Cisco Unified Communications Manager

Yesterday evening I attended a focus group on UCM (let’s just say Call Manager) and today, a roadmap session on the product. One thing that’s clear is that Cisco is moving ahead with their ideas about how customers would like to communicate and how phone systems administrators would like to manage, operate and maintain their systems.

And for the most part, they’re right. And leaving us (and others like us) in the dust.

I need to look over my notes and see what I’m allowed to write about as far as upcoming releases (NDA). The next release of CM 6.x is the one that the Cisco guys are saying is “the next 4.1.3” — in other words, the next rock-solid, stable release that customers are going to move to and sit on for a while. (Few are entering the 5.x series of the software, for a variety of good reasons.)

I believe Penn State is going to want to use this release, due in January, to make a decision: stick with Cisco and move ahead, or re-evaluate the competition? Sitting on 4.1.3 seems unwise given what’s currently out there, which is far more maintainable, feature-rich, robust… honestly, seeing what is to come but also what is available right now, I feel like I’m administering a legacy phone system with CM 4.1.3.

More to come. The exhibition floor opened up an hour ago and I’m getting ready to go down there and check it out. The show floor will be open for the rest of the conference and should provide plenty of interesting demos and information, as well as eye-candy and more swag.

Industry special-interest group: education

Hot topics at today’s special-interest group meeting for the education industry (included K12 and higher ed.; about 40 people from 20-25 institutions attended)

  • Organizations are plagued with deteriorating 7940s and 7960s–specifically, the hookswitches are wearing out. Cisco recommends proper care and cleaning of the IP phone.
  • Wireless and mobility are at the top of everyone’s list. Not much adoption of the 7920/7921 wireless IP phones. One university has deployed a small number of Vocera communications badges. (Click on the Hospitality guy. Remind anyone of Star Trek communicator badges?)
  • People are implementing emergency notification with home-grown apps and Berbee Informacast (which Penn State has out in very limited deployment). Many want to do SMS paging (text messaging); PSUTXT is a working model. Cisco’s IPICS may be a contender against Informacast.
  • 911: all over the board. No wonderful solutions offered by anyone present; I wish this topic had gotten a lot more floor time.
  • Voicemail: Interactive Intelligence’s Messaging Interaction Center (was Communite) can do unified messaging with IMAP systems, not just Exchange. A competitor to Cisco Unity.

CIPTUG 2007: Monday morning and loaded with swag

Signing in for CIPTUG this morning at the Austin Convention Center was a high-tech experience: type your e-mail address, and here’s your USB drive with all the presentation materials loaded, and a backpack full of swag. Citrix seems to have taken an early lead at this conference in the distribution of promotional materials.

We’re waiting for the educational industry special interest group meeting which is to start at 10:30am.

The convention center is impressive and stands out, along with a few new-looking hotels, in an area of the city which is otherwise marked mostly with old-looking bars and restaurants. Down the street there’s a construction crane doing its thing. It seems like the area is undergoing some nice upgrades.


I’ll be at the CIPTUG (Cisco IP Telephony Users’ Group) annual conference next week in Austin, TX. This is my first year as a member of CIPTUG (though others from Penn State have been members for several years) and my first time attending the conference. I hope to do some on-site blogging as well as review parts of the conference after the event.

I look forward to rubbing shoulders with some fellow Cisco VoIP admins, especially regulars from the cisco-voip mailing list, which, by the way, I highly recommend for general discussion and helpful advice (mostly on CallManager).

LDAP Directory Search and Dial for Cisco IP phones

(Originally posted December 14, 2005)
The stock “Corporate Directory” function in CallManager relies on user information found either in Active Directory or Cisco’s DCD. Our official directory information is kept in an enterprise LDAP server, so I put together a simple Perl CGI to take search parameters from phone keypad (text) input, query the LDAP server, and return search results to the phone, providing name, department, and phone number.

The phone number comes back from the LDAP search in the format “+1 814 555 1212” so if we want to have a Dial hotkey available, the number first has to be formatted according to the system’s dial plan. The University’s dialing plan is not too complicated, so parsing out a dialable number required only a few regexps.

There is a toggle switch in the code to allow or disallow returning of entries with no phone number in the directory. In a way, I like having entries returned even if there is no phone number. If I am searching for John Doe, at least I will see that he exists in the system, but does not have a permanent phone assignment.

Download the scripts from repos/phoneldap.

(Update on October 18, 2007)

I updated the files in the repository. There have been some changes since 2005, including a new departmental lookup function. I’m not going to go into detail about each file; start with dirmenu.txml. (Point the Directories URL in CallManager to this file.) Setup of the web server options is the same as for the weather and RSS scripts.

Here’s what it looks like…

Directories Menu screen
Directories Main Menu
Entering a name to search
Entering a name to search
Results page with Dial softkey
Search results with Dial softkey

U. Oslo visit and open-source VoIP

This morning, four of my colleagues and I met with three IT professionals from the University of Oslo who are on a five-day tour of US east-coast universities and companies with large VoIP installations. They’re planning an upgrade of their 11,000-telephone deployment to voice-over-IP and we offered some insight, tips, warnings, and random thoughts on deploying VoIP at a large university.

They, like many current entrants into VoIP, are interested not just in open standards (SIP) but also in open source.

