SIP REGISTER

VoiceGuide can register itself with the SIP provider of choice, resulting in VoiceGuide receiving calls directed to registered telephone numbers and being able to place outgoing calls using the registered accounts.

The VoIP registrations are specified in the Config.xml file, in section <VoIP_Lines>

<VoIP_Lines> contains two sections: <VoIP_Registrations> and <VoIP_Authentications>.

<VoIP_Registrations>  can contain multiple <VoIP_Registration> sections, and
<VoIP_Authentications>  can contain multiple <VoIP_Authentication> sections.

 

Section <VoIP_Registration>:

<VoIP_Registration>
<Protocol>SIP</Protocol>
<RegServer>ServerAddress</RegServer>
<RegClient>RegisteredClient</RegClient>
<LocalAlias>LocalAlias</LocalAlias>
</VoIP_Registration>

<Protocol> Leave as SIP
<RegServer> IP address of the registration server or the domain name of the registration server. If domain name is specified then HMP will resolve the domain name to IP address before issuing the registration request.
<RegClient> Client name is usually specified as:
AuthUsername@Realm or
AuthUsername@RegServer
Value of RegClient is sent in the From: and the To: fields of SIP Register messages.
<LocalAlias> Can be left blank, or a SIP URI in form of user@host can be specified. Value of LocalAlias is used in the Contact: field of SIP Register messages.

 

<VoIP_Authentication> holds information about the SIP digest authentication. It contains:

<VoIP_Authentication>
<Realm>Domain</Realm>
<Identity>AccountName</Identity>
<AuthUsername>AuthUser</AuthUsername>
<AuthPassword>AuthPassword</AuthPassword>
</VoIP_Authentication>

<Realm> The 'realm' for which this authentication applies.
If system is handling incoming calls only then it's recommended that this field is left blank, unless you are registering with multiple SIP servers. If registering with multiple servers then the "realm" used by the SIP server should be specified here. WireShark can be used to view 401/407/etc response contents to see realm setting in those responses.
It may be necessary to set this field if per-call authentication on outgoing calls is required.
<Identity> Account for which this authentication applies.
It is recommended that this field be left blank, unless you are registering multiple accounts/trunks and require a different authentication to be used for each account/trunk. If specified then this authentication entry will only be used if the Identity matches the To: field contents in the 401 or 407 response from the registration server. This field usually is in this format: sip:1010@10.1.1.11
<AuthUsername> Username used for authentication.
<AuthPassword> Password used for authentication.

 

The Dialogic HMP service must also be restarted after any changes to <VoIP_Registration> or <VoIP_Authentication> entries.
Dialogic HMP service restart is necessary to clear the old Registration/Authentication entries that have been previously loaded into HMP.
If Dialogic HMP service is not restarted then the previously loaded Registration and Authentication entries will take precedence.

WireShark can be used to confirm what SIP packets are exchanged between the SIP server and the VoiceGuide/HMP system. WireShark traces are usually necessary in determining causes of any registration failures.

SIP registration and authentication examples can be found in the Config.xml file. Information used for SIP registration is very simiar for all SIP switches/providers.

Below are some examples as well:

 

CallCentric (www.callcentric.com)

<VoIP_Lines>

<VoIP_Registrations>
<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>callcentric.com</RegServer>
  <RegClient>177711111111@callcentric.com</RegClient>
  <LocalAlias>177711111111@10.1.1.9</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
  <Realm></Realm>
  <Identity></Identity>
  <AuthUsername>177711111111</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>

</VoIP_Lines>

 

Asterisk

The registration config below demonstrates how VoiceGuide would register to accept calls to a particular Asterisk extension (ext 3000).

Asterisk was installed on another server. Asterisk server's IP address was: 10.1.1.11  VoiceGuide is installed on IP address 10.1.1.9

<VoIP_Lines>

<VoIP_Registrations>
<VoIP_Registration>
<Protocol>SIP</Protocol>
<RegServer>10.1.1.11</RegServer>
<RegClient>1010@10.1.1.11</RegClient>
<LocalAlias>sip:1010@10.1.1.9:5060</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>3000</AuthUsername>
<AuthPassword>1234</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>

</VoIP_Lines>

 

FreeSWITCH

The registration config below demonstrates how VoiceGuide would register to accept calls to a particular FreeSWITCH extension (ext 1010).

FreeSWITCH was installed on another server. FreeSWITCH server's IP address was: 10.1.1.11

Note that this is all that is required to allow multiple calls to extension 1010 to be all sent to VoiceGuide at the same time. The number of actual calls handled will only be limited by the number of VoiceGuide lines, so for example a 20 line VoiceGuide system still requires only one extension to be registered with the VoIP switch.

FreeSWITCH will send all calls to extension 1010 to VoiceGuide, regardless of how many ext 1010 calls VoiceGuide is currently handling.
You can of course register multiple extensions if you want VoiceGuide to run different services depending on which extension was called.

<VoIP_Lines>

<VoIP_Registrations>
<VoIP_Registration>
<Protocol>SIP</Protocol>
<RegServer>10.1.1.11</RegServer>
<RegClient>1010@10.1.1.11</RegClient>
<LocalAlias>1010@10.1.1.9</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
<Realm></Realm>
<Identity></Identity>
<AuthUsername>1010</AuthUsername>
<AuthPassword>1234</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>

</VoIP_Lines>

 

Skype Connect

<VoIP_Lines>

<VoIP_Registrations>
<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>sip.skype.com</RegServer>
  <RegClient>99051000000000@sip.skype.com</RegClient>
  <LocalAlias>99051000000000@sip.skype.com</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
  <Realm></Realm>
  <Identity></Identity>
  <AuthUsername>99051000000000@</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>

</VoIP_Lines>

 

BroadSoft

Many SIP providers use BroadSoft's platform.

<VoIP_Lines>

<VoIP_Registrations>
<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>sip.NSW.iinet.net.au</RegServer>
  <RegClient>0299998888@sip.NSW.iinet.net.au</RegClient>
  <LocalAlias>0299998888@10.1.1.9</LocalAlias>
</VoIP_Registration>
</VoIP_Registrations>

<VoIP_Authentications>
<VoIP_Authentication>
  <Realm></Realm>
  <Identity></Identity>
  <AuthUsername>0299998888</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>
</VoIP_Authentications>

</VoIP_Lines>