similar to: dialplan logic questions: macro not reaching DIALSTATUS based extension (s-ANSWER)

Hi, Alistair Cunningham of Integrics was our guest yesterday. We talked about Integrics new product Geons, a suite of software for building large-scale distributed enterprise applications. The recorded session is now available here: http://www.voipusersconference.org/2010/geons/ The extremely rare John Todd was sighted (and heard) at this event. If you are developing a product or service

Greetings everyone, today I spent the last part of my day trying to find a parse error inside this snip: http://pastebin.ca/740081 If there's anyone who can shed some light on why my GosubIf condition is throwing a parse error, I'd greatly appreciate your insight. This was really frustrating and is probably a stupid mistake. Regards, -Michael

In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in the s extension. Goto() is used in examples. Is the prefix "s-" mandatory? Is it related to the original extension "s"? (Apparently Goto(${DIALSTATUS}) won't work for me.) Yuan Liu

Hi, I am trying to pass DIALSTATUS to a Macro so that i can set a variable when a call is placed (call is placed via a call file to another extension first). Basically i don't want to dial a number where a call is already bridged and thats why i am setting a variable. [macro-afterdial]; exten => s,1,Goto(s-${ARG1},1) exten => s-ANSWER,1,SetGlobalVar(NUM${ARG2} = "ACTIVE")

Outgoing call on Zap channel always gives DIALSTATUS ANSWER even when BUSY? Has anyone run into this? Here is my conf files: Zaptel: span=823,1,0,d4,ami e&m=1-24 loadzone = us defaultzone=us Zapata: usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes

I'm running Asterisk CVS-v1-0-12/20/04. I'm using PHP with Manager API Here is the code: #################################################################### # Make call #################################################################### $socket = fsockopen($ask_db,"5038", $errno, $errstr, $timeout); if (!$socket) { echo "$errstr ($errno)<br /\n"; } else {

I have a list of VSPs that I use. Some are not able to terminate to different locations. It appears they are returning this error message: Sep 13 00:01:43 WARNING[22093]: chan_iax2.c:6835 socket_read: Call rejected by x.x.x.x: No such context/extension I would like to find out what the dialstatus is on this so I can try a different VSP that is able to terminate the call. Right now I have this

Hi all, I've been working on debugging a bit of a custom dialplan system, and seem to have run into some issues on our development server. Hopefully someone can give me some pointers on this one! =) In a nutshell, we have a hangup extension that's being triggered to feed data back to an api on our main webserver that seems to be randomly "losing" the dialstatus channel

Hi All, A couple of weeks ago I noticed Askozia PBX, which is a new embedded Asterisk & OS distro at http://askozia.com/pbx. This caught my attention for two reasons; it uses v1.4 of Asterisk, and it uses the m0n0wall development framework to build on FreeBSD with a PHP based GUI. I've used m0n0wall for years, and FreeNAS also, which shares the same OS/GUI framework. I booted the latest

Greetings everyone, I've been working on a (yet another) "all-in-one" Asterisk based project. It is aimed at embedded / low power systems (but scales fine on more capable hardware) and is based on Asterisk 1.4.x and FreeBSD 6.2. Because of this, I've mostly been hanging out on the asterisk-bsd list as bugs rolled in and the system's features were improved. We're

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

I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing. Below is a dialplan snippet and the resulting CLI output. This is running in an 'h' extension. Noop(DIALSTATUS=${DIALSTATUS}) Noop(CDR(disposition)=${CDR(disposition)}) -- Executing [h at pbxmax-dial-simple:1] NoOp("SIP/msx_01-0000005b", "DIALSTATUS=ANSWER") in new stack

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is

Hi, I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is the Cannes Festival of the Asterisk world) and begged him to take time off from porting Askozia from FreeBSD to linux to tell us about his project and the problems of porting as well. Hope you'll all be able to stop by and lmisten or download or stream the recordings. We are having a lot of people call in on widenabd

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 All I am trying to simplify a dialplan for a few thousand users. Would what I have below work? If someone dials exten 710001 would it go through answer and then to the macro to try dialing the SIP phone thats registered on 710001 and then onto voicemail if no answer or not signed on? exten => 71XXXX,1,Answer() exten => 71XXXX,2,Macro(71macro,${EXTEN}) exten => 71XXXX,3,Hangup()

I'm looking for a web GUI to Asterisk that I can run on some small embedded hardware. I've used FreePBX in the past but the overhead is not to my liking and it is entirely too complicated. I do not wish to change my entire OS just for the GUI either (aka AstLinux). Is there anything out there? I'd like to have only a small set of features, primarily the configuration of extensions,

Hello, I'm new to asterisk and this list. The ISO download appears to have 1.6 with the FreePBX GUI but I am looking to use the Asterisk GUI. The only option for the Asterisk GUI is to use 1.4. Is it as simple as installing 1.6 only then using the yum repository to install the Asterisk GUI? If so, what packages are needed? Thanks!

Hello I'm looking at my options to build a compact, silent, headless Asterisk server to handle one or two FXO ports. Out of curiosity, I got one of those babies on eBay for 20E: http://silicon-verl.de/home/flo/software/netstation-8364/ Before I spend time on this, can someone tell me if... 1. it's easy to install Linux on those things (am willing to add a CF-to-IDE adapter to use a

Hello my friends, I've a doubt, i want to be able to forward the incoming calls from PSTN to my cell phone...i mean, qhen i'm out of the office i need like aq macro that helps me to forward the incoming call that goes for example to my internal extension SIP 207, i 've this macro but i can make it work properly....i can't activate the forward in the phone, is quite confuse: