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TacoBandit

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  1. I am starting to experiment with 2 domains, everything is great however the localhost and new domain use the same audiocodes gateway. When calls meant for the new domain come in I get the following [5] 20071003144325: Identify trunk 5 [5] 20071003144325: Trunk AudioCodes Fairview sends call to 4629995 [5] 20071003144325: Trunk call: Could not identify user Can the system check all domains for extensions? Thanks, Kevin
  2. On my E61i, Grandstream, and Snom the switch will disconnect the call after about 2 1/2 to 3 minutes if my sets send no RTP packets, well if I am on a conference call muted that causes a problem. Any way to set the PBXnSIP to ignore RTP timeout from a muted set? Thanks, TB
  3. ITSP setup... Not company... Thats why...
  4. Any way to block caller id when calling station to station within the switch?
  5. Nice, That is too cool. Thanks for the link to the wiki. Kevin
  6. Just using the internal VM. There is no overlap and the log just shows DTMF digit received. Still no luck getting into the VM.
  7. I have a Nokia E61i which has a built in SIP client. I try to get it to connect and registration fails. Anyone have setup info for this? Thanks, Kevin
  8. I do that, it works from an extension just fine, but the auto attendant still says "this extension number does not exist"
  9. I read in other posts that when you call into the Auto Attendant dialing 8+extension should get you to the vmbox of that ext. We had an extension with 8 so I assume that knocked this down in the digit map, I removed that extension and still get "that extension does not exist whenever I dial 8 in the auto attendant. Is there somewhere I can reset the digit map to function correctly and we can check our VM by dialing straight into the auto attendant then 8 and extension? Thanks, Kevin
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