similar to: update_call_counter: Call to peer '2509' rejected due to usage limit of 1?

Does anyone know what can cause queue members to go into a status of "Unknown"? pbxtel-01*CLI> queue show cs has 2 calls (max unlimited) in 'rrmemory' strategy (24s holdtime), W:0, C:447, A:20, SL:91.7% within 60s Members: SIP/1405 (dynamic) (Unknown) has taken no calls yet SIP/1420 (dynamic) (paused) (Not in use) has taken no calls yet SIP/1442

Hi... I just had a real quick and simple question... I have a asterisk implementation setup w/ real time off of a mySQL database for SIP peers and queues, voicemail, agents etc... I after the upgrade to asterisk 1.4.3 there are some new configuration features i would like to use. I was wondering if i could just add to the database table a column for the new config option? if this will work or

Asterisk 11.1.0 One queue with strategy=leastrecent. (Full queues.conf below.) Occasionally (several times today), a caller will get "stuck" in the queue - there are operators available to take the call, but the caller stays in the queue for a long time. Any idea what might cause this, or where I can start looking to debug it? I'm going to start digging through the queue log

When agents are on the phone, and the CLI queue show command shows their status as busy, the queue still tries to send them calls. Running Asterisk 1.6.0.17 and using AddQueueMember to dynamically add agents. ringinuse is set to no for queue. Agents are using Polycom 430s. dialplan: exten => s,n,Queue(orders,itT,,,80) queue definition in queues.conf: [orders] maxlen=20 queue-thankyou=

? 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! 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 am trying to store queues.conf to a MySQL database using dynamic realtime. I have a working ODBC connection and the queueing system already works but I want to store the queues.conf file to a database. I am following the guide from Asterisk the definitive guide, the ebook can be found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html ? I have a database called asterisk

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 |

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

I have configured my mysql database by following this link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Queue The only difference is that I am using ODBC instead of MySQL with Realtime. Within extensions.conf I have the following for my queue exten => 9**2**1611,1,Answer exten => 9**2**1611,2,Queue(irock.com,tT,,,300) exten => *50,1,Answer exten =>

My settings for the queue.log are in the [general] section of logger.conf I'm running 13, I didn't see what version you said you were running. If I wanted to add a LOCAL channel to my queue I'd do it as member => LOCAL/7124 at kiniston-intern,0,John,hint:7124 at kiniston-intern On Thu, Nov 15, 2018 at 2:38 PM Ivan Demkovitch <idemkovitch at yahoo.com> wrote: > John,

Just a silly question that I'm not sure. Ringinuse is working with IAX in 1.6??? like in sip?? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090306/fa56ebfc/attachment.htm

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

We are assigning interfaces directly to our customer service queue through an application running on each agent's PC using the QueueAdd Manager API command. No agents are defined in agents.conf. Does anyone have a solution to pause or remove an interface that doesn't answer after a defined period of time? Thank you, James

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 --

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses

Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.