search for: architel

...(MACRO_RESULT=CONTINUE) exten => t,1,Set(MACRO_RESULT=CONTINUE) ;----original macro ;exten => 1,1,NoOp(Caller accepted) ; Do not set MACRO_RESULT to anything to connect the caller ;exten => i,1,Set(MACRO_RESULT=CONTINUE) ;exten => t,1,Set(MACRO_RESULT=CONTINUE) [default] include => architel include => local include => trunkintl exten => _6XXXXXXXXXX,1,Dial(ZAP/G1/${EXTEN:1},40,M(screen)) exten => _6XXXXXXXXXX,2,Hangup [findme] exten => s,1,Playback(transfer_csp-male) exten => s,n,Dial(LOCAL/6${findme1}&LOCAL/6${findme2},40,m) exten => s,n,Hangup()

...queue was not even trying to record ; according to the asterisk console (still doesn't) Set(MONITOR_FILENAME=support/${EXTEN}_${CALLERID}_${TIMESTAMP}.wav) monitor-format = wav monitor-join = yes joinempty = yes member => SIP/72008 member => SIP/72009 -- Andy Hester Network Engineer Architel

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

Hi, Being new to Rails and Active record could someone please check my database diagram: http://spectrais.com//images/0.png I am building a trouble ticket system for my company. Simply put a client has many users. Those users can create trouble tickets. Each trouble ticket can have many journals that track the progress of the ticket. An employee (of our company) can take ownership of a

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