Outbound calls on VoIP lines
When placing an outgoing call the IP address or domain of the destination or Switch/PBX or VoIP Provider relaying the call needs to be specified.
For example, when dialing another extension on the VoIP Switch/PBX the number dialed would be of the format: extension@ip_address_of_voip_switch
According to how the VoIP switch is set up, the "extension" could also be an external phone number.
when dialing another number through a VoIP provider, the VoIP providers SIP server domain/address needs to be specified: telnumber@sipprovider
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 on outgoing calls can be specified using the <CallerID> tag on the Options field when loading the outgoing call, and is in this format: <CallerID>accountnumber@voipprovider</CallerID>
For example, to place a call through a FreeSWITCH PBX which is installed on a server with IP of 10.1.1.11, and with which the user/extension 1010 has been registered by VoiceGuide, the following entry would need to be placed in the Options field:
Sometimes just user ID (or the registered telephone number) is sufficient, eg:
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:
with Skype SIP Connect, a CallerID definition similar to this would need to be used:
Also, on outbound calls the VoIP Switch will often require to authenticate the user before allowing the outbound call to be made.
To allow HMP to process the authentication request it is necessary to specify the VoIP Line registration and authentication parameters.
Please see the VoIP Line Registraton section.
In particular, the
If specific headers need to be added/modified in the outgoing SIP INVITE packet then these can also be specified in the Call Options field when
loading the outgoing call. To add a SIP header a "<sip-header>" entry needs to be specified in the Call Options field.
eg: to add a "MyHeader" header, this entry could be added to the Call Options field:
<sip-header>MyHeader: "somevalue" <sip:firstname.lastname@example.org></sip-header>
"sip-session-expires" and "sip-min-se" entries can also be specified, like this:
(or these could be just added using the "<sip-header>" approach)
Multiple headers can be specified. Below is a valid Call Options entry:
<sip-header>Remote-Part-ID: "Flowroute" <sip:email@example.com;party=calling;screen=yes;privacy=on</sip-header>
The WireShark protocol analyzer can be used to capture the traces of the actual SIP messages sent.