similar to: ChanIsAvail

I'm testing ChanIsAvail context and it is not working for me. exten => 55,1,ChanIsAvail(SIP/11&SIP/21) exten => 55,2,Cut(theChannel=AVAILCHAN,,1) exten => 55,3,Dial(${theChannel},r) exten => 55,4,Hangup exten => 55,102,Goto(s,4) It is not dialing SIP/21 when I'm talking on SIP/11, it execute Hangup instruction instruction. According to notes: The channels are checked

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

I sent this earlier under "Editting variable contents" but no-one has responded. So, the subject is now more to the problem, instead of the solution I was trying to implement. ChanIsAvail returns the channel ID plus "-<session>". How can I edit ${AVAILCHAN} to remove this session ID, so I can use its contents in a subsequent Dial statement? Dialing on Zap just gives a

Received the following message in mail to root: Message 257: >From root at desk4.localdomain Tue Oct 28 07:25:37 2014 Return-Path: <root at desk4.localdomain> X-Original-To: root Delivered-To: root at desk4.localdomain From: mdadm monitoring <root at desk4.localdomain> To: root at desk4.localdomain Subject: DegradedArray event on /dev/md0:desk4 Date: Tue, 28 Oct 2014 07:25:27

Hi, I want to create this system : Desk1 SIP Phone <adsl>------------------------<adsl> Desk2 SIP Phone | | <adsl> Desk3 asterisk Server My question is : when Desk1 call Desk2 , server (desk3) will authentificate phone but i want to known if Desk3 use

I am trouble finding a configuration that works for ChanIsAvail and SIP. My two providers are Voxee and Teliax. I have these lines in a macro exten => s,n,ChanIsAvail(SIP/teliax&SIP/voxee) exten => s,n,Cut(CH=AVAILCHAN,-,1) exten => s,n,NoOp(AVAILCHAN= ${CH}) ; Dial the available Channel exten => s,n,Dial(${CH}/${ARG1},60,t) Looking at the execution, I can see what the AVAILCHAN

Hi, all. I'm making my first foray into AEL. I'm starting with a simple macro, but I've already encountered an odd behaviour. I'm wondering if someone can shed some insight: Asterisk 1.2.9.1 macro newPlaceCallPSTN { s => { TIMEOUT(absolute)=7200; NoOp(Analog Out List: ${ANALOGOUT}); ChanIsAvail(${ANALOGOUT}); NoOp(Available Out List:

Hello I am trying to setup ChanIsAvail function in the extensions.conf file so that user should use the available channel to call out, but immediately after the function like, zap channel hangup. Here is the copy of my extensions.conf file and messages display on consol while making the call. Please help me to fingure out this issue. Thanks Deepak Extension.conf : exten =>

Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro. The logic is simple; try Zap/g1 (a group of two E1s), and if that fails, try locating a channel via DUNDi. Here's a massively cut down version to illustrate the problem I'm having. macro dialout ( dest ) { ChanIsAvail(Zap/g1); noop(Value of AVAILCHAN is ${AVAILCHAN});

A call comes in and goes into the queue, the queue dials a sip channel using a macro. The macro plays a set of options to the callee and if the callee presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends. For some reason the caller goes back into the queue rather than continueing on in the dial plan. Why is this, i could have sworn in 1.2 if i set MACRO_RESULT=CONTINUE that the

In 1.4.43 I would see things from "core show channels" like DAHDI/18/xxxxx for line 18 in Asterisk 11 its DAHDI/i4/xxxx How do I get the line number back? Jerry

I have a USB to serial converter attached to my box. pl2303: Prolific PL2303 USB to serial adaptor driver if I login to the box and send/receive serial commands over this unit it works without error EVERY time. however, if I run the same command set from with-in the extensions.conf with System() I get errors in dmesg like "pl2303 ttyUSB0: pl2303_open - failed submitting read urb, error

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

Hello All, I have managed to get my cisco and asterisk able to talk to one another I think. But cannot make a call from a phone behind call manager to the asterisk server. I have followed the cisco asterisk integration on the wiki. I have also setup a number 3000 for dialing for current local time and date on asterisk. I can call from a sip phone behind asterisk, no problems. The problem

Some progress to report: I had to run asterisk as the user logged in - actually not even that. I could not "su user -c " to that user - I had to run it as that user. Then I did a test and got audio! Great... But when I do a second test. Asterisk HANGS on ChanIsAvail() Then I thought lets SKIP that - and just let it do the Dial() - I stopped everything - got it running again. - and

How I change my configuration to reduce this issue. I have this setting on my zapata.conf signalling=fxs_ks group=1 callgroup=1 pickupgroup=1 channel=1 signalling=fxs_ks group=2 callgroup=1 pickupgroup=1 channel=2; singalling=fxs_ks group=3 callgroup=1 pickupgroup=1 channel=3; singalling=fxs_ks group=4 callgroup=1 pickupgroup=1 channel=4 and for outbound calls I have this context on my

I'm trying to add an emergency dial to my context. However, when I try to dial it, I get caught in an endless loop. For debugging, I have pared out nearly all the control flow and just have ChanIsAvail() and Dial() called. Using two different extensions to call teh same number, I get two different actions by *. Here is the vvverbose output: -- Starting simple switch on

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

Hi folks, I'm sure I had this in a previous life :-) Basically the ability to dial with autoselection of either IAX2 or SIP depending upon the registration of the endpoint. Ok, I have probably missed it in the wiki as well..... hints ? Gary .