similar to: callqueues confused :(

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi guys, I'm testing the sending of a URL to an XLite softphone when a call is in queue. See the output of the CLI below: -- Executing Queue("Zap/69-1", "q_sample|tT|http:// www.google.com/") in new stack -- Started music on hold, class 'default', on Zap/69-1 -- outgoing agentcall, to agent '1000', on 'Local/ 1000@agents-1b94,1'

Here is the full "transaction" -- outgoing agentcall, to agent '1001', on 'Local/3044@local-4fee,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-4fee,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-4fee,2", "SIP/3044|20|t") in new stack -- Called 3044 -- SIP/3044-ea92 is

I am getting this on the console once people call in -- outgoing agentcall, to agent '1001', on 'Local/3044@local-fd6d,1' -- Called Agent/1001 -- Executing Macro("Local/3044@local-fd6d,2", "sipline|3044") in new stack -- Executing Dial("Local/3044@local-fd6d,2", "SIP/3044|20|t") in new stack -- Called 3044 --

This just started happening today. I've got 1 queue and 6 agents. All logged in. I tell the service people to ignore my call if they see my caller id. I call the queue and watch as asterisk bounces me around the phones. Our agent ring time is 5 second timeout and a 5 second wait time before trying next agent. I get the same message in console for each agent attempt: -- Executing

Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name "Maximum talktime in a queue?" One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten =>

Hi all, look at these lines. I created a queue named info when a caller (extension 86) place a call he is put on queue he sould hear MOH . What's the meaning of : Jan 29 14:35:30 WARNING[2591]: file.c:509 ast_openstream_full: File 100 does not exist in any format Jan 29 14:35:30 WARNING[2591]: file.c:821 ast_streamfile: Unable to open 100 (format ulaw): No such file or directory Regards

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hi, I have a small call center with 4 Zap lines and 4 agents. Agents login using sip phones with AgentCallbackLogin. I occasionally gets a complaint that when customers call the call center, after the initial greeting is over the call gets cut after playing the thank you message. I started investigating and found that that happens when the call gets transferred to an agent who is making an

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

Hello again, I have noticed with Queues and roundrobin policy that if even if a timeout is set for a queue, Asterisk keeps ringing an available member of the queue after the timeout expires. This continues a few times before the next available agent is tried. I am using CVS of August 17 but I have read in the list that roundrobin worked fine since earlier in August. Does anyone know if this has

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

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

I am using an 8 port tdm card and also I implemented a dialer using a .call file generator. As you know on the .call you specify the channel to call and then the contex/extension/priority to let dial plan continue when the call is bridge. My actual problem is that when the call process starts, asterisk (DAHDI) sets the channel as answered when the truth is that on the other side the channel has

Can anyone shed any light on an issue with agent penalties? I have 2 queues set up with agents working both queues, but where agent 1 should have a penalty for queue 2 and agent 2 should have a penalty for queue 1. When a call is sent to either queue, it rings agents with and without penalties at the same time. I set up a second system and cannot replicate the issue on the test system. I

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 =>

Hi, I'm writing a Java client/server application that talks to the Asterisk manager interface via the asterisk-java stuff. The idea being it will give you an app to run on your desktop that monitors your phone essentially. Once I've got something vaguely working it will be released under the GPL and hopefully people will contribute to it etc... As part of this, I'm currently

Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/510@default-6b6c,1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755]

Hi, I am using * to guide my callers throught my company's support menu. But I have problem when the caller has a "pulse dial" telephony. Could * detect digits dialed on pulse telephones? Daniel

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