This is definitely a lesson in "You get what you pay for". That being said, any help on configuring this beast to work with asterisk and c*net would be greatly appreciated. This is an older (2006-2007) model with four FXS ports. I am pretty sure the firmware is just as old, and it won't take an update. I got it for dirt cheap and, while functional, its not working right. Currently, I have worked through several issues with no manuals or documentation, but one issue remains. Only the first FXS port dials out correctly. All the ports work for incoming calls, but only the first port will allow you to dial out. The other ports just return a fast busy. They are all registered with asterisk and the necessary changes to sip.conf and extensions.conf have been made. I'm sure the issue is somewhere in the ATA but I cannot find any differences between the settings from one port to the next. Any help would be greatly appreciated and thanks in advance.

Have a look at this:

https://web.net2phone.com/partnerships/product/broadband/pdf/n2paudiocodesconfigguide.pdf

Another thing to check is to make sure the speech codec Asterisk is trying to use is one that is compatible with your ATA.

Scott, I've read that manual you linked close to a dozen times. I had all the codecs set to G711uLaw. All my other ATA's are set to the same codec. I gave up and bought two 2-port linksys pap2t ATA's and I had them up and running in less than 5 minutes. Like I said, I could get the first FXS port to both receive calls and dial out just fine. The remaining 3 ports just won't dial out, not for love or money. I'm wondering if maybe the firmware is corrupted or the thing is just on the way out. I just could not find any setting that was port specific (as in a non-global) that had anything to do with outbound routing. The ports would register just fine with asterisk, and like I said, receive calls normally. I like to think I'm pretty savvy with this stuff, but that box made me feel like a first grader trying to operate the space shuttle.

I really like the Linksys ones. Good choice. My guess would be that the thing was just on it's way out. Sorry for the delay, I've been flying to Maine to be with family the pass day or so.

I agree about the linksys ATA's. They were a breeze to set up. I may buy some more, since I got them so cheap. Been thinking about voip providers for a second line, and the linksys seems like a good candidate. Only real problem, if you can call it that, is no one seems to have New Hampshire numbers available. I tried call-centric, which I know you've recommended a few times to people, but no joy. That's a topic for a new thread, though. Maine, nice! I live about 45 minutes from the Maine border, not that it means anything. Could drive another 6 hours and still be in Maine. Furthest I've ever been is Augusta. No real interest in going back, either. The supply house I worked for bought out the electrical portion of an FW Webb up there. Two days loading and transporting all that stuff back to Milford, NH. Anywho, If I don't talk to you again, have a good Christmas with the family!

Do you have different registrations for each port? If so, then you'll probably have to make asterisk skip over busy ports in selecting an outgoing trunk. In FreePBX it's the "Continue if Busy" option.

Yes, I have different registrations for each port. I'm actually running Astlinux on a thin client. How would I go about making it skip over busy ports? Also, I don't have any outgoing trunks registered. I use it to connect to the collectors network (C*net).
If you need any more details please ask. I admittedly know very little about Astlinux. I have been able to add extensions, either in the form of ATA's or a 6 line Cisco I recently added, without much trouble. Primarily by adding the info to sip.conf and extensions.conf. I haven't really needed to do much else to get things working.

Hmm, you'd probably need to write some custom dialplan to make it work. Maybe Google "Asterisk dialplan to skip busy trunks". If there's any way to convert to FreePBX I'd highly recommend it - guys are running it on Raspberry Pi's so it can be done with very little power/storage.

UPDATE: I finally got it to work, though I have no idea what I really did differently. I went through and started from scratch and VOILA!

Sometimes you just need to go back to square 1 and start over.

Yup.

That being said, this is one of the most frustrating SIP devices I've had to configure. Not that I've done many, but the Sipura stuff is a walk in the park next to the Audiocodes. There are just SOOOOOOOOOO many settings and configuration options that it starts to make your head hurt.

I know this is an old thread, but I wanted to post an update for future seekers of information. Not only that, but the next time I foul it up I'll be able to look here and remember what I did. I recently had to make some changes to my system and I was again faced with the problem of only the first FXS port being able to dial out. The major breakthrough came when I bought the same model (Audiocodes MP-104a) in a FXO flavor. It had the same issue where only the first port would work correctly. I figured at this point it can't be the device. This was a brand new in the box unit, and it seemed unlikely to have anything corrupted. At this point I decided to look at the sip.conf file in asterisk. Something had to be different. It wasn't obvious at first, but when putting the info into sip.conf, the audiocodes devices must have type=friend, and not type=peer, like all the other sip devices I have. The other change I made was srvlookup=no. Those two minor changes and all the ports on both devices instantly worked right. I spent two days futzing with the settings in the devices before it occurred to me that it must be something external. I hope this tidbit saves someone else the hair tearing I went through. Once these things are configured they are solid as a rock, that's why I like them. Just a long process to get them there.