similar to: Exiting the queue doesn't work

Our phone operators work off of an Asterisk queue. They take calls from customers and take orders with our back end systems. What I need to be able to do is tie the orders taken to the specific CDR record that reflects the call from which the order originated. The typical/sample CDR table doesn't have a primary key. I can add an auto-generated PK, but the CDR is not written until the

Hiii all, I am using asterisk 1.8.9.2 and compile all modules related to calendar. neon version is 0.29.6. OS is ubuntu 11.10. I configured ical for zimbra, caldav for google mail and ews for exchange 2010 calendar. ical and caldav setup working fine and i am getting my calendar events perfectly. But for exchange 2010 calendar i am getting following error. "Unable to communicate with

Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 0000000000533c19 sp 00007f7db9ce3af0 error 4 in asterisk[400000+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528

Asterisk 11.1.0 Various soft-phone SIP clients call center with 10-12 agents online at once using asterisk queue Occasionally an agent will get a call (or more often a series of calls in a row) where neither party can hear the other, or can only hear each other sporadically. A MixMonitor recording of the call plays only the caller - none of the agent's audio is heard in the recording.

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it

Hi, Is there a way in which rsync's -z compression (zlib) utility can be benchmarked? I'm trying to compare the compression ratio between rsync and external compression tools like gzip and bzip2. Are there any advantages to using rsync's internal compression mechanism specified with the -z option compared to solely applying external compression i.e. bzip2 to the files and invoking

Hi, while implementing the rsync protocol in one of our projects I found that the current CVS version still has a MD4 bug. I'm using the FreeBSD libmd implementation and I still had checksum mismatches with protocol version 27 for files whose size was a multiple of 64 - 4 ( - 4 due to checksum_seed). A patch for todays CVS version is attached. Someone should also review the clean_fname()

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

On Thu, 2004-04-08 at 12:36, Martin Pool wrote: > On 5 Apr 2004, Donovan Baarda <abo@minkirri.apana.org.au> wrote: > > > librsync needs a whole file checksum. Without it, it silently fails for > > case 1), 3), and 4). > > Yes, a whole-file checksum should be used with it. Presumably > something stronger than md4 like SHA-1. md4 is probably good enough for most

Hi all, I need to configure DIDs for different companies and they should reach on different extension with different context. Cant we have same extension in different context? This is what we we want Company A: Context Company_A IVR Company A Extensions: 101,102,103,104 etc. Company B: Context Company_B IVR Company B Extensions: 101,102,103,104 etc. Company C: Context Company_C IVR Company

Hi List, I have faced a problem in asterisk queue implementation. I configured a queue with 'ringall' strategy and 'ringinuse=yes' in queues.conf. If three calls come to this queue in parallel, the logged in queue agent used to get only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed. I

Hi All I have been looking at this problem for a few days/weeks now and after some advice please. I currently have a customer on 11.25.3 and I am in the process of upgrading versions and OS (Debian) and all things that involves mysql -> PDO etc The problem I have is the customer want a simple call distribution like this Extn 1001, 1002, 1003 to be called on an incoming call - if they

Hello, I was planning to use the output of asterisk -rx "show queues" in a script when I noticed that sometimes asterisk only outputs the first line of the response. e.g: debian:/# asterisk -rx "zap show channels" Chan Extension Context Language MusicOnHold debian:/# asterisk -rx "zap show channels" Chan Extension Context Language