accountnumber@voipprovider</CallerID>

eg: to place a call through a PBX which is installed on a server with IP of 10.1.1.11, and with which the user/extension 2000 has been registered by VoiceGuide, the following entry would need to be included in the options field:

<CallerID>2000@10.1.1.11</CallerID>

Sometimes just the user ID (or the registered telephone number) is sufficient, eg:

<CallerID>5625551234</CallerID>

 

And to place a call through a VoIP provider with which the account was registered by VoiceGuide, an entry that includes the domain name of the provider needs to be used. eg: with CallCentric, this would need to be used:

<CallerID>5551234@callcentric.com</CallerID>

and with Skype SIP Connect, a CallerID definition similar to this would need to be used:

<CallerID>990510045673275@sip.skype.com</CallerID>

 

Authentication

VoIP switch will often require to authenticate the user before allowing the outbound call to be made.

IP-based authentication does not require any additional configuration in VoiceGuide.

If Digest authentication is used then it is necessary to specify the Registration and Authentication parameters.
Please see here: VoIP Line Registraton

 

User Specified Headers

If specific headers need to be added/modified in the outgoing SIP INVITE packet then these can also be specified in the call's options when loading the outgoing call.
To add a SIP header a "<sip-header>" entry needs to be specified as part of the call's options.
eg: to add a "MyHeader" header, this entry could be added:

<sip-header>MyHeader: "somevalue" <sip:123456@sip.router.com></sip-header>

eg:
"sip-session-expires" and "sip-min-se" entries can also be specified like this:

<sip-session-expires>999</sip-session-expires>
<sip-min-se>55</sip-min-se>

(or these header entries could be just added using the "<sip-header>" approach)

 

Multiple headers can be specified. Below is a valid options entry:

<CallerID>12121212@sip.mycarrier.com</CallerID>
<sip-session-expires>999</sip-session-expires>
<sip-min-se>55</sip-min-se>
<sip-header>Remote-Part-ID: "Flowroute" <sip:123412341234@sip.flowroute.com;party=calling;screen=yes;privacy=on</sip-header>
<sip-header>X-Tag: mySpecialCallTag123</sip-header>
<sip-header>Diversion: <sip:333444555@sip.mycarrier.com></sip-header>

 

Where to set Call Options

Call options can be set the "Call Options" text box in the Outbound Call Loader app, or in the <CallOptions> field in the OutDial XML file, like this:

<CallOptions>
  <CallerID>12121212@sip.mycarrier.com</CallerID>
  <sip-session-expires>999</sip-session-expires>
  <sip-min-se>55</sip-min-se>
  <sip-header>Remote-Part-ID: "Flowroute" <sip:123412341234@sip.flowroute.com;party=calling;screen=yes;privacy=on</sip-header>
  <sip-header>X-Tag: mySpecialCallTag123</sip-header>
  <sip-header>Diversion: <sip:333444555@sip.mycarrier.com></sip-header>
<CallOptions>

 

WireShark can be used to verify that the headers have been set in the outgoing INVITE packets as expected.