search for: callbacklogins

sorry, was only for users list... Hi Kevin, Hi list, you are right, acting now is not needed, when callbacklogin will be removed anywhere in future... But thinking how to realice alternatives can't be so wrong. Callbacklogin gives a very simple way to use more queues for one agent, which only has to logon to only one system. No need to make dbs or tables for saving, where the agent has to be

I'm trying to set a channel variable and make it available to another channel: I thought that if I SetVar(_SomeVariable=SomeValue) or SetVar(__SomeVariable=SomeValue) then SomeVariable would be available in the destination channel. However __SomeVariable, _SomeVariable and SomeVariable are all blank. The scenario: Agents logon to the queue using callbacklogin. From what I can gather

Hi, we are using asterisk 1.2.13. When callbacklogin agent transfer call using SIP phone's transfer feature, he is always in busy status and cannot answer any more incoming call from queue until the transferee hang up the call. -- Regards! Liangliang

I m recording every call, and i want to remove the recorded call at the end of call, when the callee doesn't want the call beeing recorded. Maybe someone can point me in the right direction, having agents with callbacklogin and recording enabled in agents.conf. So if the callee doesn't want the recording, the agents is pressing 0 for deleting the file or 1 for leave the file stored.

Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to

Hi, I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is restarted. Can this be avoided in some way ? Regards, Rob.

Hi When using a call file to place a call I can't seem to figure out how to get the variable alert_info passed to the actual channel (in my case a SIP phone) that an agent is logged in at. Please can someone give me a pointer in the right direction ;) Thanx! Probably best illustrated in an example: Below works great and tells SIP/123 to pick up the call from asterisk then it dials the

Is there anyway to setup a queue with only one agent (device) which is always logged in. So when a call hits that queue the device will ring (if not already on a call) or will be put in the queue if the call is already in place? Thanks Miles -------------- next part -------------- An HTML attachment was scrubbed... URL:

...alstring. "member" means a normal member of a ; queue. An optional penalty may be specified after a comma, such that ; entries with higher penalties are considered last. ; member => Agent/1001 member => Agent/1002 member => Agent/1003 Agents.conf [general] ; ; Define whether callbacklogins should be stored in astdb for ; persistence. Persistent logins will be reloaded after ; Asterisk restarts. ; persistentagents=yes ; ; Agent configuration ; ; [agents] ; ; Define autologoff times if appropriate. This is how long ; the phone has to ring with no answer before the agent is ; automatic...

Egads. Getting queues to work is like pulling teeth. extensions.conf: exten => q_main,1,Queue(oneeighty_main||||1) exten => 80014055,1,Dial(SIP/80014018,15,tr) exten => 80014057,1,Dial(SIP/80014018,15,tr) exten => 80014052,1,Dial(SIP/80014018,15,tr) queues.conf: [oneeighty_main] musiconhold = default joinempty = strict leavewhenempty = strict strategy = rrmemory retry = 0 member