similar to: [ot]Bridging and Cisco switch

Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj

Hi, I have a small callcenter with 3 agents who login using AgentCallbackLogin. They normally receive calls, but needs to call outside also. When they call outside, though they are busy the "show agents" shows them as available, and calls gets routed to them. How can I make them busy when they call outside. Also they also need to move out for couple of minutes or to send a mails

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

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, Is it possible to define maximum talk time in a queue? ie any one who joins a queue should not be able to talk more than say 5 minutes to the agent. raj

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

Hi, I am using RT engine to log queue_log to a mysql database. My extconfig is [settings] queue_log => mysql,asterisk16_production Logging to mysql is working fine. But I find that the queue_log file now only has QUEUESTART lines for eg: 1250519094|NONE|NONE|NONE|QUEUESTART| 1250519186|NONE|NONE|NONE|QUEUESTART| How can I have queue_log in both db as well as in a file? thanks and

Hi! I have the following questions/problems with * 1.4. We have several Snom phones (320 and 360). Hints are configured in extensions.conf (core show hints shows the correct values). My Snom phone is registered to some numbers (validated by using sip show subscriptions). I see the lights blinking if someone calls the subscribed number and steady lights if the call is established. So far, so

Hi, While looking for the cause of disturbance in call I found this error coming in console RTCP SR transmission error, rtcp halted Google search only shows some bug reports relating to MOH and Hold. What could cause this message? Could this be a symptom causing call disturbance? Where should I start digging to find out the reason for this error? I am using Asterisk 1.4.19 with zaptel 1.4.9.2

Hi, I am using asterisk-1.4.15, and using AddQueueMember to add SIP interface to the queue. Each sip interface is member of multiple queues The queue does not recognize that an agent is busy and keeps trying to call the busy agent. I have identified two patches that can fix the problem, one at http://www.scopserv.com/download/asterisk-1.4.17-state_interface.diff in thread

Hi, How can I stress test an asterisk IVR? I am looking for some kind of sip phone which can be "programmed" to send out digits after specified time to simulate users pressing menu items. If it can originate large number of calls simultaneously then it's great! Does any one have any recommendations ? Any other method to stress test an IVR call flow? with regards, raj

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digium card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

Hi, I am trying to log queue_log to odbc (MS SQL) I have res_odbc.conf configured and modules.conf have preload => res_odbc.so preload => res_config_odbc.so extconfig.conf has queue_log => odbc,asterisk. When I start asterisk I get the following messages. The important one being: Realtime mapping for 'queue_log' found to engine 'odbc', but the engine is not available

I was able to get Asterisk working with the demo on FreeBSD 5.3 without crashing, but not the music on hold, so I just have that disabled for now, but I'm ready to get some IP hardware working. So I picked up a Cisco 12sp+ IP phone (mistake?) and am having difficulty finding any truly helpful instructions / troubleshooting to get this configured to work with asterisk. If I could just get

I can't believe nobody uses cisco 7970 with asterisk to help with my issue. 2 sip lines registered: Line 1: ext 260 Line 2: ext 160 How to get Line 2 blinking when Line 2 (ext 160) is called? For some reason with my setup when I call Line 2 - Line 1 is blinking. I use firmware 8.0.3 Anyone has the same problem or is it just me? Please give me some hint. Thanks, Peter

Hi, Any one using Rubi asterisk manager interface http://rubyforge.org/projects/rami/ ? How stable/usable it is? raj

> On Jun 4, 2015, at 5:52 PM, Etienne Dechamps <etienne at edechamps.fr> wrote: > > Are you sure B is correctly configured to forward packets at the layer > 3 level between the interface of the "router" tinc and the interface > of the "switch" tinc? (iptables, etc.) > No, I am not sure about this and I think this is what I don?t understand properly

Hi, I am moving my call center to 1.4. Previously I was recording calls in agents.conf with the following config recordagentcalls=yes recordformat=wav createlink=yes So I had the filename in all calls which was *connected to agents*. I am looking for a similar functionality for 1.4. I am now recording calls using the following configuration. [general] persistentmembers = no eventwhencalled =