similar to: RE: What is the difference between queeu members and queue agents

greetings - I've read a lot on http://www.voip-info.org/wiki-Asterisk, but I cannot understand what the difference between a queue member and queue agent is. Gurus, can you please explain this? When - for example - should I use "AddQueueMember" application and when should I use "AgentLogin" ? Respectfully Constantine

I'm having a problem getting outbound trunking to work using asterisk and an external SIP FXO. 7 digit dialing produces the following output: -- Executing Dial("SIP/mitel-fe17", "SIP/5925660@mediatrix-1204") in new stack -- Called 5925660@mediatrix-1204 -- SIP/mediatrix-1204-645e answered SIP/mitel-fe17 -- Attempting native bridge of SIP/mitel-fe17 and

Ok. I've done more research and testing and here are the details. It is using the dialparties.agi (http://www.sprackett.com/asterisk/) file to dial . Originally with the dial options "tr". I changed the options to "Tt" but no change. Transferring internal extensions between each other works fine. Example: 201 calls 202. 201 transfers 202 to 203. Transferring the IAX

Hi This is hard work :) I have read the Asterisk Handbook, BudgeTone User Manual, Andy Powell's useful notes, Zac Sprackett's Asterisk Resource Pages and more. I am not a linux newbie but am new to Asterisk. I have failed to find any docs that explain how to get a very very simple, minimal, system up and I am trying to get the following to work: 2 BudgePhone 102D connected on a LAN to a

Hi, I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message. My Zaptel.conf is as

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 cfilin@intermedia.net changed: What |Removed |Added ---------------------------------------------------------------------------- CC| |chip@innovates.com -- Configure bugmail: https://bugzilla.netfilter.org/bugzilla/userprefs.cgi?tab=email ------- You are

It would be nice to see a real "extensions.conf" from a live business operation, every extensions.conf I've seen posted or been able to dig up so far would fail bad in a live business operation. I just have the beginings of mine and would like to make sure I don't miss anything. Most extensions.conf files I've seen wouldn't even let you dial "911" in

https://bugzilla.netfilter.org/bugzilla/show_bug.cgi?id=498 Summary: RTP packets are not hitting NAT table Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: Fedora Status: NEW Severity: major Priority: P2 Component: NAT AssignedTo: laforge@netfilter.org ReportedBy:

Paul wrote > >;Pause/unpause Queue >exten => 424,1,PauseQueueMember(|SIP/${CALLERID(num)}) >exten => 424,2,Playback(unavailable) >exten => 424,3,Hangup >exten => 425,1,UnPauseQueueMember(|SIP/${CALLERID(num)}) >exten => 425,2,Playback(available) >exten => 425,3,Hangup > Following your suggestion and a number of postings and articles I have

Im having a trouble understanding the config files setup even with some documentation ive read such as the handbook, maybe im just illiterate. Anyway do you think some one can point me to some examples of real config files. Such as IAX, Extensions, and Sip. I just cant grasp the concept for some reason. If someone would like to help me out, maybe even explain one on one? Thanks a lot

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

I use a single queue for all incoming calls, and different people login at different times to handle the calls, however, quite often, people forget to logout again (incl me). This causes problems because eventually everyone has gone home, and people end up sitting in the queue in-definitely... I use this in extensions.conf exten => 691,1,AddQueueMember(queue,${CHANNEL}) So, how can I have

I'm a newbie, but learning as fast as I can, I have 3 questions so please be patient. 1. I'm not sure as to the correct behavior of using a Agent/@1 in the queues.conf to specify that a group of agents (in this case group 1) should be members of the queue. Currently my queues will only ring the members of the agent group in the order defined in the agents.conf assuming that more than one

Hi, I'm confused with agents and queues in Asterisk. If I use AddQueueMember() then "show queues" shows the agents that I have logged into the queue... however the agent ID has to be the extension the agent is sitting at ... kinda useless for stats tracking. If I use AgentCallbackLogin() then "show queues" shows no agents logged in, but it works and "show

Hi all, I would like to ask for some help about the queue here. I want to implement a call Queue that when there's no agent logged in, they should execute the next extension. eg. if I do it like this exten => 700,1,Answer exten => 700,2,Queue(TestQueue) exten => 700,3,Playback(noagent) exten => 700,4,Hangup When there's no agent present in TestQueue, it should tell the

Would anyone mind sending me a working set of config files for asterisk and their softphone settings? I am really looking for very basic setup stuff as I just want to show that the system works to my management before they will allow me to spend the money on phones and a telco card. Server is a redhat 7.3 w/ 512 RAM and dual 550's. Asterisk tar ball has been laid down, configured, make,

When a caller in the queue is connected to an agent, the call is placed to the extension and context specified using Agentcallbacklogin. This allows for me to add extra things to the diaplan *before* calling the agent. Now, I want to be able to use a device, rather than agents. So I can use addQueueMember and add my SIP device. However, I still want to do a couple of things before the device

Hi John, Try adding username=5702 and username=5703 to each of the configs in sip.conf. I recall I had this problem with the Grandstreams. -----Original Message----- From: John Coll [mailto:john.coll@csoft.co.uk] Sent: Saturday, January 03, 2004 11:56 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)

I am trying to set up Hunt Groups and I am having some issues. Here is what I am trying to do. All my users actually register with OpenSIPS. Asterisk is using Realtime and I have set up a MySQL View Table so that Asterisk see's all the SIP users info that OpenSIPS has. This is what I have configured queues.conf ---------------------------------- [irock.com] strategy=leastrecent

Hi I have a problem with the queues on Asterisk. The setup is Asterisk@Home v1.0 with Asterisk 1.0.7. I have 1 queue (4500) set up, with leastrecent strategy. There are no agents configured in this queue. Agents log in by dialing 4500* on their phones. All incoming calls are sent to the queue. Calls wait 120 seconds in the queue, and are then sent to voicemail extension 310. My problem is