MarkW S7

I've made progress on this. I'll detail how I registered the audio codes to the pbx (I'm in a hosted multi tenant environment) and how I was able to send calls out. Go to Voice Over IP\Signaling Protocol\Signaling Protocol 1.All settings default here except "Gateway Name - User Domain". Put in your domain/realm that you are pointing to, example: customerdomain.com This is huge so that outbound calls have appropriate header info Go to Voice Over IP\Signaling Protocol\SIP Proxy and Registrar 1. Check "Use SIP Proxy" 2. Enter your domain/realm that you are pointing too, example: customerdomain.com 3. Leave proxy port and max # of authentication retries default (unless you want to fiddle) 4. Check "Use SIP Proxy IP and Port for Registration" 5. Leave Register Expires and SIP security default 6. Check "Use SIP Proxy IP" 7. Enter your PBXnSIP's IP, example: 68.142.234.143 8. Leave the port 5060 default Go to Voice Over IP\Line Settings 1. Select "Edit" for either Line Number 1 or Line Number 2 2. For User ID, input your Account Extension that your are registering to, example: Account: 100 (extension) would be the user id 3. Don't block CID unless you want to 4. Display Name could be whatever you want (more for naming purposes) does not effect your header info 5. SIP Proxy: a. Authentication User Name: the Account that you're registering to, example: 100 b. Authentication Password: The SIP Password you specified when setting up the Account originally on the PBX 6. Leave everything else default These steps should get you registered and passing audio to the FXS ports. I don't claim to know this 100%, but this is what I've found so far and that has worked. Oh and I have t.38 signaling working, and that configuration takes place on the Voice Over IP\Voice and Fax section.

Hello, I was wondering if anyone has any experience with using the audio codes media pack line of ATA/routers. We have used them with previous systems and also with on site Faxback ATA CPE's. What I'm having trouble with now though, is that we'd like to use them with the PBX but after I reg the device to the pbx I can't make outbound calls (I receive a busy). I know this is a dial plan issue especially since I can call in just fine. Anyone out there use or looking to use these devices? On another note, I've been working with the Linksys Pap2t, SPA2102, SPA8000 as ATA's and had success with all. The SPA2102 and SPA8000 support T.38 but I am not 100% on it's full operation. The Audiocodes MP202, MP202B, MP204 all support T.38 and actually have some unique settings compared to the Linksys products. Also the Audiocodes provide traffic shaping, better firewall, and on some newer series even wireless. (I'm operating as an ITSP) Anything would be helpful, even if you're in the same boat as me, let's get these ATA's under wraps.

Hello, I've been trying to get some music on hold running for our system. (I'm in a hosted multi tenant set up) I take a MP3, use my favorite music editor (WavePad), adjust the amplification of the music to match that of the PBX's default MoH, and save it as 16kBits/s, 8,000Hz, Mono, MPEG Layer-3. The MPEG Layer-3 format is the only format that has the 16/8000/mono format that the PBX wants. But When I place the file in the moh source file and create it in the PBX, all I get is loud static. Any guidance on a music editing program or method for implementing moh? -Mark