similar to: asterisk (FreePBX) and queues

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

Dear All, I have make a queue in my dailplan and queue is not working properly,prbolem is that all call goes to same extenstion at a time.Because,I use eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into eyeBeam that call reserved by Line 1 suppose to 2nd call will come that call goes to Line 2(same extension used by Line 1) and 3rd call goes to 3rd line and so on. But i

Hello, >From some days ago, when i made changes in web interface to asterisk that comes with trixbox (freepbx), this dont reflect the changes in asterisk configuration. I has reviewed the file permissions in /etc/asterisk and all files are writable to asterisk user. In freepbx all appears to be ok (i dont see any errors...). Anyone can help me with this problem? Thanks in advance, PS.

I posted this over in asterisk-dev, realized I probably should have put it here. Hi there, We?ve been having a strange issue with a customer?s queues where a queued call will ring an available agent, agent answers, then a second or two later the agent is offered a second call which they cannot answer, since they?re already speaking with a client. This in turn causes a few issues: - Agent stats

Hi All, I am trying to install Asterisk with FreePBX while running install_amp following error is coming can any one help in this regards Thanks in advance.. Linga Reddy Connecting to database..OK Connecting to Asterisk manager interface..OK DB Error: no such tableGenerating AMP configs..OK Restarting Flash Operator Panel..OK

Hi every body, I've set up a Trixbox Server with TE110P,all things seem to work fine(Thank You Malling lists & irc & Forums), but i need your help, i ve 30 numbre from 60 to 89, i need to specify for each sip extension a Zap number for example to call the sales service the caller must call 555-4570 and automaticly the caller will be redirected to the 202 ( sales service ) so nobody

Just woundering if the intentend functionality of leastrecent and fewestcalls it to continually dial only the first chosen ext. in the queue. In other words if a memeber is logged into the queue but doesn't answer the call the call never moves on in my configuration from that ext. This could be really bad!!!! Thanks [support] announce-frequency=45 strategy=leastrecent music=default

I noticed a few issues with app_queue just wanted to know if its sip related or ata186 related: Ext 111 and Ext 112 are dynamically loged into the queue via AddQueueMember. Call hits queue with fewestcalls routing. Rings ext 111 if 111 doesn't answer. It rings ext 112. If for some reason ext 112 doesn't answer it rings back to 111. Again at this point ext 111 isn't answered it

Dear all, I can't understand the diff between roundrobin and rrmemory strategy. Could you explain for me ? and is roundrobin means each available interface ring once or several times and ring another? ; A strategy may be specified. Valid strategies include: ; ; ringall - ring all available channels until one answers (default) ; roundrobin - take turns ringing each available interface ;

Hello, I use samba-3.0.14a_1,1 on FreeBSD 5.4-STABLE. Everything is running perfectly. Today, the profile of my users are stored on the file server. For example: John logon on the Windows 2000 Professional workstation called machine01. It's the first logon of John on this machine, but him already have a profile stored on the file server. When John logon to machine01, the profile of John are

Hi, I'm running samba-3.0.14a_1,1 under FreeBSD 5.4-STABLE. I migrated all my data from samba 2 to samba 3, and I'm having a problem with some users. For example, user john can login perfectly to the network. It?s ok. But user smith not, him get the error ?The network name cannot be found?. But all configurations of john and smith are the same. I found this on Samba Official Hanbook:

Dear All, I've got a working asterisk installation which I need minor help from. Currently, I'm running a Sales Queue, which is answered by a selected group of people. Here are my queues.conf [sales-hotline] strategy = roundrobin timeout = 10 member = SIP/602 member = SIP/603 member = SIP/701 member = SIP/604 After calls come in, it works fine, however, I notice that even when SIP/602

Hello, i hope u guys can help me. This is the first time I write to the list. Sorry about my english... i got a solaris 10 machine and installed "samba 3.0.2.3c" with "openldap 2.3.2.1" , "openssl 0.9.8" and "gcc 3.4.6". i configured kerberos and all the other things. all good. i added the samba-server (solaris10) to a active directory domain. with

Hi, I have the same problem with the queue configuration When I receive 2 calls only 1 phone ring even if more agent's phone are free. The second call will go to an other agent only if the first call is pickup. Somebody have a solution ? This is my config file : Queue.conf [general] ; ; Global settings for call queues ; ; Persistent Members ; Store each dynamic agent in each queue

Ive installed asterisk and freepbx. Through the interface ive configured 2 extensions, 6000 and 6001. My problem is that when i try to call from extension 6000 to 6001, i hear this msg "Im-sorry&an-error-has-occured" and the call is terminated. As expected if i call to another number i get an error. i thought the problem might been related with the NAT but if checked and changed some

Hey all, I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen: http://www.sineapps.com/news.php?rssid=722 The script, when run as the asterisk user, works properly and streams sound to stdin. But when Asterisk starts MoH it stops it immediately afterwards with no explanation. Has anyone

hello to all, I advice you to not use CDRtool from ag-projects : Fisrt ag-projects talk about is product like a gpl software however they don't provide at least some documentation for non commercial users . try to call them !! i'll offer you some money . You can not Call them for some advices ... It's really a bad product don't waste your time to setup it. this enterprise must

Can a temporary message in Asterisk voicemail be de-activated so that the "regular" unavailable and busy messages are played. I have several users who are stuck with the temporary message. Thanks Mark

Hi List I need a quality US 800 DID over IAX for my Asterisk server, preferably one that doesn't cost the earth. Any suggestions please? Thanks -- Chris Blunt Entropy IT Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061220/4919f3cb/attachment.htm

Every few seconds I get the following message: == Parsing '/etc/asterisk/manager.conf': Found == Connect attempt from '127.0.0.1' unable to authenticate I'm trying to track down where it's coming from. I've used TCPDUMP & NGREP to monitor 127.0.0.1, no data's flowing. I've tried loading Asterisk with no modules, tried loading with a naked