Contents

VoiceGuide Help
1. Introduction
Welcome To VoiceGuide
Which version to use
2. System Configuration
System Requirements
Installing v7.x - Dialogic
Installing v7.x - VoIP / HMP
Installing v6.x - Dialogic
Installing v5.x - Voice Modems
Installing v5.x - CAPI compatible cards
Installing v5.x - Dialogic
Installing v5.x - Dialogic Wave Drivers
Text To Speech
Call Transfers and Conferencing
ODBC Data Sources
End of Call Detection
Distinctive Ring Detection
T1/E1 ISDN Configuration
T1/E1 RobbedBit/CAS/R2 Configuration
VoIP Line Registration
Command Line Options
Registering VoiceGuide
Unique System Identifier
3. Script Design
Introduction
Graphical Design Environment
Module Types
Paths
Result Variables
Call Start
Call Finish
Multilanguage Systems
Protected Scripts
Sound files
Testing Scripts
4. Modules Reference
Play
Record
Get Numbers
Say Numbers
Transfer Call
Send Phone Message
Send Pager Message
Send Email
Database Query
Run Program
Time Switch
Evaluate Expression
Run VB Script
Send DDE Command
Hangup Call
5. Fax
Introduction
6. Voicemail
Introduction
Voicemail System Manager
Voicemail Menus
Message Lamps
7. Outbound Dialing
Loading Numbers to Call
Detect Call Answer
Outbound VoIP calls
Predictive Dialers
External Database Source (v7)
8. Speech Recognition
Introduction
Grammars
Install LumenVox
9. Logs
Script Logs
Call Detail Records (CDRs)
10. ActiveX / COM Interface
Admin_TraceLogAdd
Dialer_MakeCall
Dialer_OutDialQueAdd
Bridge_Connect
Bridge_Disconnect
Line_Hangup
Line_Pickup
Play_Start
Play_Stop
Record_Stop
Record_Start
Record_2Lines_Start
Run_ResultReturn
RvGet
RvGet_All
RvGet_AllXml
RvSet
RvSet_RvList
Script_Gosub
Script_Goto
Script_Return
Serial_Tx
Vm_Event
Vm_VmbConfig_Get
Vm_VmbConfig_Set
11. PBX Integration (CTI)
Inband Signaling
Ericsson MD110 Voicemail Interface
Legal Information
Copyright & Disclaimer

 
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VoiceGuide can register itself with the VoIP 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> contains:

<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> Any string is OK here. Value of LocalAlias is used in the Contact: field of SIP Register messages.

 

Section <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
<Identity> Account for which this authentication applies. Can usually 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.
<AuthUsername> Username used for authentication.
<AuthPassword> Password used for authentication.

 

SIP registration and authentication examples can be found in the Config.xml file. Here are some more examples:

 

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

</VoIP_Authentications>
</VoIP_Lines>

 

Free World Dialup (www.freeworlddialup.com)

<VoIP_Lines>
<VoIP_Registrations>

<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>fwd.pulver.com</RegServer>
  <RegClient>123456@fwd.pulver.com</RegClient>
  <LocalAlias>123456@10.1.1.9</LocalAlias>
</VoIP_Registration>

</VoIP_Registrations>
<VoIP_Authentications>

<VoIP_Authentication>
  <Realm>fwd.pulver.com</Realm>
  <Identity>sip:123456@fwd.pulver.com</Identity>
  <AuthUsername>123456</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>
</VoIP_Lines>

 

 

Packet8

<VoIP_Lines>

<VoIP_Registrations>

<VoIP_Registration>
<Protocol>SIP</Protocol>
<RegServer>vo.packet8.net</RegServer>
<RegClient>0453101411@vo.packet8.net</RegClient>
<LocalAlias>MyName</LocalAlias>
</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>
<Realm>vo.packet8.net</Realm>
<Identity></Identity>
<AuthUsername>0453101411</AuthUsername>
<AuthPassword>Password</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>

</VoIP_Lines>

 

 

iptel.org (www.iptel.org)

<VoIP_Lines>
<VoIP_Registrations>

<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>iptel.org</RegServer>
  <RegClient>MyIptelAccountName@iptel.org</RegClient>
  <LocalAlias>MyIptelAccountName@10.1.1.9</LocalAlias>
</VoIP_Registration>

</VoIP_Registrations>
<VoIP_Authentications>

<VoIP_Authentication>
  <Realm>iptel.org</Realm>
  <Identity>sip:MyIptelAccountName@iptel.org</Identity>
  <AuthUsername>MyIptelAccountName</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>
</VoIP_Lines>

 

voxalot (www.voxalot.com)

<VoIP_Lines>
<VoIP_Registrations>

<VoIP_Registration>
  <Protocol>SIP</Protocol>
  <RegServer>voxalot.com</RegServer>
  <RegClient>123456@voxalot.com</RegClient>
  <LocalAlias>123456@10.1.1.9</LocalAlias>
</VoIP_Registration>

</VoIP_Registrations>
<VoIP_Authentications>

<VoIP_Authentication>
  <Realm>voxalot.com</Realm>
  <Identity>sip:123456@voxalot.com</Identity>
  <AuthUsername>123456</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>
</VoIP_Lines>

 

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>callcentric.com</Realm>
  <Identity>sip:177711111111@callcentric.com</Identity>
  <AuthUsername>177711111111</AuthUsername>
  <AuthPassword>Password</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>10.1.1.11</Realm>
<Identity>sip:1010@10.1.1.11</Identity>
<AuthUsername>1010</AuthUsername>
<AuthPassword>1234</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>
</VoIP_Lines>

 

BroadSoft BroadWorks platform (www.broadsoft.com) : iinet.net

iinet.net's VoIP service is built using BroadSoft's BroadWorks platform. Here are the registration details used:

<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>iinetphone.iinet.net.au</Realm>
  <Identity>sip:0299998888@sip.NSW.iinet.net.au</Identity>
  <AuthUsername>0299998888</AuthUsername>
  <AuthPassword>Password</AuthPassword>
</VoIP_Authentication>

</VoIP_Authentications>
</VoIP_Lines>

 

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