Dial Plans and International Gateways

Dial Plans

By default, VQ Conference Manager will create Call Identifiers prefixed by “700”. If a different prefix is required, this can be changed (see “VQ Conference Manager Configuration Options” document available from our Customer Portal - vqcomms.com); each new Call Id is the previous value +1 (so, default values would be 7001, 7002, 7003 etc.).

In many situations, more control over URIs and Call Ids is required; for that reason, generators have been added to the LDAP Configuration and Space Template definition pages.

The LDAP Configuration page supports the Auto-Increment (“AI”) keyword and the Space Template page supports both Auto-Increment and Random (“RND”) keywords.

When Call Id’s and URIs need to be generated (e.g., they are not driven by AD/LDAP Attributes), the “Auto-Increment (‘%AI(xx)%’) ” and “Random” (‘%RND(xx)%’) keywords can be used along with prefixes/postfixes) to create Call Ids and URIs (of given length; xx) with values that conform to the dial-plan requirements of the deployment.

When defining the length component for generators (AI and RND), please ensure you specify a length value (the generated values are leading zero filled) that will give you sufficient values – for example, length 4 would give a number range of 0…999 (one thousand values). This might sound a lot but isn’t; you should be considering 6 or 7 digits. Please ensure the number ranges generated work with your dial-plan.

Please also note that the generator functions do not currently ‘wrap-around’ and re-use unused numbers. Prefix chains can be defined so the values generated can go thru a list of defined ranges. For example, %66,67AI(6)% would generate values 66000000 thru 66999999 and 67000000 thru 67999999.

Once the generator value has wrapped around over the available set of values, it’ll continue incrementing and therefore (in this example) start generating 7-digit values which will be appended to the last prefix (in this case 67) giving a 9-digit result rather than a 8-digit result.

For more details on Generators, please see document “Generator keywords for URI and URI values”; available for download from our Customer Portal - vqcomms.com/login.

International Gateways

If your deployment is on a larger scale, you might have participants dialing into calls using different SIP trunks.

Please make sure you make tests calls through each SIP trunk connected to your CMS. We had one instance, where, for example, inbound calls (calls to CMS) in most cases worked but calls from a particular region connected as far as the CMS IVR but weren’t able to join a call. The root cause of the problem turned out to be that all the failing calls were routed through the same SIP trunk and the SIP server on the failing SIP trunk was an older version of Asterisk. The Asterisk SIP server didn’t support Video or BFCP in the SIP SDP headers and ignored the RE-INVITES containing them issued by CMS as part of the IVR exchange. In this example, there were two solution options:

  1. Update the Asterisk SIP server to a later version that supported Audio, Video and Content in the RE-INVITE SDP headers
  2. Update the CMS version from 2.2.28 to CMS 2.3; CMS 2.2.28 always issued a RE-INVITE with all media types; CMS 2.3 only issued the RE-INVITE with the media headers already selected (in this case, Audio only)