Is it true that a large institution like Penn State needs a large company like Cisco to provide and support a voice solution? It’s hard to argue against. Cisco provides excellent support. But what we don’t get is the opportunity to dig into the software code, make our own customizations and hack. From a support standpoint, this is a good thing. If we want to hack at the system, as we could with Asterisk, we lose a great deal of support. Significant tradeoff.

What’s out there right now for open-source VoIP PBX? I know only of Asterisk, sipX, Pingtel’s SIPxchange, and maybe SER/OpenSER (but these are just SIP proxies and don’t offer full PBX functionality on their own). I don’t believe any of these products or the companies behind them will be of significant interest to large VoIP customers until they look more like RedHat, for example–based on open-source but structured like a real enterprise solutions provider.

How to set up XMeeting for use with Penn State’s videoconferencing systems

XMeeting, the Mac OS video conferencing software I previously reviewed here, works very well with Penn State’s mostly-Polycom video systems. Here’s how to set it up.

  1. Start by reviewing this H.323 video systems overview written by TNS’s video group.
  2. Open the XMeeting Preferences panel. We’ll go through the tabs from left to right.

    1. General: Enter your name or user ID. This will be sent to the video bridge or remote site and display under your image, so don’t enter something silly here. You might consider including the abbreviation of your campus or department.

      Automatically accept incoming calls if you want to do so.

    2. Appearance: Everything here is up to you.

    3. Accounts: Click the + under H.323 Gatekeeper Accounts to set up a new gatekeeper.

      Account name: or your choice

      Gatekeeper host:

      User alias/ID: You can enter any five-digit identifier that is not already in use on the gatekeeper. Room systems use their ISDN or conference phone number; I chose to use the last five digits of my desk phone number. This is the number that others will use to “dial” you for a videoconferencing session.

      Phone number: same as above (five digits)

      Password: leave blank


    4. Locations: You can edit the Default Location or set up a new one. I just edited the default.

      First, a note. The H.323 protocol uses a lot of dynamic UDP ports and it can be really tricky to get it to work through a firewall or NAT. There are settings here for going through a NAT but don’t count on it working! Your best bet is to have your Mac on a public, non-firewalled IP address. Scary, I know.

      That said, let’s look at the sub-tabs for the Default Location.

      1. Network:

        Bandwidth limit: No limit

        NAT Traversal: Try different settings. I ended up choosing Use IP address translation and Automatically get external IP address. I believe this does help with NAT traversal by sending the external IP of the NAT in the H.323 headers, rather than the internal IP address.

        Firewall settings: I lowered the ranges to 30000-30010 and 5000-5099 so that I could open these ranges on the Mac firewall.

      2. H.323: Click the box next to Enable H.323. I cleared the boxes next to Enable H.245 Tunnel and Enable Fast Start. For Gatekeeper Account, choose the account you set up just a minute ago.

      3. SIP: Do not enable SIP.

      4. Audio: I used the default audio settings with uLaw as the first preference and ALaw as the second. Packet Time is also Default.

      5. Video: Click the box next to Enable Video. I set the frame rate to 30. You can lower this on a slow connection.

        The video codec selection is important. You may have to come in to this screen from time to time and change settings.

        H.264: works only with point-to-point XMeeting-to-XMeeting. It may work with other endpoints but doesn’t work with Polycoms (i.e. all of Penn State’s videoconference rooms).

        H.263: widely accepted. Looks pretty good. Works on all point-to-point calls I’ve tried, but does not work with the video bridge.

        H.261: the oldest and lowest-resolution codec, but most compatible. You have to select H.261 (and only H.261) when connecting to the video bridge (MCU).

        My normal selection is to just disable H.264 and then enable H.263 and H.261, in that order. When I connect to the bridge for a multipoint call, I disable H.263.

        I checked the box next to Enable H.264 Limited Mode.

      That’s it for the Locations tab.

    5. Audio: Choose your audio device from the list. It’s best to use a headset because otherwise you’ll get bad echo. If you get a USB or Bluetooth headset, once you’ve set it up you’ll have to select it from the dropdown lists.

      Uncheck Enable Silence Suppression and check Enable Echo Cancellation. These worked best for me.

    6. Video Input: Here you can choose which devices you’ll want to use for sending video. I have all three enabled. The Live Camera Module is normally what you’d use, but you can also send your desktop with the Screen Module or a static picture with the Still Image module.

    7. Address Book: Default settings.

  3. Hit Apply. XMeeting should register with the gatekeeper and display a green dot and the word Idle. You can press Command-I to get a status window verifying your registration and your directory (phone) number.

  4. Before you go on, if you’re using the Mac OS X firewall, go add the following two rules to the firewall for XMeeting to send/receive data and call setup information:

    H.323 call setup: TCP and UDP port 1720
    XMeeting data: TCP ports 30000-30010 and UDP ports 5000-5099

  5. Make a test call to the loopback: 39344. This will test two things: your audio/video (you should see and hear yourself) and your gatekeeper registration. If the call doesn’t complete by phone number, try dialing by IP address: If that works, go back and check your gatekeeper settings and verify that XMeeting registered.

  6. Make a test call to the video bridge: 2222test. Remember that you will have to disable all video codecs except H.261 in order for video to work.

You can utilize the Apple Address Book to store the names and numbers of room systems or other desktop video users. A directory of Penn State’s public videoconference rooms is available here; click the link on each record and look for PSU Dial Number to see the directory number you should use to connect to the room.