[asterisk-users] OrderlyStats SE 1.6.2l now available.
Hello, Just thought you might like to know that our popular call center management and statistics package, OrderlyStats SE, has just got a new release. Version 1.6.2l includes a several configuration changes and enhancements to provide seamless call integration with the popular Elastix distribution of Asterisk. To download your free evaluation version, and start benefiting from improved visibility and easy call center management today, just visit http://www.orderlyq.com/asteriskcallcenterstatistics.html Thanks for reading. Kind regards, Matt King Managing Director Orderly Software Ltd. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Queues Tutorial updated - Hot-Desking without Agent Channels
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() application. These have been deprecated since Asterisk 1.4, and the AgentCallbackLogin function is not supported as of Asterisk 1.6 The new version of our tutorial shows you how to achieve the same functionality (and more) with Local Channels, which are the recommended replacement for Agent Channels. So, if you are setting up Asterisk for call centre use, or if you are currently using Agent Channels and want to maintain your upgrade path, please take a look at http://www.orderlyq.com/asteriskqueues.html and find out how it's done. Thanks for reading. Matt King, CEO Orderly Software. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Packet Truncated - Choppy Audio
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. Could it be that Asterisk is chopping the packets? Has anyone seen this before? Any assistance would be most gratefully received. Regards, Matt King Managing Director Orderly Software Ltd. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Packet Truncated - Choppy Audio
Hello Danny, Thank you for the swift reply! As it turns out, this was an artifact from ntop, which has a default maximum buffer size of 8232 bytes. We're still getting choppy audio, but we've ruled this error message out as a possible cause. Thanks again, Matt. From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] Packet Truncated - Choppy Audio This indicates that your NIC card is not handling the throughput effectively. Is * the only application on your server? How many users are on * when this occurs? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue time to answer/abandon + OrderlyStats Server Edition.
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 that a specific queued call took to be answered or abandoned? Thanks, Gabriel Ortiz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question about queue
Two use-cases where autofill=no is desirable: 1) If it's important that you answer your callers in strict order (i.e. in order to meet estimated wait time commitments etc). 2) If your queue members/agents are local channels (as local channels are always available, so call attempts will be made regardless of who's talking). Kind regards, Matt. BJ wrote / This was something I put in a long while back on 1.2 branch because we really needed it for 1.2 to bug fix the behavior, but also needed to prevent the change in behavior for those that didn't want it to change. // // That being the case and we're in the day and age of 1.6 branches now, it'd be interesting to think of what people would think of deprecating this option completely now in /trunk in favor of the autofill=yes behavior being the only behavior available. I cannot think of any use cases where the autofill=no behavior might be desirable. That being said, I also might have blinders on so would be curious to here what the rest of the community has to say about it. // // BJ/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue logging
Hello Arjan, You can see who is in the queue, which agent is ringing, whether the agent is Paused, and which agent is connected to which caller, using OrderlyStats (FREE sign up at http://www.orderlyq.com/orderlystats.html ). This is shown in Real Time, and also in the call history logs. This will also show you whether the connection succeeded or failed as requested. Kind regards, Matt -- OrderlyQ Support. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with TE205P with TeleWest in the UK
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 had similar problems with this provider, and if so, how they managed to solve them... Thanks for reading, Matt. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Stats
