similar to: Queues Announce help request.

[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device state of this queue member, SIP/3617001000, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings. [Cary Fitch] We are running 1.4.22 and this message popped up in console. It could be causing our Queues announcement problem, because if all members

Is there a way to program an FXO device to totally ignore incoming calls? I want to put an FXO on a Fax line so that 911 calls can be sent via that line, but all other activity on the line is between the Fax machine and the phone company. Perhaps munge the ring tone detect if nothing else? Cary

I don't want to start the "How many calls can Asterisk handle?" discussion or "How many angels can stand on the point of a pin?" discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from "far away", and process the calls? I am looking for real world, been there, done that, or "check the

I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues.

Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into

I've got the call cuing all setup and working, but im trying to get the Callswaiting,you are caller #, etc, and its not working. I have the following inthere as stated: queue-youarenext = "queue-youarenext" ("You are now first in line.") queue-thereare = "queue-thereare" ("There are") queue-callswaiting = "queue-callswaiting" ("calls

I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=world&v1=l&v2=s&v3=a&pricing=99 Regards, Dean Collins Cognation Inc dean at cognation.net

Hi, I am running a 5 seater inbound call center on 1.0.9-BRIstuffed-0.2.0-RC8h and it's running well. I am now trying to upgrade it to 1.2.4. So I installed 1.2.4 from source and copied all config files from original to the new server. But when a caller lands inside the queue no queue message is getting played. The gsm files are present in proper locations, whcih I am able to play using

I'm looking for the patch that enables suppotr for the following lines in queue.conf: announce-holdtime = yes queue-youarenext = "queue-youarenext" queue-thereare = "queue-thereare" queue-callswaiting = "queue-callswaiting" queue-holdtime = "queue-holdtime" queue-minutes = "queue-minutes" queue-thankyou = "queue-thankyou" It

Hello list, There seem to be some problem with the ACD of asterisk where when we use this parameter in queues.conf . We could not get any announcement as expected. Iam useing the latest CVS-head Even weight also doesnot seem to work properly I tried like this where we have two queues one with 100 weight and another with 200 as weight when both enter into the queue when queue is empty when

Hi, I have a question about the definition of agents. The agents.conf file looks like this: [general] persistentagents=yes [agents] maxlogintries=5 ackcall=no wrapuptime=500 musiconhold => default group = 1 agent => 1311,1311,Tom agent => 1531,1531,Tim and here is the queues.conf: [general] persistentmembers = yes [queue1] musiconhold = default strategy = rrmemory servicelevel = 60

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

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP

Hi list, I'd like to have the callers to listen to the advertisement (music on hold) before the agents answer them. So, I have wrapuptime=10 in queue.conf, but the call still goes straight to the agents without delay. Here's my queue.conf: [general] persistentmembers = yes [738] musiconhold = empty ;musiconhold = default ;announce = q-738 ;strategy = ringall strategy = rrmemory

Hi, I'm trying to use Round Robin Memory with my queues. It seems to work fine... that being I call in.. first time agent 1 will get a call, second time agent 2, and so on. However, it seems if a period of time has gone by since agent 1 got a call (I don't know how much but say 10 minutes or so) when another call comes in.. agent 1 gets the call again. Can anyone confirm this? Is

I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas

Only 100? We had a single server over 300. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve

Hi, I setup a queue (number 4050) with one static agent (extension 4054). What I would like is that when someone calls the 4050 queue and there are neither "dynamic" agents logged in nor is the static agent 4054 "on-line" then the caller gets out of the queue and falls into another context (eg. voicemail or anything). Not "on-line" means that either the SIP