2 Sip Registration And Connect Channel

[twlee]

1. Only one sip account the work. Can I make two all work?
2. When an incoming call is 1111@111.111.111.111 and a link to dxxxB1C1 script, If there is a way that when an incoming call is connected to 2222@111.111.111.111 to dxxxB1C2 script?
   • <CallerId> </Callerid> 를 이용할 수 있는지?

 

Examples are shown below. What happens if you modify the config.xml file?

 

<?xml version="1.0"?>

<VoiceGuideConfig>

<Devices_Dialogic>

<Channels>

<Channel>

<Device_Voice>dxxxB1C1</Device_Voice>

<Device_Network>iptB1T1</Device_Network>

<Device_Media>ipmB1C1</Device_Media>

<Protocol>IP</Protocol>

<Script>C:\ivrdata\script\MainARS.vgs</Script>

<AllowDialOut>1</AllowDialOut>

</Channel>

<Channel>

<Device_Voice>dxxxB1C2</Device_Voice>

<Device_Network>iptB1T2</Device_Network>

<Device_Media>ipmB1C2</Device_Media>

<Protocol>IP</Protocol>

<Script>C:\ivrdata\script\BusyARS.vgs</Script>

<AllowDialOut>1</AllowDialOut>

</Channel>

</Channels>

<Parms>

</Parms>

</Devices_Dialogic>

<VoIP_Lines>

<VoIP_Registrations>

<VoIP_Registration>

<Display></Display>

<Protocol>SIP</Protocol>

<RegServer>111.111.111.111:5061</RegServer>

<RegClient>1111@111.111.111.111:5061</RegClient>

<LocalAlias>1111@192.168.109.2</LocalAlias>

</VoIP_Registration>

<VoIP_Registration>

<Display></Display>

<Protocol>SIP</Protocol>

<RegServer>111.111.111.111:5061</RegServer>

<RegClient>2222@111.111.111.111:5061</RegClient>

<LocalAlias>2222@192.168.109.2</LocalAlias>

</VoIP_Registration>

</VoIP_Registrations>

<VoIP_Authentications>

<VoIP_Authentication>

<Realm></Realm>

<Identity>sip:1111@111.111.111.111:5061</Identity>

<AuthUsername>1111</AuthUsername>

<AuthPassword>****</AuthPassword>

</VoIP_Authentication>

<VoIP_Authentication>

<Realm></Realm>

<Identity>sip:2222@111.111.111.111:5061</Identity>

<AuthUsername>2222</AuthUsername>

<AuthPassword>****</AuthPassword>

</VoIP_Authentication>

</VoIP_Authentications>

</VoIP_Lines>

</VoiceGuideConfig>

[SupportTeam]

Are you using the 1 port HMP license?

 

You will only be able to register a single SIP service if using the single port HMP license.

 

It is possible to get a temporary 2 port HMP license from Dialogic. Please see: https://www.dialogic.com/en/products/media-server-software/hmp-software/hmp-license.aspx

 

Not sure if that license still limits you to registering a single SIP service only or whether you can register 2 SIP services.

 

 

 

When an incoming call is 1111@111.111.111.111 and a link to dxxxB1C1 script, If there is a way that when an incoming call is connected to 2222@111.111.111.111 to dxxxB1C2 script?

 

You would just use $RV_DNIS to jump to different script depending on what is the dialed number.

 

eg: all posts would point to the same 'routing' script. The 'routing' script would just contain one 'Evaluate Expression' module, that would have this as the expression:

 

"$RV_DNIS"

 

and then have paths like this:

 

on {1111@111.111.111.111} goto [C:\MyDir\MyScriptA.vgs|]

on {2222@111.111.111.111} goto [C:\MyDir\MyScriptB.vgs|]

 

or the expression can be:

 

Left("$RV_DNIS",4)

 

and paths can be:

 

on {1111} goto [C:\MyDir\MyScriptA.vgs|]

on {2222} goto [C:\MyDir\MyScriptB.vgs|]

 

 

So any port 'DNIS-routes' call to the script for that number.

 

Please let us know if you have any questions,

[twlee]

I was using two HMP license. Is it normal that I wrote the script?

[SupportTeam]

Are you referring to the Config.xml file?

 

Best way to check what VoiceGuide is reading in and how the registrations are processed is by looking at the VoiceGuide traces.

 

Please .ZIP up and post trace files from today that are in VoiceGuide's \log\ sub directory and we can then see how the registrations were processed.

 

Also a WireShark trace (www.wireshark.org) capturing system startup would let us see what is going on on SIP message level. If you can post the WireShark .pcapng capture then we can look at that as well.

 

The WireShak trace will let you see how many REGISTER requests HMP sent out, and the VoiceGuide ktTel and vgEngine traces will let you see if the setting were read in from Config.xml properly.