Archive for the ‘phones’ Category

Nokia N95 8GB, Sony Ericsson w580i, Bluetooth & Sharing Ring Tones

Wednesday, July 16th, 2008

We happen to have a few cell phones laying around the house. Of course I have my new nokia n95 8gb, we also have the Sony Ericsson w580i, and we have a blackberry pearl that Summer uses. I have been messing around with the n95 and the w580i. I wanted to be able to transfer some ring tones I have been making from the n95 to the w580i. I have never messed with bluetooth before so this was all new to me. I have basically been living in the 1980’s as far as cell phone technology goes for the last 5 or so years.

Getting it going

So first on the Nokia

  • menu
  • tools
  • bluetooth

From the Bluetooth menu:

  • Set bluetooth on
  • My phone’s visibility set to show to all
  • My phones name I set to Mike (since thats my name you can pick any name you wish)
  • Remote SIM mode I have set to off

Next on the w580i open the menu and select settings, then scroll left until you get to Connectivity:

  • Select Bluetooth
  • Turn On
  • Visibility
  • Show Phone
  • My Devices
    • Select Add new device
    • Make sure your n95 is close enough and bluetooth is on
    • Click on “Mike”
    • Passcode: 00000

Now a dialog will open on your n95 asking to allow the connection and to enter the passcode. Enter 00000 and hit OK. Now the 2 devices are connected. In my case I wanted to transfer a ring tone from the n95 to the w580i. So now we need to grab the n95 and open the menu and go to tools, and then file manager. Find the file you want to send. Scroll to the file and click options then send, send via bluetooth, and then select the w580i (which by default is named W580i). Thats it.. super simple. Now you can use this file as a ring tone or text alert or what ever.

Asterisk Dual Servers with SIP

Tuesday, July 15th, 2008

I have found about a million articles on the net for connecting 2 asterisk boxes using IAX2. What I have not found is many describing how to do this with SIP, so thats what Im going to talk about. Now since I am a FreePBX user I am going to talk about how to do this using the FreePBX web GUI. You can use this method if you use TrixBox, or Elastix, or PBX In A Flash, or if you just use FreePBX with your own distro.

I have 2 systems. To make this simple Im going to call them pbx1 (which for this example is in San Antonio) and pbx2 (which we will pretend is in our Houston office). First we need to log into the web interface on pbx1 and then get to the section where you configure trunks. Click Add SIP Trunk. Im not going to mess with any of the General Settings, or the Dial Rules. I am going to go straight to the Outgoing Rules. In trunk name Im going to call this trunk-hou-peer. Next I jump into the PEER Details box. I put the following into the box:


Next, Remove all the settings from the Incoming Section. Hit submit, and then apply the changes.

Next we need to log on the web gui for pbx2. Now we are going to go from Houston back to San Antonio with this trunk. We need to add a new SIP trunk here. Follow the same steps as before, but now name this trunk: trunk-sa-peer You may notice that the name of this trunk is the username we used in the peer details on pbx1. Now in the Peer details on pbx2 add the following:


Now you will notice that the username on this peer is what we named our trunk on pbx1. The context used in both is the same. This will give you access from pbx2 to dial an extension that is on pbx1 and vise versa. Now all we need to do is submit this and apply the changes.

Next all we need to do is create an outbound route to use our new trunk. For the sake of keeping it super simple lets assume you have extension 100-199 on pbx1 and 200-299 on pbx2. We need to go to the outbound route section and create a route on pbx1, name it toHouston and it needs a dial pattern that looks like this:


Click on the Intra Company Route. Then select our Houston trunk. Next submit and apply changes. Now head over to the pbx2 web gui and do the same steps only name this route toSanAntonio and for its dial pattern use:


Click Intra Company Route select the San Antonio Trunk, hit submit and apply. Now from extension 100 in San Antonio dial 200 and bamo thats it. You can take this a step further and even do toll-by-pass now.

Why Does Fonality Choose to Deceive You?

Wednesday, June 11th, 2008

Find the answers to that here

Centralized Voicemail server for multiple Asterisk systems using FreePBX and IAX2

Sunday, May 13th, 2007

Well I never really finished up my asterisk tips posts… I’m lazy and most of the free time I do have I dont like to spend on the computer anymore (what can I say). Im going to make it up to you though by sharing with you how to make a very slick asterisk setup.

So why would anyone want to have a stand alone voicemail server anyway?? Well thats pretty simple. Lets pretend you have a small to medium sized company. Maybe 100-500 employees, maybe more maybe less… You have 3 or 4 office locations, they span several cities or maybe even other states.. It doesnt really matter where they are. The important thing is that they all need to be connected. You need to be able to call from one office to another, and at times you will even need to forward voicemails from one location to another. Asterisk does not just automaticly support this. Even if you have routes in place that make it so you can call each person on your network with a SIP call if their voicemail box isnt on the same server as yours you cant forward them a message. This can become a big problem real fast. Infact at our company we were considering not using asterisk because of this. The good news is that its really simple to do.

First off Im not going to cover how to connect the multiple servers. This has been articled to friggin death. Its covered on many sites, one of them not being this one. Next Im going to assume you have figured out how to make connections from 1 asterisk system to another using IAX2. If not please check out this guide. It works great.

Now on to the fun stuff. I used FreePBX on our 3 PBX servers to get our dial plan setup and have a nice easy to manage user interface. I kept thinking that this would keep me from being able to customize my dial plan like i needed to but I was all wrong about that. Im now going to point you over to the guide I followed: Click me!!!

This guide is fairly good. It doesnt cover doing it with FreePBX though. If you use FreePBX you may have found that if you edit one of its config files you lost your changes once you entered the gui, or made an update to FreePBX. I went to the IRC and asked how to get around this. It was simple, at the top of the config files you see a line that says #include somefile_custom.conf All you have to do is edit that custom.conf file and over ride the context you need to edit. So if you have in your sip.conf file a [foo] context you go to sip_custom.conf and make [foo] in there and then the settings for the [foo] context from your sip_custom.conf file will be whats used instead of the [foo] from sip.conf

In the article I asked you to click to see how to do the setup it said you needed to edit a macro in extensions.conf. The context you need to edit is called [macro-vm] First what I did was copy and pasted the [macro-vm] context from extensions.conf and pasted it into extensions_custom.conf then I edited the following:

; over riding the below for our central voicemail server
exten => s-BUSY,n,Dial(IAX2/toVMail/b${ARG1})
exten => s-BUSY,n,Goto(exit-${VMSTATUS},1)

exten => s-DIRECTDIAL,1,NoOp(DIRECTDIAL voicemail)
exten => s-DIRECTDIAL,n,Macro(get-vmcontext,${ARG1})
; over riding the below for our central voicemail server
exten => s-DIRECTDIAL,n,Dial(IAX2/toVMail/${ARG1})
exten => s-DIRECTDIAL,n,Goto(exit-${VMSTATUS},1)

From there I just followed the rest of the guide the other nice fellow provided and saved my settings, then setup a plain jane asterisk server for my voicemail server. I didnt use FreePBX on it at all. Managing that server is braindead simple so I didnt see the need. I hope this little tidbit will help others out there who have to set this up.