similar to: Queue problem, ringing agents.

Hello list! I?m having an issue when reloading Asterisk, I?ve had this problem in Asterisk 1.6.1.6 so I upgrade to 1.6.2.1 version, but I still have the same error. For example, I send a "reload" in Asterisk CLI and this is the output: isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found

>From voip-info.org: Queue(queuename|options|optionalurl|announceoverride|timeout) 'optionalurl' allows you to send a URL to devices that support it. Does anyone have details on the "devices" that support the optionalurl method of the Queue application? I am wondering if there is a softphone that supports this. The only thing that seems to happen is the

Hello, In my Asterisk server when i try to set the value for the queue option "Skip Busy Agents" in Freepbx GUI it is not being written into the backend file queues_additional.conf. As a result sometimes agents in queue gets calls when they are already busy with another call. So i set "ringinuse=no" option manually from backend. Is it bug ? Is there any fix for this?. I am

There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext=

I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can

Hello everybody, I am wondering if anyone know more about the fact that the official nextcloud documentation states it does not work with linux based smb servers for notify support? https://docs.nextcloud.com/server/latest/admin_manual/configuration_files/external_storage/smb.html "Due to limitations of linux based SMB servers, this feature only works reliably on Windows SMB

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

I'm not sure if this is part of Asterisk or FreePBX so I apologize if this is the wrong list to ask my question. As part of my companies call routes, I have a Time Condition for our tech support queue. I would like to add a Day/Night Control so the Time Condition can be overruled. However, I'm afraid someone will forget to turn it off again. What is the best way of resetting this control

On 7 May 2015, at 23:45, Tafadzwa Nyabasa <tnyabasa at gmail.com> wrote: > I am looking for a phone provisioning template for Snom phones, Yealinks and Polycoms. I am always doing deployments of many phones and usually configure each phone one by one for each installation. Any help will be highly appreciated There?s some excellent documentation about provisioning on the Snom Wiki:

Hi, I am trying to attach a Palm Z22 via USB to my fully virtualized Win XP. Dom0-System: CentOS 5.3 x86_64 (Kernel 2.6.18.128.1.10.el5xen) XEN: 3.0.3-80.el5_3.2 XP: 32-Bit Professional lsusb shows: USB Palm: Bus 002 Device 007: ID 0830:0061 Palm, Inc. Lifedrive / Treo 650/680 / Tunsten E2/T5/TX / Zire 21/31/72 / Z22 I found some hints that I have to put usb = 1 and usbdevice =

I need some help getting a system running for one of my company's plants. I am running AsteriskNow 1.7 with Asterisk 1.6.2.10 and FreePBX 2.8.0.2. My D-Channel keeps bouncing. The telecom tech told me he thought that I might be using the wrong sync source, and I think I might have been, but I changed DAHDI system.conf to "span=1,1,0,ESF,B8ZS" (from

Hello, My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port modular card and a single FXS module. Got the Rhino card installed and the machine sees it: root at pbx:/etc/dahdi# dmesg | grep rcbfx [ 71.985309] rcbfx 0000:04:00.0: PCI INT A -> GSI 21 (level, low) -> IRQ 21 [ 71.985440] rcbfx 1: Rhino PCI BAR0 50100000 IOMem mapped at ffffc90008d7c000 [ 71.985504]

I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009 ; If you edit this file and execute

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

I'm new to Asterisk and inherited this project so I apologize if this question has been asked a hundred time before. I did start with Google but I may not be asking the right questions, because I wasn't finding any answers. I have Asterisk 1.4.24 and FreePBX 2.5 running and using a Digium TDM800P to interface with our six analog phone lines from the telco. Currently I have a single trunk

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

Good day I am looking for a phone provisioning template for Snom phones, Yealinks and Polycoms. I am always doing deployments of many phones and usually configure each phone one by one for each installation. Any help will be highly appreciated Regards -- Tafadzwa -------------- next part -------------- An HTML attachment was scrubbed... URL:

[root@asterisk asterisk]# asterisk -V Asterisk SVN-branch-1.2-r8632M I was wondering if there was some documentation I was missing on the '*' key and queues. I have my features setup to use *x, where x is a #, but these features don't work for calls coming in from a queue. As soon as the '*' button is hit, the call is disconnected. I have a vague memory of reading about

Hi, We have a PRI Trunk (physical E1) and we are getting some rather weird and very isolocated problems. On outbound calls to specific numbers, it would seem to me that DTMF from the remote side is affecting the local asterisk system. Basically what happens: - We make a OUTBOUND call via the PSTN (PRI Trunk) to a remote System - Remote Answers, and converse - Remote sends DTMF on their site to