similar to: Ackcall works for sip, not for zap

See comments below. Henry Devito wrote: > Here's the problem. When I call 555 to login, it asks for the agent ID > which I enter as 501, it asks for the password which I enter as 1234, > then it asks for the extension I dial 501 It then says that extension is > not valid. What am I missing? Of course 501 is valid I can make and > take calls from it now. > > >

First, many thanks to Greg Boehnlein for his patch to chan_agent.c for adding a "preackannounce" option. I am running CVS HEAD from 2005/07/31, and the patch failed in a few hunks, since the code was refactored to add in some CASE statements where there were compound if statements before. Anyway, I have successfully updated the patch to work against head as of 3 weeks ago, and would

I am trying to get remote extensions to work correctly with agents. I have ackcall=yes and have agents logged in to extension 101 using agentcallbacklogin with extension 101 defined as: exten => 101,1,Dial(Zap/3/18165551234,20,tTA(custom/presspoundtoanswer)) This setup works great on local and/or voip channels, but on zap channels, the zap channel answers immediately as soon as it goes off

Hi, I am on my next venture now, Need to set up 3 queues. I would like these setup using the agentcallbacklogin. Does anyone have an example of this? I have looked through the wiki , but all that did was confuse me. One of the problems I'm having is how do I configure my extensions.conf to dial the agentcallbacklogin -------------- next part -------------- An HTML attachment was

Hello, I upgraded to cvs-head over the weekend and now agents that are logged in on zap channels have to acknowledge ACD calls by pressing #, even though I have ackcall=no in agents.conf. This doesn't seem to be happening to agents on SIP phones, and it this is the first time I have had an agent logged in on a Zap channel since the upgrade. Is there a new config param that I am missing?

Can anyone recall a bug where asterisk crashes after a callback agent presses the ackcall key in 1.2? The last logged item was before safe-asterisk restarted was "delaying member connect for 2 seconds" We have seen it two times on a heavily loaded server (1.2.12.1), but cannot find anything documenting a bug or fix. Could be coincidence, but a google search showed at least

Check my reply to your last post. Use SetGroup and Checkgroup before sending the call to your agents. Robert Jackson -----Original Message----- From: Henry Devito [mailto:hdevito@qwest.net] Sent: Thursday, September 30, 2004 10:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Queue Setup almost got it Ok I think I have the queue

I am running r79979 of Asterisk Trunk, and I am having problems trying to use app_queue.so. I want to use the extension 510 to be a line where users can call technical support. Extensions 511 and 512 are used by the operators to dynamically make themselves a Queue Member or not. So, operators call 511, and they should get added to the Queue as a Queue member. When users call 510 then, it

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

is it possible to do with asterisk something like this how ? maybe some copy of extensions and some other file ..pls. call to asterisk server -> user 100 is busy -> recorded msg say "all lines are busy , pls wait" ,"you are second caller in 'queue', pls wait" -> caller is on hold till user 100 busy then ring user 100 (ext100) another call to asterisk

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

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

Hi; I've been using Asterisk for a few months now, and I have run into an interesting issue that I thought someone else in the community may have run into: I have an Asterisk install set up to receive helpdesk calls, route them to several IAX extensions and an extension which is simply a forwarded call over the POTS to a cellphone, so that if no one is logged into their IAX extensions for

Hi, After reading https://llvm.org/docs/Vectorizers.html#vectorization-of-function-calls I decided to write the following C++ program: #include <cmath> using v4f32 = float __attribute__((__vector_size__(16))); v4f32 fct1(v4f32 x) { v4f32 y; y[0] = std::sin(x[0]); y[1] = std::sin(x[1]); y[2] = std::sin(x[2]); y[3] = std::sin(x[3]); return y; } v4f32 fct2(v4f32 x) { v4f32 y;

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

I've just installed asterisk as a new phone system for our office but am having difficulty with the queues. Specifically I need a way to redirect our sales queue to voicemail when no one is logged in to the queue. I see I can use the joinonempty=no setting, however this setting doesn't work if you use the agent functionality (at least not with AgentCallbackLogin). I could, of

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

I've been working on setting up a few queues and agents for use in our upcoming call center. Agents can log in and customers can call in and be routed properly, but I've run into one strange issue. When an agent dials 98 to login the "agent-incorrect" file is played, then the agent can login as normal. Searching the archives I found this issue brought up in Feb '04 :

Guys I am having some serious issues with my call queue and Management is breathing down my neck pretty bad, and I am running out of ideas. I have a single queue for my tech support department. I originally was using the AgentCallbackLogin for them and it tested out great on our testing weekends, but it hasn't worked out since. It would only let one of them take calls at a time, no matter