Jabber SIP URI Dialling Check List

Quick check list for enabling calls to SIP URIs in Jabber:

1. In LDAP Directory configuration set Directory URI to "mail" (i.e. use email address for URI).
2. Configure a SIP Profile for use by Jabber with "Use FQDN in SIP requests" & "Allow presentation sharing using BFCP" enabled. Any SIP Trunk carrying URI calls will also need a SIP Profile with "Use FQDN in SIP requests" enabled.
3. In Enterprise Parameters set the Organisation Top Level Domain & Cluster FQDN (e.g. example.com & *.example.com).
4. In Enterprise Parameters set the Directory URI Alias partition.
5. In CallManager Service Parameters check that SIP Max Incoming Message Size is >= 11000.
6.  On the SIP trunk to the VCS Control/Expressway-C set Calling and Connected Party Info to "Deliver URI and DN".
7. In jabber-config.xml under <Policies> put <EnableSIPURIDialling>true</EnableSIPURIDialling> & <DirectoryURI>mail</DirectoryURI>.

Alphanumeric SIP URIs for Rendezvous Conferences

Pre-Requisites

• Refer to the Cisco Unified Communications Manager with Cisco Expressway (SIP Trunk) Deployment Guide for the underlying configuration required to get SIP trunks/neighbour zones configured between CUCM & the Expressway-C or VCS-C.
• Refer to the Cisco TelePresence Conductor with Unified CM Deployment Guide for the underlying configuration required to integrate TelePresence Conductor & Server with CUCM.

The combination of TelePresence Conductor & Server integrated with CUCM allows both ad hoc and rendezvous video conferences. However there are some limitations in that currently (CUCM v10.5) can only handle alphanumeric URIs on directory numbers. To direct calls to TelePresence Conductor requires using route patterns, which have the following limitations:

• Route patterns only handle simplified regular expressions for phone numbers, so you can't specify an alphanumeric string.
• SIP route patterns work on IP addresses or domain names (e.g. 1.2.3.4 or test.com) so you can't specify a whole URI such as test@test.com.

To get round this you can create unassigned DNs with a directory URI configured, that then call forward all to the appropriate route pattern to reach a conference. This has the advantage of requiring no additional equipment and minimal configuration.
However if you have Expressways or VCSes you can offload the call routing to them instead and thus use a proper SIP URI to dial into rendezvous conferences. This way allows external participants to join without ever having to hit CUCM for call routing, but requires more configuration.

Configuration Overview 1
First create a DN, as it is unassigned make sure that active is ticked:

Don't forget to fill out the appropriate directory URI & then set the call forward all destination to the route pattern that directs calls to TelePresence Conductor:

Now you can join a rendezvous conference via numeric dialling or SIP URI.

Configuration Overview 2
First configure a SIP neighbour zone on the Expressway-C or VCS-C with all the CUCM servers as peer addresses:

You'll need a DNS A record that resolves to the TelePresence Conductor's rendezvous IP address so that we can configure a neighbour zone to the Conductor:

Note that if you have TLS verify mode on, the Conductor's certificate must have the rendezvous FQDN as a Subject Alternate Name (SAN); it's easiest to just turn TLS verify off.

Next configure a wildcard (*.*) SIP route pattern in CUCM pointing to the Expressway-C to direct calls to unknown URIs to it via a SIP trunk:

Now we need a search rule on the Expressway-C or VCS-C to direct calls to the rendezvous URI to the Conductor:

Finally Conductor needs a suitable conference alias to create or add participants to a rendezvous conference:

By also having a numeric conference alias that shares the same conference name as the alphanumeric alias, we can allow phones that can't dial URIs to also join the same conference via dialling a pilot number.