Im still kinda new to FreePBX, but Ive been playing around with it for a few months now. I am currently putting together a system for the family business. We will be using 15 of the Cisco 7960 phones.
The system is configured to about 90% of my liking but im trying to work out a few things that Im not happy with yet.
What I am looking to do is list pick from line 1 through 4, and dial a number and it goes out on that line. I would also like to see of the other phones when a line is in use.
Currently, I have each phone configured for 4 lines. But I don’t think I have this correct yet though. When I select any of the 4 lines and dial a number, it always just calls out on the first available line. I can provide any files that are needed to take a look at, but im not sure where to start.
Here is the SIP.cnf file for one of the phones.
# Line 1 Configuration 000AF4642DAA
line1_shortname: "Line 1"
Line 2 Configuration
line2_shortname: "Line 2"
Line 3 Configuration
line3_shortname: "Line 3"
Line 4 Configuration
line4_shortname: "Line 4"
The phone label, displayed in the upper-righthand cornor if the phone
phone_label: "Shipping " ; Has no effect on SIP messaging
Phone password used for console or telent access, limited to 31 characters
If anybody can point me in the direction I need to go, I would really appreciate it.
On a side note, Ive been buiding this system on AsteriskNOW 1.8 with FreePBX 2.10. Everything has been updated to the latest versions.
Thanks for the help!
Thank you for your response.
I shall elaborate.
Currently we have a Panasonic phone system. It uses the Panasonic KX-7731 handsets. This is an analog phone system. There is keys at the bottom of the speed dial section that are used for lines 1 through 4. These light up Green when you are using that line, blink green when you have somebody on hold on that line, and red solid when somebody else is using that line and blinking red when somebody else has somebody on hold on that line. When any line is on hold, anybody can pick it up. ( i believe this would be like parking a call) This system has worked great for the past 5 years, but was never up to the amount of use it has gotten. The handsets are worn out and are in need of replacing. The Cisco phones are much more suited to a call center environment and the Asterisk/FreePBX system will fill the voids out current system has.
My goal is to recreate as much of the function of our current phone system in Asterisk/FreePBX as possible and go from there. If it could be possible to use the keys on the side of the 7960 to pick up “lines” the park them with another, and recall them with the line key that the call is connected to, that would be awesome.
Regarding the example I posted, 110 is an extension. Mostly everything I’ve seen on the internet shows using the same extension for each line. I never really understood this, which is why I am posting now. What is the best way to assign the “lines”? I’m unsure of how to assign the extension to the phone and have 4 different lines. What would the phone be provisioned as?
On a side note, there is 8 phones placed into a ring group. this is for all 4 FXO lines. when the lines ring, somebody will answer one line, and so on till all 4 lines are in use. then the call should be able to be parked easily with a soft key. the line should show that a caller is still on it but anybody should be able to pick it up. calls are usually passed around the office in this fashion.
I guess what I’m trying to accomplish is the ability to answer, park and retrieve calls with soft keys and see status of each “line” of the 4.
Sorry for the excessive rambling. I hope somebody can understand what I’m trying to achieve.
What you are looking for is not possible. The concept of lines does not really carry over to the VOIP world.
I think the first thing you need to do is rethink your concept of “lines” in sip. When you think of SIP there are no physical lines as in legacy POTS systems. Each “line” is a datastream, and with VOIP telephony, the more proper term is “concurrent” calls.
I don’t know of a simple way to light an LED on the phone depending on which datastream is in use or to select a particular datastream, particularly if they’re all part of the same SIP service.
From your example, you list “110” as both the name and the authname for each line. Unless you are doing that simply as an example, then that’s ok. If you have that in your system, you’ve essentially put the same “line” on each button.
All of that being said, it’s usually better to describe what you are trying to accomplish, why you need to light up a light when a particular line is in use and what that would accomplish.
Thank you Alan,
Do you know where I can find some documentation on how a VoIP system is designed to work in a small business/ call center environment?
Mate, there is an untold amount information available on the web. Just do a simple google search.
For starters try:
Asterisk™: The Definitive Guide