An interesting debate was started today during a new Callmanager Express install:
During lunch one of the junior engineers asked why anyone would use SIP firmware on the handsets instead of SCCP.
A member of our group suggested that the only reason he could think of was to do with call-transfer options available on each platform:
When setting up the Unified CME system the engineer has to select either blind or full-consult transfer mode and set it system wide. If you intend using SCCP handsets you’ll only get this option on softkey whereas with SIP image on same handsets you get both options on softkeys.
Now we agreed that most customers normally want consultative transfers (permanant blind transfer is just rude!) but what happens if you need to do a blind-transfer say to voicemail? Is going to SIP the only option?
CME Expert, Anthony Fear, came up with the answer :
“Setup the system up with full-consult transfers but when you need to do a blind transfer – hit ‘Transfer’ as normal, dial extension and when you hear ringing tone hit ‘Transfer’ again – Viola a blind transfer in SCCP!”
“Quite why Cisco don’t use the same ‘Xfer’ and ‘BlindXfer’ softkeys that you get in SIP Image is a mystery. But hey there you go!”
Apparently SIP firmware offers a ‘semi-blind transfer’ but no-one in the group knew exactly what this meant and so although this could be a legitimate reason to choose SIP over SCCP we decided it’s a pretty lame one!
We’ve been working on integrating Trixbox with CallManager Express (Full documentation to follow shortly) and stumbled across an issue where a user had accidentally discovered the MWI on and MWI off extention number on our CME system.
To his great amusement he was able to dial the mwi extentions and switch on or off the mwi indicator on any phone on the system. By default these special CME extentions are dialable – This lead us to investigate a way of making them non-dialable like intercoms. Well it turns out it’s quite simple. Simply change the number under each number from say 9000 to A9000 and then in Unity Express (or your alternative Voicemail System) simpy change the mwi extentions to the new numbers.
To change the MWI indicator numbers in Unity Express
Reverse Telnet to CUE module : service-module service-engine 0/1 session
Then enter global config:
, then enter:
ccn application ciscomwiapplication
then change the following lines and replace the ???? with your non-dialable mwi extentions:
parameter “strMWI_OFF_DN” “????”
parameter “strMWI_ON_DN” “????”
Viola – CUE can now switch on and off your mwi lights but pesky users cannot!
We were asked today to help with a Voicemail problem on a CallManager Express system. Turns out that since the installation of the VoIP telephone system, the voicemail module has only been recording callerID on an intermittent basis.
Anthony Fear (our resident CME expert) took up the challenge –
“testing showed that only extentions that were also registered on CUE were able to leave their CallerID, all other callers were recorded as “unknown number”. This turns out to be a default ‘FEATURE’ of CUE. In order for CUE to record any number presented to it you have to enable the feature, heres how:”
“Reverse Telnet onto the CUE module as normal and enter global config, then enter the command : voicemail callerid now all callers numbers will be recorded by the system (unless they withold them of course!).”
So there you go – easy when you know how!
We are trying to replace a Netgear ADSL router with a Cisco ADSL Router – we’ve used the GUI configuration tool to get it working but we don’t how to configure port forwarding like we had on the netgear! So we can forward smtp, http and such like to our Windows server. Can you help?
This depends on the model of router you have but basically if it’s an IOS sytle router then I’m guessing you have used something like the SDM to configure the router? I’m also guessing that you are using NAT (network address translation) on your router which is probably configured on a dialer0 interface. Have a look through the config and you should see a line like:
ip nat inside source list 100 dialer0 overload
If you do then you are use using NAT. You can configure port forwarding (PAT) with a command like:
ip nat inside source static tcp [internal ip] [port number] interface [interface with nat outside] [port number]
ip nat inside source static tcp 192.168.1.1 25 interface dialer0 25
be careful if using this type of command because if you have http enabled on your router then you won’t be able to forward port 80 to your server. You’ll have to disable it first!
alternatively if you have a fixed ip address on your adsl interface or you have a range of ip addresses on your adsl interface then you can do this:
ip nat inside source static tcp 192.168.1.1 25 188.8.131.52 25
where 184.108.40.206 is your static ip address or one of the range of addresses you have allocated by your ISP.
you can also leave out the port number to have everything passed directly to you internal this is useful for debugging or if your internal system has a firewall of it’s own!
ip nat inside source static tcp 192.168.1.1 220.127.116.11
Hope this help.
I was trying to add a new user on a client’s SBS 2003 server today – Did all the usual stuff, set the home directory, created a roaming profile, created an exchange mailbox but even after waiting a few minutes (well ok 60 mins to be exact – over the years I’ve seen some servers take up to 30 mins to finish creating the exchange mailbox) I noticed that the email addresses specified in recipient update policy were not being applied to the account.
I checked everything and double checked again. So then i tried removing the mailbox from the account in ADU&C and adding it back again. Still Nothing….
I started going through everything obvious and then going through the not so obvious:
Application Log – No Errors!
System Log – No Errors!
Sophos Log (ok by this point I was getting desperate) – Still No Errors!
It was at this point that I happened to spot that the same exchange alias name existed in another account “ahhhh! that must be it!” I said to myself – feeling rather chuffed that I’ve solved it…….made the necessary changes and put my feet up waiting for impending success – But NO! Still nothing!
