similar to: Interrupt rates and voip traffic

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, 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 been using qmail-ldap for quite some time and now moved to postfix/dovecot. One feature that I miss is that provided by qmail-secretary. qmail-secretary basically is a mail list manager with following features: 1 no limit, just explode to all members 2 members only, as the name says; only members are allowed (based on envelope sender, so not very secure, everybody can fake

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 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, 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

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, 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, 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 answer to two messages from this thread (I use digest). lartc-request@mailman.ds9a.nl wrote: > > Message: 3 > Date: Fri, 16 Jul 2004 10:51:37 -0700 (PDT) > From: ibro tj <ibb_linux@yahoo.com> > Subject: Re: [LARTC] QoS for Voip. > To: lartc@mailman.ds9a.nl, alessandro.ren@opservices.com.br > > Hi, > > the hint from Martin A Brown which I am

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

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

Hello all, I had gone through the wiki, but could not find any page about writing new plugins. Any pointers towards plugin api/docs will be very helpful. with warm regards, raj

What is the best way to monitor the total incoming / outcoming network traffic of CentOS server. I think that the solution is to monitor the network interfaces and to send SNMP packets to remote server. But is it possible? regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am using collection_select to build a drop down list. I would like to have the current value selected. How can I do that? My current code is straight from the example in Pragmatic Programmers book [2nd ed, p480] copying here for reference. <%= @users = User.find(:all, :order => "name" ) form.collection_select(:name, @users, :id, :name) %> I went through the api docs

Hi, Any one got Oracle client running under wine? I am trying to get a VB application run under wine. I got to the point where it asks for Oracle Client. I tried to run the native windows installer, but failed. Any one with a better way? Possibly adding some dlls and some entries in registry? raj

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 =

Does anyone know of a way to track TCP/IP traffic by TCP/UDP port and/or by process/daemon with CentOS? I know a variety of ways to track it in total (e.g., ifInOctets & ifOutOctets with SNMP), but I'm not sure how to be approach seeing traffic by application, port, or process. One option would be to write something that would sift through different stuff for each daemon (for example,