search for: orderlysoftwar

...In addition to the restrictions of the LGPL, the following restrictions apply: 1) OrderlyCalls MAY NOT be used to automate 'cold-calling'. Orderly Software takes a strong stand against SPAM. If you wish to use OrderlyCalls to call people without their prior consent, you MUST write to m@orderlysoftware.com explaining why you need to do this. At our discretion we MAY decide to issue permission in specific cases. 2) OrderlyCalls MAY NOT be used to provide or augment call queuing without the prior written permission of Orderly Software. The reason for this is that Orderly Software provides an a...

Hello, I'm posting this to the list in case others run into the same issue. I've recently been connecting * to a legacy Avaya InDEX switch over E1 ISDN PRI here in the UK. Everything was working OK, except that DTMF digits were not being recognised by * when sent by the Avaya switch to the * system. Instead, the background noise of the call centre would be silenced while

Hello, I know there's a problem with Asterisk at the moment in that while it's easy for one caller to dial another (using the dial command), it's tricky to connect two calls that are already in progress. I've been using MeetMe to achieve this (with each caller's call being directed to a dynamically created conference room programatically), and this is working - kind of -

Hello Adam, Matt King <m@orderlysoftware.com> writes: >> I am familiar with the OSI definitiion. I've read it again, but I >> can't work out exactly how asking for permission contravenes this >> definition. > > Then Adam wrote: > "6. No Discrimination Against Fields of Endeavor > > The...

Hello all, >>From your system command line (not asterisk), type 'mpg123' and tell >> us what version of mpg123 you're running. >> >> If its not v0.59r or v0.59q, then get one of those installed. >> (Lots of notes say v0.59r only, however I've been using v0.59q >>. on RHv9 and Fedora 3 boxes with no problems.) Andrew wrote: > FWIW, I have

Hello Jason, I've just come across your post to the Asterisk-Users group regarding OrderlyQ (from a web search - sorry it's taken so long). You wrote: > What experience can be shared about installing and running the OrderlyQ application? I have a bunch of > queues set up and want to start adding some additional apps and this one looked promising but I have > very

Hello all, I've got Asterisk and a TE205P. One port on the TE205P talks over E1 ISDN PRI to the outside world (thorugh BT). The other port talks to an Avaya switch, also over E1 ISDN PRI. All is working well, except that when people try to dial out from the switch through Asterisk (with a TE205P) there are no DTMF tones transmitted to Asterisk. DTMF seems to be fine when the

Hello, So, I've got a bunch of agents logged in using AgentLogin. They're not using VOIP phones, so I only have DTMF tones to play with. A call comes in to the agent, who wants to put the caller on hold (with music) while she talks to her supervisor. When she's finished speaking with her boss, she wants to retrieve the caller from music on hold. How would you do

Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature

Hi there, We've got a problem connecting Asterisk with a TE205P to a TeleWest E1 ISDN line in the UK. We get a lot of "HDLC Bad FCS (8) on Primary D-channel" errors, and every so often the Primary D-channel goes down and all the calls got dropped. We've fully tested the card and made sure it's got its own IRQ. I was just wondering if anyone out there has

Hi Gabriel, Yes this information is shown in real-time and also in historical reports with the OrderlyStats system. OrderlyStats is now available as a Server Edition you can download and install yourself, as well as the FREE managed service. You can get it at http://www.orderlyq.com/statistics.html Hope this helps, Matt. Gabriel Ortiz wrote: Hi all, Is there a way to get the time

Hi there, We're having some complaints of choppy audio from our SIP customers. Asterisk is showing no errors, but I'm getting a lot of these in my syslog: Feb 17 13:34:31 ntop[2863]: **WARNING** packet truncated (14654->8232) The first number varies, but the last number is always 8232. I've read that this is a common MTU size, but none of our interfaces have an MTU of 8232.

Hello, Just to let you know, our popular tutorial on setting up Asterisk for call centres has been updated. The tutorial covers everything from initial Asterisk installation to full call centre configuration with dynamic login and hot-desking support for agents. The old version of the tutorial used Agent Channels (e.g. Agent/1001) to distribute calls to agents through the Queue()