similar to: Parking Lot CallerID

Came upon a problem today that I thought I'd see if it's by design, if I'm missing an option somewhere, or if my fix is the way to fix it. We setup a remote location with a server, same as we've done with others, but for some reason when they would transfer an outside call anywhere it would pause for a few seconds and hang up the line. Well, after spending most of the day on

Now on to another problem that we've had as far as I know since the beginning of using Asterisk 9+ months ago. I've been trying very hard to knock this problem out but regardless of what I do, it's still there. So, the problem is, when a call is in the parking lot, it then times out after whatever time frame and dials the extension that put it on hold. After 60 seconds of ringing

Hello, I am experiencing a weird problem on asterisk when I place an outbound call, park it and then retrieve it. I am using extensions.ael with macro and switch and I get something as SW_456_... that is autogenerated by asterisk when compiling the extensions.ael This doesn't happen when the call comes from outside. The bad CallerID is displayed only on Cisco 504G phones and it is

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

We are planning to have about 100+ parking lots defined in features.conf , each with about 4 unique park positions. Asterisk will be handling all the parking and unparking (we don't exclusively use Park/ParkedCall in the dialplan): [parkinglot_a] parkpos => 1-4 context=parked [parkinglot_b] parkpos => 5-8 context=parked As far as I can tell, Asterisk adds/removes extensions to the

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Hi folks, I'm stuck with a problem that I suspect has a trivial solution... I have a function, call it foo, that takes a number of arguments which it uses as parameters for a simulation. For later analysis, foo stores the names of the objects passed to it along with the simulation results in its output (written to a database). The objects names are accessed with deparse(substitute(argname)).

Any ideas? Thanks. [root@localhost root]# asterisk -r Unable to connect to remote asterisk [root@localhost root]# asterisk -vvvvvgcd Parsing /etc/asterisk/asterisk.conf Asterisk 0.7.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Parsing

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Hi guys Ive been trying to get this to work for ages now, basicaly im trying to do if ${woteva} = "" (nothing), or its none existenant then do label 1, else label 2. for my last called function, so it will play a different message if theres no last call in the system or it was anonymous. ive tried exten => 1000,1,GotoIf($[${last-call${CALLERIDNUM}} = ""]?4:5) and heaps of

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y

Hi all, I am trying to install asterisk on my system, the compiplation and installation process all seem to work fine (make ; make install ; make samples). But astersik fails to start. Is the sample configs not supposed to work out of the box? Even more confusing, it seems to fail at different points every time I start it, but this is probobly because of threads starting differently or something?

Hello all, I think I have most of my AAH 1.3 setup running (Asterisk 1.0.9), but somehow something is not quite right with my vMail setup. I would have sworn this was all working, but maybe I was just dreaming. Anyway here is what is happening, say I am on extension 200 and I want to call to extension 201. If extension 201 is no connected, then it rolls right into vMail with the message the