search for: chan_agents

O -fPIC -c -o chan_agent.o chan_agent.c chan_agent.c: In function `login_exec': chan_agent.c:595: parse error before '<<' token chan_agent.c:602: parse error before '>>' token make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels'

hello there: did some one compiled shadydial with asterisk on gentoo successfully, if some one plz help me I am getting compilation errors during asterisk compilation after replacing the files provided with shadydial thank you here is my log, please help gcc -pipe -I=/usr/local/pgsql/include -pipe -Wall -Wstrict-prototypes -Wmissing

chan_agent.c: In function `login_exec': chan_agent.c:595: parse error before '<<' token chan_agent.c:602: parse error before '>>' token make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1

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 using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

Hey, We've been working on tracking all inbound calls to certain call-centre members and have hit a snag, it seems when a queue delivers a call to an Agent, chan_agent will call Local/whatever but provide no means of associating the call in the queue with the local call. To that end i added in an event called AgentAssociate which looks like this: 'DestChan' =>

Is it possible to specify a 'wrap-up' time in a queue so agents will have a specified amount of time to complete tasks between calls unless they hit a key on the phone? As it is they can recieve a call moments after they hang up with no 'down time'. Thanks Jim Friedeck

I am beating my head against a problem where queue calls offered by Agent channel to a SIP UA cannot be REFER transferred if the target UA/extension hasn't accepted the call. If the members of the queue are SIP channels, this is not a problem. I suspect chan_agent isn't flagging the bridge from Zap/n -> SIP/n properly, or this is by design. The following line is what is spoken before

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

after recompilling asterisk (trunk-r75109) after system (mandriva cooker) update (new glibc 2.6, gcc 4.2.1), sound starts very choppy, when codec translation is performed, if translation isn't needed, it sounds OK any idea? until update, everything worked fine. I'm using ztdummy as clock source. during compile, I got lot of errors... ael_main.c: In function ?ast_context_add_ignorepat2?:

Yea, I have faked that with a silent mp3, but to do it right it should also be a config flag in the agent.conf file for each agent, prolly add another arg to each agent definition for the MOH class, & the arg 'none' means don't play music for that agent -----Original Message----- From: James Golovich <james@wwnet.net> To: asterisk-users@lists.digium.com

AgentCallBackLogin is going to be deprecated, so I've decided to emulate chan agent using AQM and RQM funcions and Local channel. I use asterisk 1.2.13 and latest 1.2.x. zapata. I used example 2 from http://www.voip-info.org/wiki/view/Agents+without+agent+channel and example from queues-with-callback-members.txt from asterisk 1.4 doc directory. My dialplan is very similar to Digium's

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I was wanting to use the new MuxMon application to record calls - it seems to be a "nicer" way of recording than the Monitor application. However, there is a slight issue with agents - we use the recordcalls option in agents.conf to record inbound agent calls - and I believe from looking at the source code that is uses the monitor application. Is there any way to get chan_agent to

Hello, I think Device State for Agents don't work correctly My configuration: agents.conf [general] [agent](!) autologoff=15 ackcall=no acceptdtmf=# wrapuptime=5000 musiconhold=default recordagentcalls=no custom_beep=beep [2000](agent) fullname=Fulano [2001](agent) fullname=Zutano [2002](agent) fullname=Mengano queue.conf (Agents Related) member => Agent/2000 member =>

Hi, am building the latest version of Asterisk (1.4.13) on a self-build Linux host (based on LFS-6.3). Version 1.4.12 built installed and worked fine. Last night I upgraded the kernel to 2.6.23 and rebuilt the zaptel driver package 1.4.5.1 against it. That seemed to build and install O.K. too. I dl'd the 1.4.13 source tarball and tried to build that: ./configure ran O.k. 'make

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

Hello Previous context :- After Looking up sip and IAX2 that require configuration at router level which may cause some problems like connection break etc... so i left them ......... and start wondering if there is some thing that dont require configuration at router layer. The task to accomplish to make and recieve calls from outside local network using any protocol whose soft phones are

...nnection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Nothing about chan_dahdi, app_queue, cdr or chan_agent But, for example if I change something in agents.conf (anything), the output of the "reload" in CLI shows me that chan_dahdi, app_queue, chan_agents, etc now reloads its configuration. isb152*CLI> reload == Parsing '/etc/asterisk/extconfig.conf': == Found == Parsing '/etc/asterisk/manager.conf': == Found == Parsing '/etc/asterisk/manager_additional.conf': == Found == Parsing '/etc/asterisk/manager_c...

So I have read and read and read... google is my friend and the wiki is by brother... However, I am still unclear on what the preferred method of using the pound sign is. If the Pound sign is set aside for Transfer.. Then when I make an outbound call to my bank I can not "Enter my PIN followed by Pound" Likewise if I turn off the ability to transfer when initiating a call, my bank pin