VoIP with asterisk
Well I have now with somewhat mixed results implemented what appears to be a working asterisk server with digium card and even correctly configured for Australia (loop start) and not the default which I guess works well in the US.
Have to give a big thanks to Ben who must have gone to a lot of effort to write Asterisk@Home Without Tears which was a very valuable reference guide and did assist in getting things finally working. I will just point out though if you do happen to be in Australia and are using this reference and installing a Digium card – you will need to change the default signalling from fxo_ks and fxs_ks to fxo_ls and fxs_ls in the zapata-auto.conf file.
I have found NEHOS (a VoIP providor) to provide competitive pricing and also good service, and allow the connection of IAX trunks directly to the asterisk server.
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Stephen,
Thanks for the plug
I do appreciate it…. and thanks also for finding a
correct zap setting. I am overseas at the moment and as soon as I get to my
X100P I will give your setting a go and include a correction in the guide.
I am currently using the original setting and it seems to be working
although there are many many many others having problems.. you probably have
come up with the remedy and I am sure others will be delighted.
If I can takea a few minutes of your time, I would like to know what are
the differences in performance with your discovered setting?
Regs – Ben
Hi Ben
I am not sure about the difference in performance, but there is a couple of advantages (I think) in the way I have set things up.
Firstly in order to meet Austel approvals you need to be using Loop Start signalling. If you are not there is potential (I think remote) to cause damage to telco equipment (when connecting pstn lines). In terms of benefits, I think although I have not tested, that it should increase the reliability of the cards, in making and receiving calls. It should not impact at all on the quality of the call.
The other change I have made is to use IAX trunks instead of SIP. This means that I do not have to open a whole heap of ports on my firewall – a bonus. But secondly the VoIP providor I am using supports IAX trunk protocol. This means that I can run multiple calls simultaneously over a single connection. So I have a single number with the carrier – over which I can configure multiple channels – the theoretical limit I understand is in the tens of thousands. In terms of bandwidth the first call would use much the same bandwidth as SIP, but for each subsequent call the overhead is reduced.
p.s for your info I am currently running asterisk@home 2.5 however the config above was the same as I used on version 2.1
Follow these guidelines and you will build that new home with little, or no, problems. temo sun room can help…
Interesting, what think company`s coe about it?
Interesting Article i also keep one voip blog