3rd Party PSTN provider

I would like to install Asterisk on a dedicated server from a hosted internet provider. I, however, would like to handle only VOIP calls within my organization and let a 3rd party handle my outgoing PSTN calls. Is there a possibility to selectively transfer calls to a third party for handling? i.e. if one of my employees dials a fixed or mobile line the handling gets “transfered” to another party by my Asterix PBX. I need this because I can not possibly handle the installation of one of the telephony cards myself and figure out it would be easier to “outsource” that and get it charged on my companies credit card. Does Diguim offer such a service? If not, is there any 3rd party you could recommend that can handle my request? Can I find examples somewhere on how to configure my PBX for such a scenario?

Thanks in advance ,
Roland. :smile:

No problem, I use easyspeedy.com and a myriad of 3rd party voip providers:


(to name a few)

all based on a softswitch only implementation.

I’ve had reliability issues with VoIPJet’s service. VoicePulse is much better, but also more costly. I don’t see either VoipBuster or IPKall listing any Asterisk support on their site.

Can anyone recommend other PSTN termination providers that support IAX??

Thanks very much… I am already so excited to know my needs can be handled.
I had a look at the PSTN termination providers and noticed thier price schemes vary largely. Is there an in-built possibility to implement Least-Cost-Routing on my asterisk server? i.e. if(destinationNumber == x) use provider A, if(destinationNumber == y) use provider B, and so on ???
My company is spread around Africa and difference in prices vary by up to $0,50/Min. for some countries which makes a big difference. No one seems to offer the best prices across the board.

Thanks again very much for your tips,


Thank you sooo much MuppetMaster,

I hope I can get everything started this week-end.

Have a nice one,

I have, but only with GSM mobile phones, which many VoIP providers suffer with due to multiple transcodings (IMHO).

Works fine.


Works fine, as you may forward to any SIP URI you want to, not just FWD.

One last question before the week-end…

…is it possible to associate prepaid accounts for my employees so I don’t loose control of the costs? e.g. Each employee gets say $100/Month for PSTN calls and automatically can no longer call PSTN terminals after using up this amount except he/she gets some more credit allocated.
Are call records available from the Asterix PBX. i.e. Originating-Duration-Destination records?

Thanks in advance,

Have a look here for a myriad of approaches and applications to your question:


Learn to love the wiki:wink:

for more voip providers, a shameless plug for my new site:


About your question for the billing, google for astcc or astbill.