similar to: Conference with an AGI inside Queue for password change

Hi List; I need to do a call transfer using analoge phone connected to fxs, but I do not need this to be done using flash hook, let it to be using the # or * or any code, but how I can configure that this code is for transfer? Also, I do not need the flast hook to be used for trasfer as it cause usually a confusion to distinguish between the hangup and the call transfer. Any advise? Regards

Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up... I checked rhe SIP debug and it seems that I

Hi How do i set up so that everyone can dial, for example *21* to forward all calls to a cellphone or another extension and how do I enable so that cals can be transferd between extentions. I use asterisk 1.6 and have my phones in unistim.conf and my extensions in extensions.conf. Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113

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 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, 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

Dear all, i have a programming problem that should be simple, though i am stuck with it. Please note that this is not a specific geonames problem, though i use it as an example - it´s just a basic problem with lapply. I use the geonames webservices with the R geonames packages which works almost smoothly. I have a vector of Zipcodes and i want to do a geonames lookup for all of them, which

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 callcenter running with v 1.2 with AgentCallbackLogin and now trying to move to 1.4 using the example doc, doc/queues-with-callback-members.txt. From what I understand the basic idea in the example is to 1. Authenticate a caller with VMAuthenticate 2. Get his SIP Channel number 3. Use

Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during "background(menu-filename)" method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting

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, 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

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

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

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

Hey all, anyone know the status of disconnect supervision on POTS lines in India? Set up an asterisk box, TDM cards, in Mumbai, and doesn't seem to have disconnect supervision...... Thanks -- Chris Earle System Solutions Specialist

hi, can you recommend "clean&simple&stable" solution for incoming call popup (in browser)? i'm using flash operator panel now but i want something without flash (maybe something in AJAX?) thanks --------------------------------------- Marek Cervenka =======================================

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 =

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