| Jason, There are a couple things we can try to fix your problem. Your firmware shouldn't be an issue, but latest I've got now is: MP118_SIP_F4.80A.034.004.cmp Let's try some quick things first though: In your web interface, go to advanced config - channel settings / voice settings There are some options here you can play with: "Voice Volume" (IP side of this thing) - by default this should be set at '1'. Try bringing this down slowly, I'd say in increments of 5 (-4, then -9, and so on). Range on this option can be anywhere from -32 to +32, you really shouldn't need to go beyond -15; but you're actual volume on the calls should still stay reasonable. "Input Gain" (telco side) is another option you can slowly change as well (set to 0 by default). There should also be spot where you can specify the "codername", you could possibly try changing this to another codec such as G.729 or G.711u-law (should be the same codec being used on your Asterisk system) try changing packet size from 20 to 40 or 60. This may also help. If none of this stuff helps, let me know. We can then start getting really technical. Jessee Holmes Atacomm / Ataractic Corporation www.atacomm.com V: 1-877-700-VOIP Looking for voice over IP products? Visit our VoIP store at http://voipstore.atacomm.com/ On Nov 1, 2006, at 1:45 AM, Jason Kim wrote:
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