Questions On A Possible Asterisk Setup

I’ve been asked by a client to investigate the possiblity of setting up an asterisk box for his small company (5+ users) to use.
What he wants to do is basically make internal calls (which seems fairly easy to do) but also use Asterisk as a sort of VOIP bridge to make calls online, in effect making cheaper calls. He has used Skype but he wanted more advanced features, such as phone id blocking (which I’m not sure is even possible).
Is this setup even possible? I plan on running Asterisk on Debian. I’ve never touched Asterisk before so this is unknown ground for me, but I just wanted to see if what he wants is even possible before I start.
Thanks in advance to any replies

Yes, it is possible. This is exactly what asterisk does.

Ok, good. But how would I connect it to VOIP? Will I need to setup Asterisk with a VOIP provider such as Skype? As it’s external sources the user wants to call via VOIP to save on costs.
I’m really new so please excuse my newbie questions.

Most VOIP providers use SIP to connect (not SKYPE, that’s their own protocol). Asterisk supports SIP as standard and is pretty good at routing calls over SIP VOIP providers.

There are a few providers who support IAX and in some ways this is a better choice but you may not be able to find one with the call rates you want.

You (generally) can get * to make outgoing calls via SIP without firewall / NAT issues, so long as the voip ports are not blocked on your router.

Basically you set up an account with the provider and, usually on their website, then set up your sip.conf (or equiv if using a GUI) with the server, username, and password and a few other bits. Then you set up entries in the dialplan to route specific calls over voip (e.g. local,national,etc).

There are all sorts of VOIP providers out there. Not just SKYPE . :wink: Some charge setup fees some do not. Some chare monthly, some are pre-pay, some give free calls, some give free minutes. You need to search around.

It’s pretty awkward to make * taly to SKYPE, by the way, and not free.

You may want to splash out on some G.729 licences from digium ($10/chan) when you are up and running as this will conserve bandwidth over your link if your VOIP provider supports it.

Thanks very much for your reply you have really cleared a lot of things up for me. My client wants to use a pre-pay VOIP service, can you recommend any in the UK (long shot I know).
Thanks again for the reply.

I use (not however beware, the company, betamax, has a lot of sites with different rates but they all use the same core net/servers. Good rates comparison here also at

Check about any restrictions before committin any credit. That said, its been working great for me. Beware of justvoip, it has great rates but is softphone only.

Other possibilities are who do iax too. can give you free inbound (SIP & IAX), as can with less restrictions.

Oh, final word, this can be useful:-

; - resiliant, SRV, fallback to 1899 ; ;exten => _823.,1, Dial(SIP/${EXTEN:3}@voipcheap,,TL(3600000)) exten => _823.,1, Set(IDDNUM=${IF($["${EXTEN:3:2}" = "00"]?${EXTEN:3}:0044${EXTEN:4})}) exten => _823.,n, Set(i=1) exten => _823.,n, While($[${i} <= 6]) exten => _823.,n, Dial(SIP/${IDDNUM}@voipcheap-${i},,TL(3600000)) exten => _823.,n, noop(${DIALSTATUS}) exten => _823.,n, Set(i=${IF($["${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?$[${i} + 1]:1000)}) exten => _823.,n, EndWhile exten => _823.,n, GotoIf($["${DIALSTATUS}" = "CHANUNAVAIL" | "${DIALSTATUS}" = "CONGESTION"]?822${EXTEN:3},1)
What it does is to make SIP to a multi server cluster more resilient (most providers have 2 ore more servers). 823 is my trunk code for th provider, I can supply more info if you need it.

Thank so much for the information. I will check out this VOIP providers.
No doubt if I do have any problems I will contact you.
Thanks again


Please dont mind me asking this, i am really stuck. Can i just get some tips on configuring a * server.

Scenario is , i have a VOIP provider who has some minutes of traffic to be sent over to my * box, on my side i have a Digium TE210 card, and a E1 (isdn) from my local Telco. So i should take traffic from my VOIP via sip and call the local telco fones.

Which config files would i need to play around before i get this working and also the local telco should not be able to call back , only one way.

VOIP --> Internet —> * —> local telco

Please !!![/img]