similar to: Asterisk 1.8 -- queue not recognizing that agent is busy

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

I'm trying to move the asterisk server to an Amazon Web instance. We have teliax for our sip provider. I'd like for our DID lines to be connected to a users cell phone. Seems simple enough, but I'm getting the dreaded one-way audio, even with nat=yes everyplace I can think of. The dialplan is real easy: [from-teliax-sip] exten => _j.,1,NoOp("From teliax sip with exten

Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI> queue show 0317998989 0317998989 has 0 calls (max unlimited) in

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

How to get rid on following.. why its Invalid ? holler*CLI> queue show queue1 queue1 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s Members: Agent/7201 (Invalid) has taken no calls yet Agent/7202 (Invalid) has taken no calls yet No Callers -------------- next part -------------- An HTML

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

I am trying to send calls from (telx-NY17S) to (telx-nyc) via an IAX2 channel. However the call is being rejected on the (telx-nyc) server. See error below copied from telx-nyc CLI> Apr 22 13:56:57 NOTICE[147465]: chan_iax2.c:5390 socket_read: Rejected connect attempt from 192.168.0.251 I have icluded the following conf files 1. extensions.conf (telx-nyc) 2. iax.conf (telx-nyc) 3.

I need clarification on couple of issues of Realtime Queue. It seems that when Agents(Memebers) are added using AddQueueMember, Asterisk puts this Queue-Member relationship information into AstDB, So that on asterisk restart this can be preserved. My question is, why does asterisk not store call information for Queue (holdtime, talktime, W, C, A, SL%) in AstDB, So that it can also be retained

Hi! Trying to understand how wrapuptime is working... I have written a small php script that let agents log in/out off a queue. That part is working as a clock but wrapuptime is not doing what I expect. Input Interi?r - Queue Manager 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (5s holdtime, 94s talktime), W:0, C:8, A:1, SL:0.0% within 0s Members: SIP/0317998971

Asterisk: 1.6.2.15 On the production server I've modify the /etc/asterisk/queues.conf file. Now in CLI I wan't to reload queue configuration gracefully. I did: virtual-pbx*CLI> queue reload members office virtual-pbx*CLI> But `queue show office` tells me that nothing has changed. I tried to reload all -- `queue reload all': virtual-pbx*CLI> queue reload all [Dec 28

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

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so

Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten =

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

I am reading at http://www.asteriskguru.com/tutorials/queues.html They are using member in both static and dynamic method. member => <technology>/XXXX -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110519/ec8d19e1/attachment.htm>

Hello, I have queues.conf setup with a group like so: [Sales](StandardQueue) announce = first member => SIP/FF4C119EEBF8-SLS member => SIP/FF9EF375CCFC-SLS member => SIP/13145555555 at callcentric ;Eric's cell member => SIP/FF1565AABB2D-SLS ;Eric's Yealink So, my idea here that it should ring all 4 phones at the same time. And it does work but randomly.I did trace a call and

I have a 100MB internal lan. aastra's are wired. asterisk box is wired next to the switch. But look: sip show peers ........ 142/142 10.10.10.42 D A 5060 OK (137 ms) 144/144 10.10.10.44 D A 5060 OK (136 ms) 145/145 10.10.10.45 D A 5060 OK (168 ms) 150/150 10.10.10.50

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

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,