similar to: Outgoing call queues

Anyone thought about something like outgoing queues? I mean, having same info that has for inbound queues but for outbound calls, and grouping members there. For example, before using dial application put an app outqueue that get all the statics. Talked time, member status, last call, completed calls, failed calls, reset statics, and maybe some more. So its possible to get more control and has

Hello I was wondering of it is possible to have Queue Agents with the same priority (penalty) but with a certain order ? So I have 20 Agents. Agent 1 till Agent 10 has penalty 1. Agent 11 till Agent 15 has penalty 2. (only contacted if 1 -> 10 are busy) Agent 16 till Agent 20 has penalty 3. (only contacted if 1 -> 10 and 11 -> 15 are busy) Within the range of Agent 1 till 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

My suggestion would be the one-line eyeBeam phone under development. Check out support.xten.com. //Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r afoc@interconnessioni.it Skickat: den 18 oktober 2005 14:48 Till: Asterisk Users Mailing List - Non-Commercial Discussion ?mne: Re: SV: [Asterisk-Users] Queues

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Hi, This issue has been discussed probably a million times on every asterisk forum in the world and I have the same problem too. Another problem you would have with the agents is that when they make an outgoing call they are not regarded as "busy" by asterisk and it sends more calls to the agent if it has call waiting enabled. This behaviour is totally senseless since the whole purouse

On Apr 17, 4:12 am, Chris Lattner <sa... at nondot.org> wrote: > On Apr 16, 2008, at 11:25 AM, Dan Gohman wrote: > > >> So, my idea is that these changes are performance neutral. > > I strongly agree with Dan that we need to measure performance to > ensure there is no significant performance regression. Dan, Chris, finally I am in possession of hard performance data

Hello, Current moment, I've successfully put the incoming calls into Queues and dial to an idle agents. When the agents answer the calls, the agents can hear the pre-recorded message to incidate what's the service that the call is calling. But there one problem that I'm not able to make it having the Caller ID display on the X-Lite. Even I try to make a call direct transfer from

What kind of help do you need then? Regards, Jan -----Ursprungligt meddelande----- Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Zach A Skickat: den 6 februari 2006 18:31 Till: 'Asterisk Users Mailing List - Non-Commercial Discussion' ?mne: RE: [Asterisk-Users] Help on queues There is no good help on wiki and voip-info.org, I've

On 2015/3/15 16:40, Michael S. Tsirkin wrote: > On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: >> Hi,Michael >> >> I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. > > Why don't you Cc all maintainers on this email? > Pls check the file MAINTAINERS for the

On 2015/3/15 16:40, Michael S. Tsirkin wrote: > On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: >> Hi,Michael >> >> I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. > > Why don't you Cc all maintainers on this email? > Pls check the file MAINTAINERS for the

On Mon, Mar 16, 2015 at 05:24:07PM +0800, Linhaifeng wrote: > > > On 2015/3/15 16:40, Michael S. Tsirkin wrote: > > On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: > >> Hi,Michael > >> > >> I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. > >

On Mon, Mar 16, 2015 at 05:24:07PM +0800, Linhaifeng wrote: > > > On 2015/3/15 16:40, Michael S. Tsirkin wrote: > > On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: > >> Hi,Michael > >> > >> I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. > >

Hello Duncan, > >> what version of LLVM are you using? I don't think LLVM 2.7 uses a > >> "indirect call node". However it does use a "external node", which > >> is used when a function external to the current module is called. > >> > > > > The "external node" makes sense. However, I am using LLVM 2.8. (See >

Campon, mini-queues, see asterisk tips and tricks on voipinfo... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zach A Sent: Monday, February 06, 2006 1:01 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Help on queues I need practical examples showing

On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: > Hi,Michael > > I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. Why don't you Cc all maintainers on this email? Pls check the file MAINTAINERS for the full list. I added Cc for now. > > static netdev_tx_t

On Sun, Mar 15, 2015 at 02:50:27PM +0800, Linhaifeng wrote: > Hi,Michael > > I had tested the start_xmit function by the follow code found that the tx queue's state is stopped and can't send any packets anymore. Why don't you Cc all maintainers on this email? Pls check the file MAINTAINERS for the full list. I added Cc for now. > > static netdev_tx_t

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my

I attempted setting up a queue with agents that log in, and get called with incoming calls: Agents log in using: exten => *88,1,AgentCallbackLogin(${CALLERIDNUM}|${CALLERIDNUM}@test-sip) Calls get into the queue with: exten => 6029995654,1,Queue(test-noc|t|||60) queues.conf: [test-noc] strategy = rrmemory context = test-sip timeout = 10 retry = 4 member => Agent/@2

Over the past 2 weeks I have been able to compile and get an asterisk system up & running on a debian Linux box. I have setup 5 internal sip clients on the lan and all works great! I can also call from outside (PSTN) into the system and reach extensions and services no problem. All is up & running behind a nat firewall with proper ports forwarded and locked down on each device to work