Found a strange issue after switching to Asterisk v11, the BLF buttons on our Panasonic SIP phones stopped working.
Eventually thought it was worth trying setting the ‘presence server address’ which previously hadn’t been set (and in Endpoint Manager is set to blank) to the PBX and the lights immediately started working. Much simpler solution than I anticipated.
SIP Trunking is a great option to lower the cost of your phone calls. We installed a new Avaya IP Office 500 phone system at the beginning of last year, so of course I was keen to get VoIP setup quickly through 2talk to cut the cost of calls going over our ISDN lines.
The Avaya system doesn’t seem to be particularly common in New Zealand so I couldn’t find much in the way of resources about setting up a SIP trunk on the IP 500.
A year on and we have used SIP Trunking with 2talk for the majority of our outgoing calls. Here is a configuration guide with the settings I’m using. If you know a better way of doing it, please let me know!
2talk & Firewall Configuration
Make sure your 2talk account is setup for SIP trunking, with your firewall configured to forward SIP traffic through to the phone system.
We are using the 2talk Plus SIP Trunking service, (trunk.plus.2talk.co.nz which uses the IP 126.96.36.199). Works great in that we can lock down the firewall to that one IP, helping prevent SIP fraud and spam.
Our firewall forwards the following traffic through to the phone system:
TCP 5060, UDP 5060
UDP 49152 – 53246
IP Office Line Configuration
1. Fire up the IP Office Manager and add a new SIP Line to the line groups. Here are the settings I used:
(Note, leave the ITSP domain name as 2talk.co.nz if you aren’t using 2talk Plus)
2. Transport Tab
3. Under SIP URI, add at least a URI for your pilot number.
Click add, set the Local URI & Contact to your pilot 2talk number, such as, 03281XXXX.
Display name can be set to whatever you like, I have it set to Use Internal Data.
A sidenote on Line Groups
We have a bunch of ISDN & SIP channels. All of our lines are set to incoming group 0. ISDN lines are set to outgoing group 0, my pilot SIP URI is set to outgoing group 1, and the rest of the SIP URI’s are set to outgoing group 9.
This lets us route calls nicely – we want all incoming calls to be dealt with the same using Incoming Call Routing, so they all use the same group. Outgoing calls, by default I want to send those over VoIP, so our primary ARS puts outgoing calls over line group 1. Emergency calls, 0800 numbers etc, go over line group 0, ISDN.
Either add URIs in the same method for your other 2talk phone numbers, or if you are running IP Office 5.0 or higher you can setup a wildcard URI as pictured above to accept calls for any number.
5. Under the VoIP tab I have the call initiation timeout set to 2.
IP Office ARS Configuration
We have IP Office configured so we don’t have to dial any number to get an outside line – just dial the phone number straight away. This is possible because we use extension numbers starting with 7, and there are no local numbers in Christchurch beginning in 7. We also have the ISDN lines setup as a failover if VoIP is down.
To simply things I would suggest one of the following; either get the users to dial a different prefix to make a call if it isn’t working normally, or you can set up an automatic failover to PSTN using two ARS routes as shown below.
User initiated manual failover
If you dial 9 to get out, set that shortcode to go over VoIP. Then, setup another short code ‘8N’ (or similar), which forces calls over a normal phone line, so if VoIP is down people can just dial 8 to get out instead of 9. This won’t be an option for some people though, so I’ll share our configuration below.
For PSTN, set it up like this, with line group 0 being the outgoing line group of your PSTN or backup phone lines
Setup the Out of Service Route on the Main ARS to go to the PSTN ARS plan.
Setup the main outgoing call shortcode, i.e. ? to Dial to the VoIP ARS line group (i.e. 50)
Now, if VoIP stops working for whatever reason the calls will go out over PSTN. Unfortunately they will take longer to go through and there will be some horrible beeps in the process which I haven’t worked out how to disable yet!
Asterisk has been around for a number of years now, most tech consultants will be aware that there are open source PBX solutions that can run on your PC. However, until I discovered the Atcom IP series of Asterisk appliances, I knew I would have a hard time convincing small businesses of the benefits of VoIP.
There is a strange sense of security that goes with having a dedicated appliance for something as critical as telephony. Businesses that are used to having a phone system on the wall somewhere may not be totally comfortable having a phone system running on any old PC sitting in the server room (although this could be a safer option, as you always can drop the hard drive into another machine if something goes wrong with one).
While there have been Asterisk appliances from major names like Digium, their cost is not much cheaper than a POTS system, making migrating from a traditional phone system hard to justify. The Atcom IPxx series on the other hand which you can get the base unit for under $500, depending on how many analogue trunks and extensions you want.
I should note that it is not a trivial cost associated with moving to VoIP – if you have existing analogue phones the cheapest way will be to use ATA devices such as the Linksys SPA8000. VoIP phones start at around the $200 which soon adds up.
The ATCOM looks fairly easy to set up. There will be a learning curve, but it is definitely within the scope of in house IT staff to create new extensions, reconfigure IVR menus, if not set up the whole system from scratch.
For the price, I would buy two, and keep one as a spare that I could drop the flash card into should anything happen to the other.
The Atcom IP01 / IP04 / IP08 can handle around 30 concurrent calls, which is plenty for any small to medium business in New Zealand.
My immediate reaction when I saw this was to buy it just because it was so unbelievably cheap. Unfortunately, being a student means that isn’t quite a good enough reason to buy one.
Update: July 2009
I still haven’t tried out any of the ATCOM gear. I have to say I am nervous about the quality, I’d love to think it would be excellent, but until I have tested it I won’t know. In New Zealand of course you are probably still going to want analogue trunks instead of relying on 2talk/SIP trunks unless you have a really nice internet connection.