similar to: CDR messed up when using queue

On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327

From the moment when the UPS rapports "Low Battery" to the end of the shutdown, too mutch time is passing. So the computer does not have enough time shut down properly. Can I shut down the computer on the "On Batery" status? Or can I shut down when the UPS rapports 50% of the battery? Thank you.

Hello R users, I found the odfWeave package to create an odf document. It seems to be a very nice tool. So i tried to used it to create a report with multiple plot: I create an odt file with some code inside: I connect to a mysql database I get a list of projects foreach project I would like to make a plot (a map exactly) then in a R console I use the odfweave (inFile, outFile) function.

Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to

Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout |

Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. -------- Original Message -------- Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR

Hello list, I keep on getting the error : ERROR[1707] cdr_addon_mysql.c: Cannot connect to database server 127.0.0.1: (1045) Access denied for user 'asteriskcdr'@'localhost' (using password: YES) I have a 'cdr' table in my MySQL-DB. On this table the user 'asteriskcdr' has select, insert, update privileges. GRANT SELECT , INSERT ,UPDATE ON `Asterisk`.`cdr`

Hi, is it possible to create a Samba share with full Windows ACL support based on a mounted NFS share? What is needed e.g. as NFS mount options? Regards Tim

Hi, I'm using dynamic realtime with asterisk 1.6.0.24, I'm having a strange problem with queue_members... If I update only 'membername' field on queue_members table asterisk won't refresh the change, but if I update another field like interface everything works as expected, I've found this problem also deleting a existing agent on queue_members and then inserting a new

Hi! Did play around with queues and need some help. I thought that MEMBERINTERFACE and MEMBERNAME should be set to the ?device? the call was queued to not the device that called the queue, or do i miss something? Running: Asterisk 1.8.2.3 built by root @ sip on a i686 running Linux on 2011-01-31 13:38:23 UTC 0317998985 calls Kinna (0320209030) Tomas Ekman (SIP/0317998972) receives the call but

Hi, I'm having problems with CDR's and Queues in Asterisk 1.6.1. Heres three examples: Normal call: User A calls in to asterisk, gets a PlayFile, and hangs up. This gives 1 CDR as expected. Call to a Queue and then a playfile afterwards: User A calls into asterisk, goes into a queue, asterisk calls user B. When user B hangs up a CDR for User A is generated. (no CDR for user B)

Hello list, I'm using the following macro when calling an external callphone/GSM number : [macro-press1] exten => s,1,NoOp() exten => s,n,Playback(/var/lib/asterisk/sounds/prompts/press1) exten => s,n,Read(INPUT,,1,1,1) exten => s,n,NoOp(input : ${INPUT}) exten => s,n,GoToIf($["${INPUT}"=="1"]?exit:hangup) exten => s,n(exit),NoOp(call accepted) exten

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

Hello list, google returns a discussion on the dev-list when I search for how to mail a voicemail to multiple mail addresses. Is there yet a seperator that actually works to define multiple mail addresses ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Hello, When I bridge an incoming and outgoing call (attempting to simulate call-forwarding) I'm only getting one CDR -- that of the outgoing call. A (PSTN) calls B (residing on Asterisk) and the Asterisk calls C (cell phone on PSTN) and bridges the call. The only CDR created is from B to C. I have even tried using Answer() and ForkCDR() to get two CDRs, but to no avail. I am starting to

Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel does not receive the QueueMemberAdded Event. This will only be sent if the QueueAdd Function is

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello list, I notice that the dialplan method SIPAddHeader is not working : in dialplan : /exten => s,n,SIPAddHeader(Privacy: id)/ in SIP invite no trace of this header : /INVITE sip:0473 at sip.domain.be SIP/2.0 Via: SIP/2.0/UDP 192.168.1.106:5063;branch=z9hG4bK-5b2b1b97 From: "VC" <sip:voip2 at sip.domain.be>;tag=729476652f511c67o2 To: <sip:0473 at sip.domain.be>

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in