similar to: Call In queue stucks

I have queue set up in realtime on Asterisk 1.4.2. Below is the senario that is happenening :: I have created a test queue with only one agent. Once I call the test queue the agents phone rings if the aagent is logged on. everything till here is fine. Now if the agent does not pick up the call, the call automaticaly disconnects after 15 secs as set for the queue, till here also it is fine. But

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Hi all, I have 2 question. I have a call center queue with 5 agent; the following are the configuration files: *queue.conf* [name_of_queue] musicclass = default announce = queue-name_of_queue strategy = ringall servicelevel = 60 context = callcenter timeout = 60 retry = 5 wrapuptime=15 autopause=no maxlen = 0 announce-frequency = 60 periodic-announce-frequency=30 announce-holdtime = yes

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hi to all, I have a strange behavior in my asterisk server. I have a queue for 5 agents, the calls enter the queue an go to the agents normally, but if I need to transfer or dial directly to an agent extension that is already in a call, the pbx hung up the actual call (not the transferred call). This is what I see in the log. Called 103 -- Agent/103 is ringing --

Hi all! My problem is that calls being placed in the queue, and are waiting while the agents are busy, when an agents is then free they gets connected to the agent but there is silence (no voice). If a caller has not to wait in the queue, there is no problem. My agents have an iax2 client, and imcoming calls are over SIP. queue.conf: persistentmembers=yes autofill=yes ringinuse=no

Hi, I have 1.2.9.1 installed. It always rings first available agents for 15 seconds, then rings and hangs up the next agents straight away, then ring the next agents for 15 seconds. It goes as a loop. Any one has the following same problem? Thanks. Agents.conf [general] persistentagents=yes [agents] autologoff=60 wrapuptime=15000 ackcall=no group=1 agent => 7130,7130,agent1 agent =>

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I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev

I am having problem while having asterisk work with ODBC (Postgres) The error that I am getting is "config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" I really donot know what has went wrong. I have set the ODBC connection properly I have verified it using :: [root@asterisk ~]# echo "select 1 " | isql asterisk

Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'

I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev

Agents have been deprecated and are going to be removed. The replacement, is some complex dialplan using "Local Channels" which the admin will have to dream up for themselves. I'm quite happy to use some new method, but I don't really understand how yet as all the docs I can find point to using agents.... Ideally I need to be able to a> Log into a queue, both by dialing

Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev

Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev

my environment： 192.168.0.13 puppet.uc.local 192.168.0.14 puppetca.uc.local 192.168.0.15 report.uc.local 192.168.0.16 fileserver.uc.local 192.168.0.17 agent01.uc.local i want run a master as fileserver (fileserver.uc.local) the puppet.uc.local and fileserver.uc.local use one ca.pem on puppet.uc.local, i wrote a class for test class test { notify { "hello

Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9

What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay.

Does anyone know why i would get this RINGNOANSWER events in queue_log when clearly the agent is busy and call-waiting is disabled. 1309550595|1309550570.399965|2253|Local/05 at from-internal/n|CONNECT|2|1309550593.399966|0 1309550632|1309550533.399961|2253|Local/11 at from-internal/n|COMPLETECALLER|1|74|1 1309550663|1309550640.399969|2253|NONE|ENTERQUEUE||zzzzzzzzzz

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: