similar to: Blind xfer issue -- URGENT!

[ -netters, please Cc me or security@ with replies. ] I'm running into trouble integrating dynamic racoon-based IPSec into a network with ipfw and natd. I need to be able to allow VPN access from any address from authenticated clients. I've got the dynamic VPN working, with racoon negotiating SAs and installing SPs, but the problem is that I can't tell whether an incoming packet on

My SIP users need to transmit the "#" key as part of data entry. Asterisk intercepts and initates a transfer function. I'm almost positive I've seen this discussed somewhere, but none of my searches are finding it. Anyone have a handy answer? Tim McKee -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef

I've been unable to figure out how to make blind xfer transfer a call on a zap interface by using hook flash instead of a native bridge. Is this even possible to change how a blind xfer works on the 7940s? zapata.conf: threewaycalling=yes cancallforward=yes transfer=yes

I was wanting to automate entirely a blind transfer. We are not yet using a powerdialler, so when we hit an answermachine we have to manually leave a message. In order to make this a little quicker, I want to leave a standard message on the answermachine. attempt #1. Use the blind transfer feature. set blind transfer to be "**". extension 22 is exten 22 => Goto

Hi all I am having problems with atxfer if I do the extact same thing with blind xfer it works fine when I hit press #2 (defined in conf for atxfer) i get "transfer" I dial the number I want and i get the following on the console -- Playing 'pbx-transfer' (language 'en') -- Executing Dial("Local/18005558355@jesnjer-f97a,2", "/18005558355")

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I have a problem where my recorded queue calls stop recording once the call is transferred to a different extension. Is there some additional parameter I need to set so recording continues? Is it even possible to do this? Thanks, Jay

Here's how it's currently working: 1) Call comes in 2) Operator parks call (700) 3) Operator picks up call on another phone (701) 4) Operator tries to transfer to a different phone (we use #0) but the transfer doesn't work. We can transfer initial callers all we want and it works fine. Once a call is parked, however, we can no longer transfer the caller. Any ideas? Thanks, Jay

Hi, I'm trying to use Round Robin Memory with my queues. It seems to work fine... that being I call in.. first time agent 1 will get a call, second time agent 2, and so on. However, it seems if a period of time has gone by since agent 1 got a call (I don't know how much but say 10 minutes or so) when another call comes in.. agent 1 gets the call again. Can anyone confirm this? Is

Egads. Getting queues to work is like pulling teeth. extensions.conf: exten => q_main,1,Queue(oneeighty_main||||1) exten => 80014055,1,Dial(SIP/80014018,15,tr) exten => 80014057,1,Dial(SIP/80014018,15,tr) exten => 80014052,1,Dial(SIP/80014018,15,tr) queues.conf: [oneeighty_main] musiconhold = default joinempty = strict leavewhenempty = strict strategy = rrmemory retry = 0 member

? 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

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi all, I am having the following trouble with recording calls: When calls come into the support line did number, the call starts to record on the first queue, but appears to hang up when the call actually connects to the engineer (ie I see "got hangup request" on the cli and then mixmonitor ends.) I am guessing this has to do with the announce file that is played to the engineer

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

Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part..

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

I have a queue setup to 'roundrobin' (NOT roundrobin with memory). I have three agents. We'll call them 101, 102, and 103. When a call comes in.. I want it to always try 101 if no answer try 102.. if no answer try 103, etc. However, what it is doing is... it will ring 101... if 101 answers, next time a call comes in it will go to 102. This isn't at all what I want. Any ideas

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

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 --

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