search for: _6xx

I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but

...07,Hangup include => dial-sip include => dial-e100p [from-sip] include => dialstring include => dial-sip include => out-voip include => dial-e100p [dial-sip] exten => 600,1,Dial(Zap/g1/100,60,tr) exten => 9600,1,Dial(Zap/g1/100,60,tr) exten => _6XX,1,SetMusicOnHold(random) exten => _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN}) exten => _6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-$ {STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49) exten => _6XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0)) exten =...

...07,Hangup include => dial-sip include => dial-e100p [from-sip] include => dialstring include => dial-sip include => out-voip include => dial-e100p [dial-sip] exten => 600,1,Dial(Zap/g1/100,60,tr) exten => 9600,1,Dial(Zap/g1/100,60,tr) exten => _6XX,1,SetMusicOnHold(random) exten => _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN}) exten => _6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-$ {STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49) exten => _6XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0)) exten =...

...string > > include => dial-sip > > include => out-voip > > include => dial-e100p > > > > [dial-sip] > > > > exten => 600,1,Dial(Zap/g1/100,60,tr) > > exten => 9600,1,Dial(Zap/g1/100,60,tr) > > > > exten => _6XX,1,SetMusicOnHold(random) > > exten => _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN}) > > exten => > _6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49) > > exten => _6XX,4,MixMonitor(...

...; Activate the message waiting light if this ; voicemailbox has messages in it And an abridged extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes [bogon-calls] exten => _.,1,Congestion [from-sip] include => parkedcalls exten => _6XX,1,Dial(SIP/${EXTEN},30,T) exten => _6XX,2,Voicemail(u${EXTEN}) exten => _6XX,102,Voicemail(b${EXTEN}) exten => _6XX,103,Hangup exten => _04.,1,Macro(dial-mobile,${EXTEN}) [macro-dial-mobile] exten => s,1,SetGlobalVar(NumToDial=${ARG1}) exten => s,2,SetGlobalVar(theCHANNEL=ZAP/3...

I am checking on the SetGroup / CheckGroup commands, but I have some troubles to undestand the examples. SetGroup(moh) can be moh anything as I like? Usually moh stands for "music on hold" CheckGroup(1) checks if somebody in in group "moh". Does it mean I can only have one SetGroup(xxx) ?? When I look at example 2 than I see two SetGroup commands and one CheckGroup

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no answer) to a specific number on * (5901) that is my x-lite software client. If 5901 is

I am revising our DialPlan strategy for our Asterisk system (1.4.2) and looking for some info on 'best practices' for this. Here's what I'm trying to do: I have an ACD menu that gives the caller the options as follows: - Press 1 for sales - Press 2 for support - Press 3 for customer service - Press 8 for a 'Dial by Name' list or enter the extension number at anytime

...ur grandstream 100 phones. The phones use SIP, asterisk is 1-0-5 on debian GNU/Linux (sarge). Outside connections are via 2 ISDN BRI (British Telecom) lines using 2 billion isdn cards and bristuff. The phones are set up to be in context [internalphone]. I numbered all the internal extension with _6XX That works well with early dial. As soon as the 3rd digit is dialed, the phone connects to the internal extension. I also have an extension like so: exten => _9X.,1,Goto(dialout,${EXTEN},1) [dialout] exten => _X.,1, <magic agi scripts to resolve callerid for billing purposes> ... ......

Hello, i want to use asterisk as a gateway for H323-Phones. But i cant get it work. I'm using a gatekeeper on another computer. My IP-phone is registered there. Does anybody can sent me an oh323.conf and extension.conf as examples? Thanks in advance Erik Bastian -- NEU : GMX Internet.FreeDSL Ab sofort DSL-Tarif ohne Grundgebühr: http://www.gmx.net/dsl

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a