similar to: QueueMemberStatus/Status Field

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Hi,all, Now,I am working at make an realtime monitor for the call center based on asterisk. and ,I had search the archive and wiki.Through the return info from the management API, I can get the waiting calls ,abandoned calls ,hold time, etc,but I don't know how to get the number of incoming calls. The info like following :

I've been poking for the definition of QueueMemberStatus and all the source file indicates is that it is a integer member of the "member" structure. Anyone know where I can find the CONSTANTS definitions? -- Warm Regards, Lee

I've tried every combination I could find on the net and so far there is no joy The thing is I can do this update from the command line: Maybe some new eyes might find the answer? exten => update,1,MYSQL(Connect connid localhost root password dax) exten => update,n,MYSQL(QUERY resultid ${connid} UPDATE\ caller\ SET\

Hi all, What format is the LastCall variable of QueueMember event? I'm looking at: 1201897536 for instance. Thanks ! -- Warm Regards, Lee "Everything I needed to learn in life, I learned selling encyclopedias door to door."

I have a queue I configured as "strict" and a cron script I use to QueueAdd and QueueRemove agents according to my company's requirements. Usually I have 2 or 3 agents at a time and the ring strategy is ringall. These agents use non-open-source Windows softphones that do not let you configure it so that if they're on the phone, a second call will be rejected (agent busy).

Hi all, After * starts the command "queue show" would not show any of the realtime queues, but just the ones that are in the queues.conf file. In this state de AMI would not send any "QueueMemberStatus" for that queues until a call is received by that realtime queue. Anyone knows any whay to load this information in *'s memory without the need of the queue receiving a

Hi All Before I go out and buy a DELL PowerEdge 2850 has anyone had problems (or any other useful experience) getting a TE411P to work with it. I also have a legacy TE110P. Has anyone had problems with this combo. -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The

Hello list ! I have some configuration issues with queues, but I'm sure they are minor and for someone who has already configured queues it could be trivial. This is my queue configuration : [VC_support_queue] musicclass = default strategy = ringall timeout = 20 retry = 5 wrapuptime=15 autofill=yes autopause=no maxlen = 0 setinterfacevar=yes announce-frequency = 0

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hello list! I?m working on a fresh Asterisk install over CentOS7 base. I?m using ?Asterisk. The Definite guide? book as a reference. I connect and work using SSH Problem I have - I can?t connect to asterisk from remote. Getting error: $ sudo asterisk -rvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) Yes, it exist, and service runs: [asteriskpbx at

It would be nice if Digium updated the hardware compatibility list on asteriskpbx.org to indicate that the LineJACK can't be used for dialing out. I've seen several people on IRC be burned by not knowing this. --Eric -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Dear All i'm planning to develop for a customer a particular implementation of Asterisk. The aim of the project is to share different users between different Asterisk inbound call center . I'm planning to have a sync for some of the QueueMemberStatus informations between all the nodes, then a particular (external) ACD algorithm will decide to transfer a parked call to the final user. I

Hello folks. I've got a strange issue. When I modprobe TE220 I do not see mesages like Launching card: 0 <..> Setting up global serial parameters. You can see how I loaded and unloaded the card for several times - http://asteriskpbx.ru/pastebin/11 lspci can detect the card: 03:08.0 Communication controller: Digium, Inc. Device 0220 (rev 02) dahdi_hardware also: astpbx ~ # dahdi_hardware

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

Hello list, one of our clients is going to be deploying a system with over 200 differently composed queues and 100 agents. We are going to do a full test of the viability of this solution before deployment, but I was wondering if anyone has experience of such a setup and if there are any obvious problems or no-nos. Any suggestion welcomed, l. -- Home of QueueMetrics -

Hi Folks, Does anybody have experiences with plugging 3 TDM400P cards in one PC?? I think about a Asterisk box handling 8 incoming analogue lines and providing 4 lines to an old analogue PBX. I read a lot about trouble with the TDM400P cards so this idea seams to be not really god, or? Ciao Joerg -- _____________________ Don't PANIC

We just recently upgraded from Asterisk 1.2 to 1.4, and quickly noticed a change in the behaviour of the queues--a change that we cannot live with. We've used AddQueueMember/RemoveQueueMember to manage logging into and out of our queues for over a year now with Asterisk 1.2, and in that version the queue members were sorted in such a way that the person who had been logged in the longest

HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls