Grouping Channels With Specific Sip Registrations When Running Dialogi

[jaytee]

One of our systems has 24 lines using Dialogic HMP 3.0. I've looked through the sample config.xml files for multi line systems, through the Voiceguide help file and searched the forum postings but I can't seem to find any info on how, for example, I could setup 4, 8 or 12 lines with one sip registration and 4 other lines with a second or possibly with more than two sip registrations in different combinations. I'd like to be able to route calls to specific Voip line registrations to channels that are setup to play different scripts if that's possible.

[SupportTeam]

You can have multiple SIP registrations - just add more <VoIP_Registration> entries to the <VoIP_Registrations> section and add corresponding </VoIP_Authentication> entries to the </VoIP_Authentications> section.

 

To setup limits of now many calls the system will handle from each registered source you will need to check in script how many calls are currently being handled from a particular registered source and if limits is exceeded then hangup/transfer/play+hangup/etc.

 

Use a 'Global RV' as a counter of how many calls from particular source are currently on system, and use $RV_DNS to see to what registered number the incoming call is directed to.

 

The msin script that answers incoming calls should look at the $RV_DNIS and then check the counter and then route call to particular script after incrementing the counter (or hangup etc).

The 'on-hangup' scripts would need to decrement the counters.

 

Please let us know if you have any further questions,

[jaytee]

I guess my first posting wasn't clear about my issue. What I'm trying to accomplish is to group channels based on a specific sip registration, e.g.: sip account 1000@192.168.1.20 registers to a SIP provider or PBX and routes calls to the first 4 licensed channels in Voiceguide.

 

I currently have it setup to work with Asterisk at the moment for all channels but not specific channels. What I'd like to do is have 4 channels of the type that show up in the sample configs which I've adapted take calls from a specific sip registration and have other channels take calls from other sip registrations if this is at all possible. It's obviously easy to control traffic to specific channels when using a Dialogic analog card by controlling the number going to a specific port on the card but how do I accomplish the same thing when it's SIP using Dialogic HMP? I know I can have multiple registrations but that doesn't help me unless there's a way to associate a specific VGS script with a specific SIP registration in the config.xml file.

 

below is a paste of a portion of one of my installations using Dialogic HMP 3.0 instead of a board. These are the "channels" I'm referring to.

<Channels>

 

<Channel>

<Device_Voice>dxxxB1C1</Device_Voice>

<Device_Network>iptB1T1</Device_Network>

<Device_Media>ipmB1C1</Device_Media>

<Protocol>IP</Protocol>

<script>C:\Accel\VGScript\Callscript1.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:\Accel\VGScript\Callscript2.vgs</Script>

<AllowDialOut>1</AllowDialOut>

</Channel>

<Channel>

<Device_Voice>dxxxB1C3</Device_Voice>

<Device_Network>iptB1T3</Device_Network>

<Device_Media>ipmB1C3</Device_Media>

<Protocol>IP</Protocol>

<script>C:\Accel\VGScript\Callscript3.vgs</Script>

<AllowDialOut>1</AllowDialOut>

</Channel>

[SupportTeam]

HMP will chose the first available 'port' when the incoming VoIP call arrives. There is no way to get HMP to direct calls from specific registrations to specific ports. In HMP a 'port' is just an abstract device.

 

Do you need specific ports to carry certain calls as you want to limit maximum number of calls from certain source, or do you want to run specific scripts for calls from certain sources, or is there another reason?

[jaytee]

Based on your last reply I would say I want to run specific scripts for calls from a certain source, the source being a call directed to an extension on a pbx that sends it's calls to a registered sip account from Voiceguide.

The samples files show the script file being referenced by a specific IP "channel". If calls are distributed randomly across multiple ports how do you route calls to different scripts based on sip registration? The online help does show any examples for that. I'm considering the idea of a "front end" call script that reroutes to other scripts based on $RV_DNIS but I'm not sure if that info is being passed via SIP to Dialogic and Voiceguide from Asterisk.

[SupportTeam]

how do you route calls to different scripts based on sip registration?

By using the information of th waht number thje caller was directed to. This information is usually stored in $RV_DNIS.

 

I'm considering the idea of a "front end" call script that reroutes to other scripts based on $RV_DNIS but I'm not sure if that info is being passed via SIP to Dialogic and VoiceGuide from Asterisk.

Can you post VoiceGuide's ktTel and vgEngine traces capturing an incoming call. This way we will be able to see what indial information is passed to the system.