Hello Jay, Sounds like quite a complicated set up. Most queue statistics packages will break your callers down depending on which queue they were actually answered in (or hung up on). If you want your stats listed as if the callers were in a single queue, you can sign up for a FREE OrderlyStats account at http://www.orderlyq.com/orderlystats.html - once you're all done, let us know and we'll show you how OrderlyStats can show these calls as if your three queues were just one. Hope this helps, Matt. Jay wrote: Greetings, list! My boss would like some statistics on how many calls are answered out of specific queues during a given time period, and I'm not sure how exactly to obtain those stats. Here's how our queue system works. 1) Call comes in and enters our 'ring' queue where the phones ring for 15 seconds (caller hears the standard ring tone). 2) After 15 seconds, the caller falls into our 'music on hold' queue, a message is played and the caller hears our music on hold while the phones are rung again. 3) After 30 seconds, if the caller is still in our 'moh' queue, they drop out of queue and immediately re-enter the 'moh' queue again until the call is answered or the caller hangs up. How can I find out how many calls are answered out of each queue during certain times (1st shift, 2nd shift, etc...)? Also, I'm curious how I can track how many times a call repeats the 'moh' queue. Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues monitoring software - OrderlyStats now FREE
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 allows you to reassign agents and penalties on the fly. Our Agent Bar tool also helps agents log in and out of their queues, and enter Pause for wrap-up at the touch of a button, in addition to displaying realtime wallboard-style queue information. Furthermore, we automatically produce a wealth of historical statistics, allowing you to track caller trends and analyse staffing requirements across your queues. You can sign up now at http://www.orderlyq.com/freestats.html . It would certainly be worth your while to take a look, especially if you're considering spending money on QueueMetrics. Thanks for reading, Matt. [EMAIL PROTECTED] wrote Hello all, A client of us, needs a queue monitoring system. In realtime he needs to now the PRI status, the agents logged in and logged out, the number of received calls by agent, ,etc. I am not a call center specialist and i want to find a call center software to offer to my client that fits his needs. I need a monitoring solution for incomming and outgoing calls and a queue management interface to create and/or modify queues or agents. Any one of you could has instalesd this kind of software? Which one? Which one could you recomend me? Thanks in advance. Voipcrazy. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting Asterisk to Avaya Definity over H.323
Hello, Does anyone out there have experience or settings they can share to help connect Asterisk to an Avaya Definity system over H.323? If so we need your help! Please email me directly. Many thanks, Matt King Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SJPhone and Asterisk over H323
Hello all, I'm using Asterisk h323 (default/NuPhone) with some success with SJPhone. I say some success because while I'm able to receive audio from Asterisk, I seem unable to send audio to it... Any suggestions? Anybody managed to get this to work? Thanks, Matt King Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] !! Got a UA, but i'm in state 1
Does anybody know what these are? Started getting them last night when I upgraded from 1.2.6 (Zaptel 1.2.6) to 1.2.10 (Zaptel 1.2.7). Then my E1 ISDN PRIs go down... I've had to roll back to 1.2.6 :-( Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird transcoding error (SIP, local channels): sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256)
Hello, I wonder if anyone's hit these before. I'm trying to bridge two callers using parked call. It works fine when I'm using two sip channels directly (one hits a ParkAndAnnounce, the other hits ParkedCall), however my application requires an intermediate local channel. The flow is 1) Manager originates a dial out to the target Sip extension with a local channel at the other end. 2) When the sip extension answers, the local channel is parked with ParkAndAnnounce. 3) The second (source) sip channel is then directed to ParkedCall with the relevant number. This *should* connect the two callers (and seems to be the best way of bridging two already-connected calls) BUT if the two SIP phones are using different codecs, then I get a whole load of messages like this: Jul 5 20:19:46 WARNING[2917]: chan_sip.c:2561 sip_write: Asked to transmit frame type 64, while native formats is 256 (read/write = 64/256) We end up with the target SIP channel able to hear the source caller, but not vice versa. I'm guessing that the local channels use frame type 64 (SLinear), and that for some reason this is causing the barf. We're using 1.2.9.1 Any suggestions very gratefully received. Many thanks, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: DTMF Talk off
With recent versions of *, you can increase the detection time in zapata.conf with the toneduration variable. The default setting is: toneduration=100 We had the same problem and solved it by increasing this to 200. You can also increase the threshold volume for detection of DTMF by setting VPM_DEFAULT_DTMFTHRESHOLD in the relevant zaptel wctX.c and recompiling (though if you increase this too much you risk losing your ability to detect DTMF at all). Hope this helps, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bridging two existing calls (MeetMe, Sip Reinvite)
