UK Vintage Radio Repair and Restoration Discussion Forum - View Single Post

julie_m · 7th Jun 2019, 9:09 pm

Asterisk is awesome! If you ever messed with phones when you were younger, Asterisk is the mythical "sky blue pink box with yellow spots on", and then some more. It got there first and it was already Open Source, which avoided the possibility of multiple deliberately-incompatible competing proprietary standards. You cannot wind up locked into a single vendor's ecosystem, nor locked out of features it's entirely capable of supporting if you just pay for a licence.

You can get analogue line cards that plug into a PC running Asterisk, which then accept a number of modules; either FXO to connect to an exchange line, or FXS to connect to an analogue telephone. There are also unofficial clones available if you look, which are lower-cost and fine for experimental purposes but I wouldn't advise anybody to use one of them for running business-critical services. At least have a spare handy if you are going to chance it.

If you have at least one FXO and one FXS port, you can sit your Asterisk box between the exchange line and the PABX to act as a fancy answering machine out of hours, or (if you are into PHP or CGI scripting) as configured via a web interface. A second FXO port would allow you to connect your Asterisk box to the PABX as an extension, allowing messages to be retrieved by dialling into it.

The next logical upgrade would be to replace the analogue phones and PABX with VoIP phones added to Asterisk as extensions.

The final phase would be to have your number(s) transferred onto a SIP trunk instead of the PSTN. Your SIP trunk provider will be able to arrange this, and also provide stanzas to paste into your various configuration files (such as /etc/iptables/rules to open the necessary ports just to their IP addresses and not any random hacker, /etc/asterisk/extensions.conf to set up a context for incoming calls and rules for routing outgoing calls and /etc/asterisk/pjsip.conf to create a SIP endpoint for the trunk). You will be able to route outgoing calls over the SIP trunk even before the incoming number is transferred, so it will be possible to run the two systems in parallel and switch over as seamlessly as possible.

7th Jun 2019, 9:09 pm	#3
julie_m Dekatron Join Date: May 2008 Location: Derby, UK. Posts: 7,735	Re: Using Asterisk alongside an old BT PBX Asterisk is awesome! If you ever messed with phones when you were younger, Asterisk is the mythical "sky blue pink box with yellow spots on", and then some more. It got there first and it was already Open Source, which avoided the possibility of multiple deliberately-incompatible competing proprietary standards. You cannot wind up locked into a single vendor's ecosystem, nor locked out of features it's entirely capable of supporting if you just pay for a licence. You can get analogue line cards that plug into a PC running Asterisk, which then accept a number of modules; either FXO to connect to an exchange line, or FXS to connect to an analogue telephone. There are also unofficial clones available if you look, which are lower-cost and fine for experimental purposes but I wouldn't advise anybody to use one of them for running business-critical services. At least have a spare handy if you are going to chance it. If you have at least one FXO and one FXS port, you can sit your Asterisk box between the exchange line and the PABX to act as a fancy answering machine out of hours, or (if you are into PHP or CGI scripting) as configured via a web interface. A second FXO port would allow you to connect your Asterisk box to the PABX as an extension, allowing messages to be retrieved by dialling into it. The next logical upgrade would be to replace the analogue phones and PABX with VoIP phones added to Asterisk as extensions. The final phase would be to have your number(s) transferred onto a SIP trunk instead of the PSTN. Your SIP trunk provider will be able to arrange this, and also provide stanzas to paste into your various configuration files (such as /etc/iptables/rules to open the necessary ports just to their IP addresses and not any random hacker, /etc/asterisk/extensions.conf to set up a context for incoming calls and rules for routing outgoing calls and /etc/asterisk/pjsip.conf to create a SIP endpoint for the trunk). You will be able to route outgoing calls over the SIP trunk even before the incoming number is transferred, so it will be possible to run the two systems in parallel and switch over as seamlessly as possible. __________________ If I have seen further than others, it is because I was standing on a pile of failed experiments.