Getting started questions

We are in the process of seeing if Asterisk will work for us. We have already reviewed most of the VOIP PBX providers - none of which have stellar customer service records so the thinking is we could do a better job internally.

So I just need to getting started information. We will be using a NAS that supports Asterisk and will need to connect in at least one SIP line.

My first question is if a call comes in on the single sip line can you also have a POTS line connected too? How do calls roll over to an open line?

Again, this is just the start of playing with the system to see if it will fit out needs.

Please note that Asterisk GUI is unsupported, (obsolete) even though it comes with the only NAS we’ve heard of that includes Asterisk. All answers will assume you are coding Asterisk configuration files directly.

You need to explain more about your environment as “single SIP line” is a constraint of that environment. Asterisk supports multiple lines which can be a mixture of many different types, e.g. SIP, H.323, analogue, ISDN… If you have the right hardware, you can connect to an analogue POTS line. (You can also do so indirectly using SIP for the first hop.)

For circuit switched lines, you assign them to one or more groups and invoke them with Dial(DAHDI/g), where represents the group number. The DAHDI driver will automatically choose the free line. VoIP destination intrinsically allow arbitrarily large numbers of simultaneous calls, although there may be commercial limitations, and you may run out of bandwidth.

If you want to overflow between different technologies, or between different VoIP peers, you need to use consecutive Dial applications for each one in turn. Ideally, you should check ${HANGUPCAUSE} and ${DIALSTATUS} to determine whether the call is likely to fail on the alternative route. For VoIP, if you capacity is bandwidth limited, you may need to use group counts to detect when to overflow.