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 - but this results in a conference instead of a bridged call, so - we can't use the normal Dial parameters for transfer etc, - the other caller is not disconnected automatically when one party hangs up, and - (most importantly) we can't get SIP to reinvite. The SIP reinvite issue results in increased bandwidth costs, extra latency/echo and reduced call quality when compared with Dial (as the media path has to include Asterisk with MeetMe, but not with Dial). Does anybody know of any other way to bridge two existing calls with Asterisk, that will allow SIP to reinvite? I've already asked on the IRC channel, searched the list archives and had a look through the bug tracker. I'm cross-posting this to the dev list too as this my last resort before making a feature request/bug post... Hope this helps, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: FastAGI Connection Failure and Hangup
Steve, you need the FastAGI contingency patch, part of the Asterisk Queues Tutorial available at http://www.orderlyq.com/asteriskqueues.html It's near the bottom of the page. Anybody know why this still hasn't made it into trunk? Matt. Steve wrote: Does anyone know how to make fastagi continue to the next priority if it can not connect to the remote AGI Server? Right now I am just getting Hangup and can't find anything on the net about this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Missing 31 DTMF tones over ZAP
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 users hit the keys on their phones - echo tests and RecordFile produced a flatline response. I had at first thought that the Avaya switch may not be sending them, however this was working when * was not in the call path. With further testing, I've found out that it is in fact only the first 31 DTMF tones that are missing - those following are picked up OK. I've got no idea why this should happen, and have kludged a fix by having the Avaya switch send out 31 'fake' tones before the user starts entering data (using Translation inside Route List). If anyone has come across this before and knows of a 'proper' fix, or even what might be causing the issue, I'd be very grateful for the information. Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Missing 31 DTMF tones over ZAP
We're using a TE205P. lsmod indicates that it's using the wct4xxp driver. Hope this helps; I'll give it a try with disabled vpmdtmf. Matt. C F Wrote: what zap device are you using? IIRC disalbing the vpmdtmf on a 406 or 411 might help you. I think it's done in wctxx4p.c On 2/24/06, Matt King [EMAIL PROTECTED] wrote: 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 users hit the keys on their phones - echo tests and RecordFile produced a flatline response. I had at first thought that the Avaya switch may not be sending them, however this was working when * was not in the call path. With further testing, I've found out that it is in fact only the first 31 DTMF tones that are missing - those following are picked up OK. I've got no idea why this should happen, and have kludged a fix by having the Avaya switch send out 31 'fake' tones before the user starts entering data (using Translation inside Route List). If anyone has come across this before and knows of a 'proper' fix, or even what might be causing the issue, I'd be very grateful for the information. Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queues and On Hold
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 this? I've tried it with call parking (by pressing # then 700), but then the system passes through the next caller to the agent immediately, and it's impossible to retrieve the caller without hanging up, then dialling the parked extension. Any ideas? Thanks, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk + Avaya DTMF problem
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 Avaya switch is connected directly to the PRI, however. Any ideas? The lack of tones being passed to * means that people can't dial out at the moment... Many thanks, Matt. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Call simulators
Hello Rob, Our OrderlyQ system is designed to pass (real) calls to call centre agents and queues at a constant rate (or at least can easily be configured to do this). I can think of several ways the system could be 'rigged' to produce the calls automatically too... We've also built our own call centre simulators as part of the development effort for OrderlyQ. Let me know if we can help, Matt King, M.A. Oxon. http://www.orderlyq.com - the world's most advanced queue system. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Queues Tutorial updated...
