So here is the plan, firstly set up Asterisk as an internal PBX only for configuration and playing around with! (I have one PC with soft phone and 1 IP phone at the moment, but plan to get 3 more IP phones eventually and replace the PC with one of these. This will all be connected up via LAN.
I then intend to purchase a Digium card to enable me to connect to PSTN, and run all my PSTN calls through Asterisk PBX to the IP phones, using my current telephone number etc (I only have one line)…
I would then like to try using VOIP, but this is where I get lost…
Is it possible for Asterisk to have two external interfaces, one for PSTN, which it uses whenever I use an IP phone for dialling to a PSTN phone, and a network connection to my broadband router for whenever I dial a VOIP number?? I am confused because I see a number of services such as Vonage which seem to do this with a seperate box and using a PSTN gateway service??? Is this for people who want to get rid of PSTN entirely and use Broadband for all calls…to both PSTN and VOIP users?? I do not understand!
Also if all of the above is possible, (Which I really hope it is!) How do I get a VOIP number?!
Also as a general interest query what is the average network bandwidth for each call on the LAN?[/quote]
I currently have an internal Asterisk setup running version 188.8.131.52. I’m not an expert but know enought to be dangerous. Right now I have a POTS line connected to a Digium X100P card, two VoIP Providers (Qwest Communications and Gafachi), and two 8xx numbers.
Based on what you want to do, this is what I’d recommend. I’d wait to install asterisk until you get your FXO card. Once you get that, install all required modules for SPARC and the drivers for your FXO card.
Once that is done, you can setup rules in your extensions.conf file to handle how you want to route the call. For example, each IP phone (hardware or software) has it’s own extension. With a regular pots line, inbound calls ring all lines connected to your wall jack. Since each phone is it’s own number(extension), you’d create a rule to call them all at the same time and then will bridge your call to the caller depending which phone you answer.
Basic Example: (extensions.conf)
exten => 8000,1,Answer ;answers call
exten => 8000,2,wait(1) ; waits one second to let sip/rtp sync up
exten => 8000,3,Dial(SIP/5000&SIP/1000&SIP/5001,23) ; dials extension 5000, 1000, and 5001 at the same time for 23 seconds
exten => 8000,4,Macro(vmessage,1000@voicemail) ; goes to voicemail if no answer
exten => 8000,5,Hangup ; disconnects call
exten => s,1,wait(2) ; wait 2 seconds
exten => s,2,Goto,internal|8000|1 ; goto context ‘internal’, extension, priority
exten => s,3,Hangup ; disconnect call
The way I have this setup is everything internal is sip/voip. I use two Cisco 7912G phones and the XLite softphone. Extension to extension calling is SIP, and inbound calls from the land line is converted to SIP once answered by asterisk. Same with outbound, if I make an call from my IP phone, it goes VoIP internally until it leaves the FXO card which is converted to analog to go out a regluar land line.
When you’re ready to have an actual VoIP phone number, it’s really no difference functionally wise, just have more call compacity. I recomment keeping your POTS line and use the voip line as secondary line. Nice things about VoIP line is you can have as many simotanious inbound/outbound calls as your bandwith will support. Down side is if you internet connection goes down or lose power, you’re out phone service.
The options of how you want our outbound calls to route are quite robust. I currently have mine setup to route though Qwest VoIP first, if that fails, will go though Gafachi VoIP, if that fails will go out the FXO card via pots line.
If you not familar with VoIP technology vs TDM technology, all you really need to know for now is if you have a regular land line using and FXO card, internally you’re using VoIP technology and protocols. So all your routing and internal features are VoIP. When you use VoIP with a VoIP provider, you’re sending the same VoIP messages your asterisk box produces internally such as SIP Invites(from softphone to asterisk) and registration, except you’d be sending those messages to the remote carriers servers with your specific credentials you have with them. Like I say though, baic functionality internally will be identical whether you receive the call from a POTS line or a VoIP line.