How Will I Call Automatically Through The Sip

[elastix11]

Hi to everybody,

 

I've installed HMP3 for windows edition and VoiceGuide_7.2.3.exe. And also changed the C:\Program Files\VoiceGuide\conf\config.xml

but i dont know if it is true or not. Will i erase the example lines? I've configured only the lines below. Is it true and enough to work?

 

1) conf.xml It is true or not ? Will i use only this part in the xml or will i keep the other lines in xml?

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display>VOicent</Display>

<Protocol>SIP</Protocol>

<RegServer>192.168.3.120</RegServer> This is my asterisk SErver

<RegClient>5000@192.168.3.120</RegClient>

<LocalAlias>5000@192.168.3.41</LocalAlias> This is my windows machine that installed the voiceguide

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>Voicent</Display>

<Realm>192.168.3.120</Realm>

<Identity>sip:5000@192.168.3.120</Identity>

<AuthUsername>5000</AuthUsername>

<AuthPassword>veribilim5000</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

 

2) How will i start calling process. I've load the numbers in outbound call loader. But i can't see any start button. It dialing times module. Will i use thispart ? Also I've tried to make a script.

 

I will be happy if anybody help me .. Have a nice day . Thank you very much.

Config.zip

[SupportTeam]

The registration section should be more like this:

 

<VoIP_Registrations>

 

<VoIP_Registration>

<Display>VOicent</Display>

<Protocol>SIP</Protocol>

<RegServer>192.168.3.120</RegServer>

<RegClient>5000@192.168.3.120</RegClient>

<LocalAlias>5000@192.168.3.41</LocalAlias>

</VoIP_Registration>

 

</VoIP_Registrations>

 

<VoIP_Authentications>

 

<VoIP_Authentication>

<Display>Voicent</Display>

<Realm>asterix</Realm>

<Identity></Identity>

<AuthUsername>5000</AuthUsername>

<AuthPassword>veribilim5000</AuthPassword>

</VoIP_Authentication>

 

</VoIP_Authentications>

 

We edited the Config.xml file. Please use the attached Config.xml on your system, then after system startup please try placing call into the system (ie. to the extension registered by VoiceGuide).

 

Please post VoiceGuide traces of system startup. The traces are located in VoiceGuide's \log\ subdirectory.

If the system does not answer your call then please post the WireShark trace as well which captures the system startup, and hence the SIP registration process (see www.wireshark.org). We can then see what happened during SIP registration.

 

Once incoming calls are working we can move onto making outbound calls.

Outbound calls are made automatically by VoiceGuide once their activation time is reached. (calls with no activcation time specified are made immediately)

Config.zip