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Hi all, I am having the following trouble with recording calls: When calls come into the support line did number, the call starts to record on the first queue, but appears to hang up when the call actually connects to the engineer (ie I see "got hangup request" on the cli and then mixmonitor ends.) I am guessing this has to do with the announce file that is played to the engineer

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow

I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not

Can anyone point me to info on how to change the functionality of the SIP (7.4) 7960's. We previously had an SCCP firmware on the phone and the users want the phone to work like it used too. Here are some examples: The users do not want to push the new call softkey or the speaker button in order to dial a call. They want to be able to just begin dialing the number. The users do not want

Hello all, I need to setup a new asterisk system with the following requirements: 1. Will be moving from chan_sccp to sip (7960's), but I want to support the sccp phones until everyone has been migrated. 2. Need to maintain current portability of the 7960's. (ie a user can unplug his phone from the internal LAN, take it home or wherever, and plugin and have the phone register and