So, of course, I started Googling but didn’t find much. Except a few distant entries that suggested restarting exchange services might help – “yeah right” I thought to myself. But figured I’d give it a go in the evening when everyone was off the server. So this evening I begrudgingly logged on to the Client’s server, navigated to the services app then stopped and re-started all the exchange services.
And yup you’ve guessed it – by the time I’d opened ADU&C the flaming email addresses had been generated and everything was working perfectly!
Sometimes you know – I really hate computers!
Question from Wellingborough Electrical
“Having Installed Business Contact Manager, when I open up Outlook it says that This Version of Business Contact Manager does not work with Exchange email accounts…..”
You have the old version of BCM. Download the latest version from Microsoft and re-install. This version has some other updates including a fix for this.
Question from Brown Dogs Chest Freezer
“My blogger rss feeds are not updating when adding new posts. 3 of the original entries are there but nothing added after that is appearing!…..”
Turns out it was problem with the settings in the ‘site feed’ page.
Site Feed URL should be your full http reference i.e. http://www.computerboffins.com/blog/atom.rss
Site Feed Filename is atom.xml
Site Feed Path should be the path on your ftp server – it had been put down as a http reference. Instead it should be the relative path from your ftp login. i.e. /computerboffins/blog
It is now resolved and working as intended!
I was recently asked by a client what all this VoIP stuff is about! So here’s a quick introduction.
What is VoIP?
VoIP (or Voice over Internet Protocol) is a great new way to make and receive phone calls using your computer network instead of your standard phone line. VoIP systems convert your phone calls into data that zips through your high-speed Computer network or Internet connection just like email. It comes out the other end like a regular phone call. Your callers will never know that it’s any different since it sounds just like a regular phone call.
Why Use VoIP?
Cost and Flexibility! With traditional phone systems, you predefine how many lines you require and pay rental on each line. When you have used up all your lines, callers get a busy tone. With an IP phone system there is no concept of lines as information flows over your computer network. So for example a single telephone line with ADSL Internet connection can handle about 7 concurrent calls with VoIP, saving over 700% compared to traditional PABX systems.
Furthermore because your phones use a computer network they no longer have to be physically located in the office. They can, for example, be located at employee’s home, perhaps a remote office or with a business partner such as outsourced IT department! Phones can be located virtually anywhere where there is an internet connection! But they can be called and used exactly as if they were connected to your office system.
Do I need expensive Phone Equipment?
No! VoIP systems are flexible meaning you can use standard household phones with an adapter or a business class IP phones with easier access to the advanced functions. Your not tied to manufacturer or a supplier so you can use any equipment you choose – you can even use a PC or laptop to make your calls instead!
I’ve been investing in some Cisco Equipment and training over the past few months – particularly the CallManager 4 systems because I believe they offer an excellent opportunity to tie some of our in-house skills together and also because I see an amasing opportunity to provide client’s with an excellent product that not only has an excellent feature set (Video Conferencing, Remote Handsets, Integrated Messaging) but also potentially a competitive advantage product by developing applications to go onto the handsets (but more about that in a future blog).
Now I’ve been recommending Cisco equipment to my client’s for years simply because of it’s unrivalled reliability (when was the last time you had to replace a faulty Cisco device on your network?) Ok I know they do fail, but compare it to any other manufacturer and you know it’s good stuff! But it never occurred to me that they might produce a decent Telephone system. Afterall some fo the buig name telephony manufactureres out there have been doing it for years – how could a networking company possibly compete? Well I’m happy to admit when I’m wrong.
Now this is just a personal opinion but let me make this point very clear – CallManager Express IS NOT A GREAT TELEPHONE SYSTEM – it lacks some basic features available on other systems, it’s expensive compared to other VoIP solutions out there and it’s a pig to administer – however as a complete solution incorporating network infrastructure, integrated communications, security, Qualify of Service and all the rest of that jazz that we all need – Cisco’s offering is quite simply the best solution out there.
Now don’t get me wrong there are better telephony solutions available and there are cheaper solutions out there (Asterisk PBX is the obvious choice here and one I’m familiar with so I can speak with some confidence on this matter) but everyone of them brings their own issues. The simple fact is that whatever VoIP solution you go for – Asterisk, Nortel, Microsoft?!? you’re going to need a reliable, configurable network infrastructure to put it on and what’s the obvious answer there? Cisco of course!
So take my advice and make your life easier and implement a full Cisco solution – you’ll thank me in the end!
I’ll cover some of the main advantages (as I see them anyway ) in a future blog – but the phone is ringing and I’d better get back to work demonstrating to my client’s why this CallManager Express system is so darn good!
Well after much discussion, toing and frowing and general wondering – we’ve finally decided to create a blog.
With the help of our friends at www.blogger.com we’ve setup an account, edited a template and posted out first Blog page.
The aim of this blog (as well as being a outlet for our general rantings about all things I.T.) is to provide a mechanism for us to provide a general question and answer forum to anybody who requires it – that includes potential, existing and never-to-be clients. So if you have a question and would like Free Technical Support – drop us a line. We aim to answer within 24 hours.
Computer Boffins Team