Hello, Just a note to say the Asterisk Queues Tutorial at http://www.orderlyq.com/asteriskqueues.html has been updated to take account of changes in the 1.2.0 release. Anybody who has used our tutorial to create their queues, or uses queues and is thinking of upgrading, will probably find this new version useful. Comments feedback welcome - though message me privately please to avoid bugging the list Many thanks, Matt King Managing Director, Orderly Software Ltd. http://www.orderlyq.com - the world's most advanced queue system. P.S. You can also check out our new statistics package, OrderlyStats, at http://www.orderlyq.com/statistics.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel Outbound Caller ID on E1 in UK
Hello, We've got a shiny new TE411P hooked up to four E1s at a carrier (not BT) here in the UK. Everything's working great, except that we can't seem to work out how to set caller ID on outgoing calls. When I dial out to my mobile, the caller id is shown as the pilot number of the range we've got assigned to *, whatever we seem to do. We really need to be able to set this arbitrarily, and/or to the caller ID of incoming calls when bridging. Any help would be greatly appreciated. Here's our conf so far... zaptel.conf: span=1,1,0,ccs,hdb3,crc4,yellow span=2,2,0,ccs,hdb3,crc4,yellow span=3,3,0,ccs,hdb3,crc4,yellow span=4,4,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 bchan=32-46,48-62 dchan=47 bchan=63-77,79-93 dchan=78 bchan=94-108,110-124 dchan=109 loadzone = uk defaultzone=uk zapata.conf: [channels] context=orderlyq switchtype=euroisdn pridialplan=local prilocaldialplan=local signalling=pri_cpe rxwink=300 usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no callwaiting=yes usecallingpres=yes sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callprogress=no callerid=asreceived channel = 1-15,17-31 extensions.conf: exten = 1000,1,Dial(ZAP/g1/07743898503|20) Thanks in advance... Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. 01392 421 078 or 0774 38 98 503 http://www.orderlyq.com - the world's most advanced queue system. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zaptel Outbound Caller ID on E1 in UK
Got it - the carrier's hardware will only pass on caller ID within the DID range assigned to the E1 PRI. Matt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Cluster
Hello, I'm going to need to take up to 10,000 simultaneous calls on a single number. I'm going to need lots of * boxes to do it. How many * boxes will I need, and how do I load balance the calls between them? Matt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk Cluster
Hello, OK some more information. We've got a client who takes very large call spike when tickets go on sale. We're specing to handle 10,000 simultaneous calls. We're hoping to use our VOIP provider to handle incoming calls from landlines, and to route the calls out to our client's call centre. Our * boxes sit in the middle, coloed with the VOIP provider. We can use SIP or IAX with the provider. It's been suggested that SIP is the better choice, as it should allow the provider to route call streams directly through to the call centre once they've been through our system (more details of what we're doing at http://www.orderlyq.com). Someone has suggested SER as a good way of doing the load balancing ( http://www.voip-info.org/tiki-index.php?page=SIP+Express+Router ), but I haven't used it before - anybody know how to get it to load balance? Also, if we got 20,000 calls instead of the anticipated 10,000 calls, how would we drop the extra 10,000? We don't want to use up more of our provider's lines than we can handle calls. Please feel free to respond privately if you like. Many thanks, Matt. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which AGI Development Software is fastest on Asterisk?
Hello, I'm looking to develop some custom AGI that will be MySQL intensive. It appears Asterisk supports many different development environments. Which would be best suited for Asterisk and MySQL? It's generally fastest to use FastAGI (over TCP/IP), rather than regular AGI as this means the OS isn't starting a new process for each call (just like it's faster to use PHP or Servlets rather than old-school CGI for serving web pages). This also means you can run your AGI application on a different server, if you want to, so as not to compromise Asterisk performance. If you know Java, you could try OrderlyCalls at http://orderlycalls.sourceforge.net (disclaimer - written by me!) which has full FastAGI and Manager support, reusable object pooling, and can be run inside Tomcat to build integrated web and telephony applications, though there are other packages out there, including Asterisk-Java ( http://asterisk-java.sourceforge.net - written by Stefan Reuter). Hope this helps, Matt King, Orderly Software http://www.orderlyq.com - probably the most advanced queue system in the world! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OrderlyQ
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 little java experience and it doesn't seem to be running properly. I just thought I should point out that there is no Java experience necessary to run OrderlyQ, and no Java applications to install. We run OrderlyQ as a managed service for you over FastAGI and Manager - all you need do is make some minor modifications to your Asterisk config files (and we'll do this for you if you like). Perhaps you've come across OrderlyQ through our open-source OrderlyCalls platform? There is a java demo of OrderlyQ as part of that package, but it's not required to use OrderlyQ. More details about how to integrate Asterisk with the OrderlyQ managed service (and lots of other useful asterisk queue information) are at http://www.orderlyq.com/asteriskqueues.html . Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MOH - request to schedule in the past SOLUTION and New Asterisk Queues Tutorial.
