[agauger]

I am looking for a trustworthy person to help me get my Asterisk@Home server running correctly.

I have successfully installed the Asterisk@home v1.5 software on a dedicated server (AMD Sempron 2800, 1 GB Ram, 30 GB HDD, 100 Base-T NIC, in a 2 RU chassis). I’ve purchased 2 Sipura SPA-841 IP telephones and a Sipura SPA3000 FXS/FXO interface from www.VOIPSupply.com (great company).

This telephone system is going to be used for my residence and home office. We will eventually have between 6 and 8 extensions (all IP phones except for one analog cordless telephone-hence the need for the FXS port on the SPA3000) using only 1 PSTN line to make all inbound as well as all outbound calls through. We did set up BroadVoice as a VOIP provider to check that everything was running correctly. It worked great! No problems using this service at all. I have since canceled this service as we have no plans to use a VOIP provider at this time. We are interested in taking advantage of all the features an IP telephone system has to offer while only using a single, standard analog telephone line to connect to the rest of the world.

I have successfully configured both SPA-841 IP phones and the FXS port on the SPA3000 as extensions 200, 201 and 204. I can dial between them successfully. Both the FXO and FXS ports on the SPA-3000 as wells as both SPA-841 telephones are successfully registered with the Asterisk server.

We finally got our Asterisk configuration to allow incoming calls to ring a single extension with everybody’s greatly appreciated help. Calls sound great despite hearing a bit of a delayed echo of my own voice in the telephone when I speak into it. Unfortunately, at this point we are only able to answer incoming calls. When we attempt to dial out, we receive the all too familiar prompt, “All circuits are busy now. Please try your call again later."

Can someone help me troubleshoot this problem? I have remote HTTP and SSH connectivity to my Asterisk@Home server available to whomever agrees to assist me. I’d also be willing to offer cash as an incentive as we have been struggling with this for nearly 2 months.

Thanks for your consideration. If interested please reply to this post or send me an e-mail:
GaugerProductions[at]Yahoo.com

Thanks,
~Aaron

[AlexanderBell]

Have a look at http://voip-info.org/tiki-index.php?...erisk+at++Home and http://NerdVittles.com. Those won't cost you a dime and cover pretty much everything you want to do.

[thekiwi]

Originally Posted by AlexanderBell

Have a look at http://voip-info.org/tiki-index.php?...erisk+at++Home and http://NerdVittles.com. Those won't cost you a dime and cover pretty much everything you want to do.

Yes ... use the Nerd Vitles pages to get your system setup. I have exactly the same system as you, ie no VOIP just all calls going out via PSTN and it works a treat

http://mundy.org/blog/index.php?p=61

http://mundy.org/blog/index.php?p=62

http://mundy.org/blog/index.php?p=63

http://mundy.org/blog/index.php?p=65

http://mundy.org/blog/index.php?p=66

[thekiwi]

Originally Posted by agauger

We finally got our Asterisk configuration to allow incoming calls to ring a single extension with everybody’s greatly appreciated help. Calls sound great despite hearing a bit of a delayed echo of my own voice in the telephone when I speak into it. Unfortunately, at this point we are only able to answer incoming calls. When we attempt to dial out, we receive the all too familiar prompt, “All circuits are busy now. Please try your call again later."

Have you configured a trunk for your PSTN?