First time running Asterisk and I’m looking forward to learning something new. However, the more I dig into the system and read through the docs…the more I realize there is A LOT to learn and I’m a little overwhelmed at the moment.
I’m setting up a new Asterisk system for local charity. Asterisk and FreePBX (2.8.X) are running on a small server, and I have a Digium card in it with 2xFXO and 2xFXS ports.
The charity has two incoming POTS lines (2 phone numbers) that will plug into the FXS ports. Internally there are two analog phones that I will connect to the FXO ports (a fax machine and a phone in an elevator). I’m planning to purchase Polycom Soundpoint IP phones for the rest of the building.
I can see the DAHDI drivers are running, and FreePBX sees them in the admin interface. Are they ready to go? Do I just plug in the incoming lines to the FXS ports and start setting up extensions and routes?
I can do some testing with the second incoming line without interrupting the office too badly…and planned to start internally with some software SIP phones before I went and purchased a bunch of Polycom equipment. So I have some room to experiment here…I’m just struggling a bit with where to begin.
Hopefully someone here can point me in the right direction.
I’m putting a simular system together, but I’ve made it a little further. I’ve got my DroidX, some soft phones, and some Cisco (SIP) phones all working inboud/outbound/internal. I’ve got a very basic dial plan configured, but I’m at the point where I need to really start building on that.
Give me your current status, and I’ll see if I can get you as far as I’ve gotten. Let me know if you run into software echo configuration issues, I just got mine figured out.
FXO (them/Office), FXS (me/Subscriber)… got it.
After posting the other evening I was able to get a few things figured out.
I ran the dahdi_genconf script…I think the hardware is running from looking through the dmesg log.
I haven’t plugged anything into the FXO/S ports yet, but have been able to setup a couple basic extensions with a softphone on my laptop and SIPDroid on my phone. I configured some DNAT and packet filter rules on the firewall and was able to successfully make calls between my laptop and SIPDroid (yea!) from outside the location…but had no sound (boo!). Something to work on there obviously.
I’m going to try plugging an external line into one of the FXO ports next and see if I can get that configured.
I used AsteriskNOW to setup the server. Getting that installed and running was no problem… I’ll just keep pressing ahead with this and will eventually get it all figured out.
I also recommend you plug the POTS lines into the FXO ports and the equipment into the FXS ports. If you connect it in the fashion you describe you risk damaging the card and it also won’t work.
No matter how you installed you need to run the dahdi_genconf script to configure your hardware. Often the files the script generate need a bit of hand tweaking to get the hardware up.
DAHDI driver config is in /etc/dahdi
/etc/asterisk contains chan_dahdi and dahdi_channels_* the Asterisk DAHDI channel driver configs.
Drivers in Linux and Asterisk can be a bit of a challenge to someone new to the technology. How did you install Asterisk/FreePBX?
i recommend you install PBX in a Flash since it comes preconfigured for a small office. Its very useful since you can see what how things were done and maybe you can start guessing from there.
PBX in a Flash does not come with any speficic hardware card drivers (FXS or FXO) so you probably would have to add those yourself.
you can also read this…carefully…its incredibly clear and useful.