similar to: Manager - QueueAdd

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list What's the syntax of the QueueAdd command for the manager ? I want to add interface FOO to queue BAR using a manager. Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFBWDAVJUm/Bor63CERAhbtAJ0YykdT+TNTNKXlC3NldtzCWLo1owCfZEoz

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Hi, I'm currently migrating to 1.4 and have problems changing deprecated AgentCallbackLogin to AddQueueMember. I have dynamic queues and dynamic agents (MySQL Realtime), and pseudo-dynamic agents.conf (with huge amount of possible agent numbers). Agent login is done trough manager API: * AgentCallbackLogin * QueueAdd In 1.4 seems AddQueueMember can do all the same, but there is no such

I have a queue I configured as "strict" and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy).

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

Hi, Asterisk 1.4.31 here. I have a queue with members defined, and those member have member names "member 1", "member 2", etc. They are in a realtime DB. When I modify those member names (column membername) the changes aren't reflected in the queue status ("show queues" from cli. They aren't reflected when a new call comes in, or when I reload the

On 03/24/2015 04:28 PM, Richard Mudgett wrote: > > > On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere <jeff at jeff.net > <mailto:jeff at jeff.net>> wrote: > > > Hello, > > I am wondering if asterisk does anything at all to RTP packets > passed from channel to channel if no transcoding is involved? Can > I assume that the packet that

Hi Everyone, I am doing a queue show and I can't see any column that shows a queue member priority. Is there any other command that can show the member priority on the Asterisk 1.4x CLI? We are using this format of dialplan to login agents: exten => 123,Answer() exten => 123,n,AddQueueMember(500|Local/${CALLERID(num)}@from-internal/n) exten => 123,Hangup() ^^^^ Where 500 is the

Guys, I have a problem. I have a queue on asterisk 1.8 that members are added dynamically via the AMI QueueAdd. When you run the CLI a "reload app_queue.so" all members who were in the queue disappear. This is a bug or some parameter that I do not know? Would have another way to do the reload queue without any risk to members who are already in it? tks Ed -------------- next part

Hi Marie Thanks! I was just worried about thread safety if I had to use a global variable, e. g. it might be set to a value by one call (since I'm using the same global for every incoming call to transfer the accountcode gotten from my HTTP endpoint to the same macro, and there can be several calls simultaneously all inserting HTTP-sourced values at more or less the same instant) and then

On 6/26/23 5:19 PM, Jeff LaCoursiere wrote: > On 6/26/23 9:00 AM, Joshua C. Colp wrote: >> On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: >> >> I am connecting to the ARI with subscribe all, so I can see >> channels being created.  I now want to extract a variety of >> header variables (at the moment the from and to tag).  I

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

Hi, We recently dumped a Xorcom box that was no end of trouble and replaced with a Digium G100.  PRI came right up, and we have been using it fairly flawlessly for several months now, with one caveat.  Calls that arrive from the PRI are sent to the asterisk instance (13.23.1, chan_sip), then routed by the dialplan to various other gateways or upstream providers.  When the call finally lands

Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

On 6/26/23 9:00 AM, Joshua C. Colp wrote: > On Mon, Jun 26, 2023 at 10:57 AM TTT <lists at telium.io> wrote: > > I am connecting to the ARI with subscribe all, so I can see > channels being created. I now want to extract a variety of header > variables (at the moment the from and to tag).  I tried to read > them from the ARI but Asterisk refuses since the

I’m in training, so I have to demonstrate something SIP related. I figure it would be cool to hack a call, hanging it up while in progress from outside Asterisk. Doing so will demonstrate use/knowledge of ARI, AMI, SIP, route-sets, UDP, etc. Practical value: zero :) Who knows, maybe this will have an actual application for someone someday. In practical terms I think building a proxy

First, I apologize if this info has been covered, I tried doing a search on Google and the wiki and found nothing, but I could be searching for the wrong info. Second, I am new to asterisk and linux. My knowledge is Asterisk is much better than linux, so excuse the ignorance in some areas. Configuration: Novell Linux Desktop Asterisk 1.0.6 Intel CPU/MB TDM12P digium card (I think that is it -

Hi, is it possible to create a Samba share with full Windows ACL support based on a mounted NFS share? What is needed e.g. as NFS mount options? Regards Tim