similar to: Asterisk stops responding

Hi all, can somebody explain this ? Thanks, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing. Thomas H?ger Potsdamer Str. 18 A 14513 Teltow FON: +49 (0) 3328 3077731 FAX: +49 (0) 3328 334779 Email: thomas.haeger@beronet.com *******************************************

I installed the 7/7/03 CVS today, and my customer service reps said that there were problems. So I went back to an earlier version. They could log in, but when they received a call, they hear the beep, but not the announcement or call. I am using macros, if night is on, I dial the 800# that is passed. Otherwise I play the Thank You (ty_pn), and I also pass the announcement that the agent hears

Hi, I am running Redhat, I loaded the mysql and mysql-devel RPM's and then recompiled *.. I thought it would be that simple but it looks like I have missed something becasue it doesn't look like the module has been complied.. What did I leave out? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Hi i have an issue getting agentLogin working /etc/asterisk/queues.conf member => Agent/1001 member => Agent/1002 extension.conf exten => 110,1,Wait,1 exten => 110,2,AgentLogin() now, i call 110 by a firefly client, trying to login in as 1001 agent: Aug 12 16:31:36 DEBUG[1103408048]: chan_sip.c:4423 build_route: build_route: Contact hop: <sip:sip3@192.168.1.151:5060> --

Hi all. I have some question about cdr_csv. We want to have some IVR system for routing incoming calls to multiple directions. So when user make a call, he must put his PIN and phone number. Then asterisk will Dial this call to appropriate destination. Billing system deal with cdr_csv records. But the "Billable seconds" field - all time of the call, including the IVR part (that

Every time I have upgraded via CVS in last few weeks, the queue and agent program doesn't appear to function together. The calls are not getting passed: Agent 308 is logged in and idle. Yet I have a customer that has been on hold for over 8 minutes. Any Ideas? =================================================== pbx*CLI> show agents 301 (Lee) not logged in (musiconhold is

Hi, I have Asterisk 1.4.18 and I have been running call center queues on it. Today it suddenly stopped adding inbound calls to queues. I am facing with following error: app_queue.c:3939 queue_exec: unable to join queue "myqueue" In extension file: Queue(myqueue|t|||120) And my agents are joining in following

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

Could someone please assist me soonest. when i run asterisk with one X101P card and a S400P card asterisk will run but the card will not work.this is the message i get: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [cdr_mysql.so] =>

Does anybody know what this message means? [2009-10-17 08:11:59] DEBUG[9598]: sched.c:204 sched_settime: Request to schedule in the past?!?! I've tried all I can find on google with no change It seems to happen when call is at a READ() or Background() - What the caller hear is small delays after a press during a message. Still using asterisk 1.4.22.1 because it seems to have the best DTMF

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next

Dear List, I have made an update to asterisk RC1 - all works well :) but I am getting all the time an error message: Aug 2 13:34:57 NOTICE[15375]: sched.c:221 sched_settime: Request to schedule in the past?!?! Aug 2 13:34:57 NOTICE[18446]: sched.c:221 sched_settime: Request to schedule in the past?!?! I cannot figure out if this error message is important or not and if there is a real

Hello All, It seems that this question is very stupid, but anyway. Do I need any additional configuration for cdr_csv.so? This module is loaded by default at Asterisk's startup (asterisk -fvvv): [cdr_csv.so] => (Comma Separated Values CDR Backend) But when I place call I didn't see anything in /var/log/asterisk/cdr-csv. There is also no errors or warnings regarding this

Hi, Whenever someone leaves a Voicemail in our system we get this message on the console: NOTICE[18447]: File sched.c, Line 209 (sched_settime): Request to schedule in the past?!?! Does anybody know what it means? Ricardo.

Hi, my Asterisk 1.2.9.1 suddenly freezed ("stop now" did not work!!) . I found the following errors inside /var/log/asterisk/message: Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many open files Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio session: Too many open files Feb 14 14:55:43 WARNING[11273] acl.c: Cannot create socket Feb 14

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Hello, Let say all the SIP devices will be registered on the proxy like kamailio. Agent is a member of Support and Billings Queues on the asterisk servers. Support queue on "Server A" and Billings Queue on "Server B" for example. This will be done via RealTime Queue. I want Agent to dial 1234 on a sip device and it will prompt to enter a pin number to Login via

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without