similar to: loggedoff extension - why does * say "isonthephone"

Same here, I just changed the b to u. Unavailable message is more generic, but it beats it saying busy when its not. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Henry Devito Sent: Tuesday, October 05, 2004 8:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}

Hi, I have two local SIP extensions (both bt100). One is on remote location behind another nat (16), but everyithing seems to be setup correctly as it can register and is listed as OK(57ms). However I can only call in one direction between those two. Extensions are defined in same context: exten => 11,1,Macro(oneline,SIP/11) exten => 16,1,Macro(oneline,SIP/16) both using same macro

Hi!, I'm trying to set up a Queue (which works fine now :-) Sip clients can login in to the Queue with dialing 91 on there phone. And as soon as there are customers the Queue calls the agents back. I would like that the queue calls the agents also if it's phone is call-forwarded. With agents (sip clients) are added with the following extensions: exten => 91,1,AddQueueMember(myqueue)

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Is anyone out there using app_dbodbc (http://www.bkw.org/~brian/app_dbodbc.c)? Any problems with it? I was able to get it all working, but it causes * to segfault every now and then. It does not appear to be related to any specific function (ODBCget,ODBCput,ODBCdel,ODBCdelltree). It is 100% repeatable. If I noload the module, everything works fine, but when its running, after calls to any of the

Hello I've been having the following problem today : I have a quite simple dialplan made to receive a fax: [answer-extension] exten => 1,1,Answer exten => 1,2,Macro(setcallerid) exten => 1,3,Ringing exten => 1,4,Wait(3) exten => 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},$ {EXTENSION}) exten => fax,1,Goto(faxreceive,1,1) The Wait(3) is there simply to let

Copying to asterisk-users, as it's of use there too. I copied this code years ago from the net, it may have been modified since... This however is only used by managers, as it allows the manager to log a user in and out. For agent logged in/out status: where 8501 is the queue number and 8512 is the agent's extension, and SIP0001 is the agent's device. in extensions.conf

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

Hello, When a user forwards his SIP phone to another extension (say an absent boss to his secretary) I'd like the unanswsered forwarded call to end up in the new destination's voicemail. With my current diaplan the call is handled by the original recipient's voicemail: [macro-stdexten] exten => a,1,VoicemailMain(${MACRO_EXTEN}) exten =>

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi all, I'm sure this has been done before but I just can't figure it out. On my * box I have a simple IVR: [tolc_menu] ; Welcome and information to callers exten => s,1,Answer() exten => s,n,Wait(2) exten => s,n,Background(welcome-to-tolc) ; Say Hello exten => s,n,Wait(1) exten => s,n(tryagain),Background(enter-ext-of-person&or) ; Enter extension number if known, or

Hello, and I'm glad to be a member of this list. Perhaps an enlighted person can help me. An entry in my extensions.conf has: [macro-stdexten] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Wait(1) exten => s-ANSWER,2,Monitor(wav,record) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(default,s,1) exten

Hi List; I have this example for Macro and I am not able to understand some line, if any one can help me plz :)- [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,2,Goto(incoming,s,1) exten

Hi, I'm just in the process of replacing a crappy Siemens PBX with a new and shiny Asterisk system. To connect Legacy equipment I hooked up a small ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI card. That port is configured for NT Point to Multipoint (Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX to the other Asterisk extensions but the other way

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??