[bbbeavis]

OK, I've followed every variation to config Asterisk for SIP providers; I fail every time (I do remember once making outgoing calls on it). I even resort to my Microsoft training and do the same thing over and over and expect a different result: no joy. IAX2 works peachy with multiple providers, though. I'm behind a Linksys WRT54G with Sveasoft Alchemy-V1.0 v3.37.6.8sv and on Cox residential cable. I'm in the DMZ. So, is it the router, the service, or is the problem between the chair and the keyboard? Is there something that all, and I mean all, the tutorials miss because, at this stage, it is not caused by typos (see Microsoft training)?
:?

[dualarrow]

Which providers are you trying to use ?

I'm using FWD (SIP but I have had it workin with IAX), freecall.net.au, bbpglobal.com, sipphone, simtex.net.au, mutualphone and a few others.

[mberlant]

My Asterisk runs fine behind a WRT54GS with Alchemy. I have the Asterisk in the DMZ mostly because I'm too lazy to make the port forwarding entries in the router.

How are you configuring your Asterisk? Are you hand configuring or using AMP? I hand configure and have built my configuration one service at a time. My guess is that you have some silly error in sip.conf and that you can't find it simply because you have been looking at it for too long.

When you issue "sip show registry" do all/any of your SIP services show as Registered? If not, what is the status? If the services are Registered, are they Reachable (sip show peers)? If your services are Registered and Reachable we will need to examine extensions.conf

Let us know, and please post sanitized snippets of your sip.conf if you are not Registering properly.

[bbbeavis]

I'm using AAH 1.5. I just rolled back from AAH 2.0 beta2 as it was too buggy. I will have another go at SIP with a Callpacket free account I just got (thought I was getting a local number from them; instead was randomly assigned another city and their paid for programs are overpriced :x ) If I fail (again), I'll post my sip.conf file. I'm coing to try AMP then hand configuring.

[mberlant]

Beta 2 was so buggy that Beta 3 was released just one day later. Beta 4 was just released, so you may want to try that out some time.

Personally, there is only one feature in Asterisk@Home 2.0 that I am interested in (the SIPGetHeader command), so I don't see any reason for you to be burdening yourself with beta code at this point. Feel free to post your progress and let us help you debug your configuration.