similar to: Play announcement only once in a Call Queue after 10 seconds

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

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

Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...

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

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

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 list, is it normal that when adding new moh-files to the directory /var/lib/asterisk/moh/, asterisk does not see these new files ?! When I do a "moh reload", then Asterisk is aware of the new files... Is there a solution that does not need a "moh reload" ?! Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on

Hello list, what is the correct syntax ? exten => s,n,Queue(${queuename},,,,${timeout},cleanpickup.agi^${CHANNEL}) [Sep 13 10:23:58] WARNING[23551]: res_agi.c:886 launch_script: Failed to execute '/var/lib/asterisk/agi-bin/cleanpickup.agi^SIP/329909007906-0000017a': File does not exist. Kind regards, Jonas. -------------- next part -------------- An HTML attachment was

Hello list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I

Hello list, I have configured extconfig.conf to save queue log into my MySQL-DB. I notice however that there is still logging too in /var/log/asterisk/queue_log. How can I disable queue logging into text files ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' --> 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ?

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Queue not sending call to Agent I am having an issue and i am not sure if it is a bug or a config issue. I was originally running Asterisk 1.8.1.1 when I noticed this issue. I upgraded to 1.8.4.2 to see if that would fix it but it didn't. The issue is that I have a call queue and the agent dials a number to log into the queue. When someone calls the queue the first time the call is

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:

Greetings fellow list members, I have what I think is a relatively simple question, but it did not appear to be addressed on the wiki. I am trying to setup a queue so that it plays an estimated holdtime announcement, but not a queue position announcement. Currently my dialplan does both, and while I know how to take out the estimated holdtime without affecting the queue position

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 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, has anyone experience with auto provisioning IP-phones on different locations through a central public provisioning server ? You use http or https ? Is there a danger that one uses a different MAC-address in the provisioning link to obtain SIP username / password settings ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: