similar to: Agent timeout then Dial() ?

Hi All, We are using Asterisk 1.2.18 in a CentOS box. Implemented a queue (maqueue) structure for handling customer calls. There are 4 queue members (85744,85766,85511,84888). These 4 members are logged in using AgentCallbackLogin application. But at some point, one of the agent's SIP phone does not ring for an incoming call to this queue. I checked the agent status and it is not in paused

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4 FreePBX = 2.11.0.4 I am trying to understand flow control in Asterisk dial plans and not having very much luck. I have read the Asterisk book from O'Rielly, or at least those parts I believe might apply, but that has not helped me much on this particular issue. What I wish is to set three distinct ring tones on our Snom phones for

Hi, Looking around I can't seem to find a way to show the number of agents currently logged into a queue and if possible who they are. Is there a way to do this ? Thanks -b -- ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the

Support vector machines (SVMs) have established themselves as one of the preeminent machine learning models for classification and regression over the past decade or so, frequently outperforming artificial neural networks in task such as text mining and bioinformatics. Dr. Lutz Hamel, author of "Knowledge Discovery with Support Vector Machines" from Wiley will present his online course

I thought other might benefit from this lesson learned and thought it maybe should be added to the man-pages. I thought my network connection was glitchy and hence set rsync up for --timeout=120 but I found out that I was actually causing the glitch with this script: #! /bin/sh - while true; do rsync -avz --progress --timeout=120 --delete /media/rsync_gb01/movies/ myserver:movies; sleep 120; done

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

Here is the current dial plan section: [custom-michael] exten => _900,1,Playback(custom/extn-xfer) exten => _900,2,SayDigits(${EXTEN}) exten => _900,3,MixMonitor........... exten => _900,4,Dial(SIP/${EXTEN}|${DEFRT}) exten => _900,5,Playback(custom/extn-xfer2) exten => _900,6,Goto(custom-michael,901,4) exten => _901,1,Playback(custom/extn-xfer) exten =>

Hi, Just got some CISCO 7960 phones. They seem to work great except if I make any SIP call using the speaker phone (leaving the hand set in the cradle)the call will disconnect in about 5 or so seconds. If I pick up the hand set and make a call, it's fine. Has anyone else run into this ? Any solution ? The phone is on SIP v6.1 - it did the same thing on 4.4 5.0 and 6.0. Thank you in

Hi, all I'd like to configure Asterisk to receiving call from PSTN. After PSTN phone call in, Asterisk will prompt user to enter a number, then Asterisk will transfer the call to a SIP phone by this number. Please help me check the following extensions, is that OK? thanks! [from_pstn] exten => _.,1,Answer() exten => _.,2,GoTo(Xfer,s,1) [Xfer] exten =>

Can anyone offer any insights as to why with one of these examples I can do a dial to the sip hone, and with the other I can't? DOESN'T WORK: -- Executing Dial("SIP/109-d35d", "SIP/101|5|tr") in new stack -- Called 101 -- SIP/101-c9ff is ringing -- Nobody picked up in 5000 ms -- Executing SetCallerID("SIP/109-d35d", "Xfer Andrew

I`m having (a lot of) trouble changing the call parking timeout behavior. This is my SIP context [internal-local-only-hamel] exten => s,1,Hangup include => parkedcalls What I am trying to accomppish is a quick test where I park a call, wait 45 seconds, and it hangs up. Here is my execution in the CLI: == Parked SIP/0004f2134384-1-0943e8a0 on 101 at parkedcalls. Will

I have a group of agents logged in to a queue that is set for ringall. The agents are set to auto logoff if they don't answer in 15 seconds incase they step away without logging out. That works fine, however, if they are on a call and a new call comes in, they are getting logged out too. The phones are ATA's connected via SIP. One thought is that the phones may be allowing a second

hello my friend has the proxy set up his extention set up his password set up but he isn't getting a dial tone is there a second setting we need to put the address in? he is going to advenced settings line1 and in the proxy address box he is putting the info in below is the way he has it set up Sipura SPA Configuration Sipura Technology Inc Info System SIP Provisioning Regional Line 1 User 1

Hi, I am having a very weird problem with call parking. I have defined call parking correctly, as it work well when parking calls and picking them up. The problem is what happens after the the 45 seconds have expired. The behavior wanted is that the person who put the call on "park" is called back after 45 seconds. What ACTUALLY happens is that the phone who got put on park

Dear all, I would like to solve a trivial problem (I guess it is) but can't find the right way. Maybe someone can help me ? I've got a table with two factors (station = station ID, buffer = buffer size in meters) and a value for each unique combination of those two factors (S = number of habitats within each buffer around each station) like this: TABLE 1 station buffer S Abaia01 200 2

Hi, I tried to sync two files (50 MB and 100 MB) with my webdav folder using rsync 3.0.5 with Mac OS X (10.4.11) Terminal and X11. With the Terminal application the sync always failed with both files. With X11 I once was successful synchronizing the 50 MB file. The error message is always like this (also when I set the timeout to eg. 7200 seconds): io timeout after 1003 seconds --

I realize this may be less of an Asterisk question and more of a... well... everything but asterisk, but still relating to asterisk question. I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between

Hi List, I'm attempting to set up a queue and agents using agent call back. This is all working fine with the queue and the agents login etc However. In my dial plan I a set variable when a call is entered into the queue to identify the origin of the call, then when the agent is called I test to see if the call is from the queue. If it is, the dial plan does not go to VM if the

Good Morning/Afternoon/Evening. I'm a bit dejected at the moment as all my googleing / howto reading has brought me no joy, Apparently there are people out there who have successfully set up Samba as member server in M$ 2003 ADS. However I have miserably failed / given up. What I want to do is have my people print to a PDF printer on Linux using cups after which they will be emailed the

Hello everybody, Can someone explain [gGrR] in Dial() function? To dial external extension 18005551212 over channel 2 we will use: Dial(DAHDI/2/18005551212) To dial external extension 18005551212 over one of channel from group of channels (nr 2) we will use: Dial(DAHDI/g2/18005551212) So lets assume that group 2 consists of 5 channels. How does Dial() function choose channel: - randomly? -