similar to: Agents without agent channel

Greetings fellow list members, I have what I think is a relatively simple question, but it did not appear to be addressed on the wiki. I am trying to setup a queue so that it plays an estimated holdtime announcement, but not a queue position announcement. Currently my dialplan does both, and while I know how to take out the estimated holdtime without affecting the queue position

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

Given that all agents are on SIP phones (Cisco 7940), and all outside lines are EuroISDN channels (using a TE405P), is there any way of finding what zap channel is being used by an agent channel within a dialplan ? If you type "show agents" on the CLI you get information like: 6038 (Agent 6038) logged in on SIP/6908-8445 talking to Zap/63-1 (musiconhold is 'default') What I

Hi All I have a problem with queues that I have been trying to solve for many months - the customer has now picked back up onto this and wanting a solution - any guidance, ideas or solutions welcome. This is the situation :- We have a number of agents in a ringall group, a call joins the queue and all handsets ring - great. An agent answers and handles the call - great. The call

Hello, Just to let you know, our popular tutorial on setting up Asterisk for call centres has been updated. The tutorial covers everything from initial Asterisk installation to full call centre configuration with dynamic login and hot-desking support for agents. The old version of the tutorial used Agent Channels (e.g. Agent/1001) to distribute calls to agents through the Queue()

Dear List after a lot googling and watching source example of FastAGI i cant find a simple way to convert a very simple perl AGI script... perhaps im not a developer.. Why i have need to use FastAGI?...Very load CPU usage on my box... with only 100 calls.. So i have two way res_perl or FastAGI on some other box.. I cant test res_perl becasue when i try to compile it i have this error:

Time to reboot and re-start Asterisk, well, hrrm, monthly, news. It's been a hectic fall with a lot to do, both before and after Astricon. At this time, we're preparing for two Astricon shows in 2005. And no, we haven't made a decision on where to run the European Astricon, not yet. I am preparing to travel to the USA again this coming week. Today, I'm spending my time finding

Dear all, now we have the same problem of res_perl compilation with asterisk 1.4. It is the same problem that was present when asterisk was upgraded to version 1.2. I hope Anthony Minessale will be able to solve that problem as he did on that case. But if any of you know a hack to this problem please let us know. Here is the same compile problem again: gcc -Wall

Hi, I wonder if on Asterisk 1.2.X calls from queue answered by Agent channel still must be transfered only by Asterisk internal transfer (features) like on 1.0.X ? The wiki says on http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Queue "Transfers of calls that are answered out of a queue must be done using Asterisk '#' transfers (enabled with the 't' option

Hi guys, as I wrote in a previous thread I was experiencing dropped audio (apparently randomly) and SIP + IAX peers getting REACHABLE / UNREACHABLE without reason, servers were in the same LAN. Investingating deeply in the problem I also noticed that 'show channels' command on the CLI, sometimes were returning strange results, for example it wasn0t showing some channels I was sure

I (have to) would like to move my agents out of agents.conf in preparation for the deprecation of agentcallback login. Everyone I have spoken to is upset about this but the functionality can be accomplished in the dialplan and that is fine by me. I do have an issue with losing the features contained in the agents.conf though. I have to have the ackcall-yes working. All of my agents login on home

Hi, when I'm trying to call the following extension exten => 6002,1,Verbose(1|Extension 6002) exten => 6002,n,Dial(Agent/6002) exten => 6002,n,Hangup() the call is terminated and I get the following warning from asterisk: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) When calling the agent with Dial(SIP/6002) no problem

This is my nth post regarding this matter. Where in the dialplan should I put AgentMonitorOutgoing? Can somebody show me how to use it? in extensions.conf: exten => x,1,AgentMonitorOutgoing(c) in agents.conf I set updatecdr to yes. It supposed to put agent/agent id in the "channel" column in the CDR, but instead it puts SIP/<extension number>. This link seems to be the

Well because I had sooo may problems with chan_agent.c I wrote this. I'm releasing it under LGPL but if you use it or anything please let me know. It'd be interesting if anyone finds this more useful than just a pile of junk. I've included a (working) example extensions file. SIP phones are assumed to have the same identifier as their extension number, but it'd be trivial to

Hi, Can anybody tell me which version of res_perl I have to install on Asterisk 1.2.4. I tried to compile res_perl version 3.5 on Asterisk 1.2.4 and I got the following error. gcc -Wall -DRES_PERL_BASE="\"/usr/local/res_perl\"" -DMULTIPLICITY - D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -Wdeclaration-after-statement

Dialing an Agent in version 1.0.7 works fine, here's what shows up in the CLI: -- Executing SetCIDName("SIP/518-6071", "Calling Agent4000") in new stack -- Executing Dial("SIP/518-6071", "Agent/4000|30|t") in new stack -- outgoing agentcall, to agent '4000', on 'Local/517@sip-007f,1' -- Called 4000 -- Executing

Having trouble running make on res_perl: [root@charlie res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk \"-DASTVARLIBDIR=\"/var/lib/asterisk\"

Hi all, I have agents who are members of more than one queue. When an agent is busy with queue A, he is not considered busy by queue B, and receives call (since his EyeBeam Softphone has 6 channels). Besides that, I use a monitoring tool that connects through the manager interfaces and run "show queues" and "show agents" to know agents statuses. I need Asterisk to consider

Question for users of Agents and Queue. In our queue.conf file we have entries for both agents and some devices. We do this so that some dual pupose dispatchers can also take a call if they have a spare moment. We went live with a dispatch client over the weekend a number of issues came up the most pressing is that when a call comes to a member of the queue that is a device as opposed to an

Hello, To connect to our oracle database from an asterisk application we use res_perl. Sometimes one of our asterisk server will 'freeze' and work anymore. I have to kill the job safe_asterisk and start it again, so that the application asterisk works again. If I look in the log files it look like that asterisk will 'hang or freeze', if two callers calls exactly at the