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 0.59r (on Sarge) and I still get this from time to time (usually when the system is temporarily busy). I don't have a timing source, but nor do I have any particular problems... I presume the music jitters at the time but there's usually no one using it at that moment. We've been able to solve the problem by replacing mpg123 with a player that plays RAW files instead. We haven't seen the problem since. The raw player has the added advantage of increasing Asterisk performance, and it is suitable for use in commercial environments (unlike mpg123 which is not licensed for commercial use without the permission of the author). Full instructions in our new and comprehensive Music On Hold and Queue Tutorial: http://www.orderlyq.com/asteriskqueues.html Hope this helps, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. Author, OrderlyCalls and OrderlyQ http://www.orderlyq.com Tel: +44 1392 421 078 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls
Hello Adam, Thank you so much for taking the time to write to me. I can understand your concerns; let me see if I can address them. Matt, Sourceforge.net is exclusively for hosting software whose licensing terms meet the OSI's definition of Open Source: http://opensource.org/docs/definition.php Your licensing terms include the following, which is not compliant with the OSI definition: Usage Restrictions In addition to the restrictions of the LGPL, the following restrictions apply: ... OrderlyCalls may not be used to provide or augment call queuing without the prior written permission of Orderly Software. 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. Perhaps you could clarify with a more specific reference? Here's the relevant section from the OrderlyCalls licence file (available at http://orderlycalls.sourceforge.net ): --- 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 [EMAIL PROTECTED] 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 advanced queue management system called OrderlyQ, that lets callers hang up and call back when they reach the front of the queue. OrderlyQ is patent-pending, and we do NOT allow the use of OrderlyCalls to provide similar functionality. By adding this restriction, we are erring on the side of caution, so if you want to use OrderlyCalls in conjunction with call queuing, but you are not intending to emulate OrderlyQ, you MUST write to [EMAIL PROTECTED] and explain how you intend to use OrderlyCalls. We anticipate that we will be very happy to give consent in most cases. --- So my first question is, are you objecting to the first usage restriction regarding SPAM calls? We feel this restriction is very important as we sincerely do not wish OrderlyCalls to become a nuisance to anyone. Or are you objecting to the second restriction only? The purpose of the second restriction is to ensure that OrderlyCalls is not used to infringe the intellectual property embodied in OrderlyQ, even by accident, with a view to avoiding litigation and other troubles *before* they can happen. OrderlyQ is a very specific application, and we would only consider witholding permission in cases of clear conflict. We do not wish to restrict the use of OrderlyCalls beyond these boundaries, and by asking people to seek permission before they make the investment of coding, we can co-operatively ensure and verify that their plans do not involve such a conflict. This is to the developer's advantage, as once we've issued permission, the developer can ensure that he/she is not exposed to litigation risk from us. We feel that specifically eliminating this 'grey area' as early as possible in the development process is therefore to everyone's benefit, hence the restriction. I really don't expect we'll be witholding permission very often, if ever. While I understand your motivation and empathize with the plight of open-source business, unfortunately you must either: a) remove this restriction - or - b) remove your project from sourceforge.net Please take action soon so that this matter does not need to be escalated to the sourceforge.net admins. I'm more than happy to refer to sourceforge.net for guidance on this matter, and will do so myself if necessary, however I know they're very busy people, and I'd hate to bother them inappropriately. I also need more information on the specifics of your objection before I can take action. Might I suggest therefore that for the moment we continue this discussion in a spirit of open and friendly co-operation, with a view to finding a solution together, and thereby avoid adding to their workload? I'd also like to suggest that we move this discussion to the OrderlyCalls mailing list, [EMAIL PROTECTED], as I feel this is a more appropriate place for the discussion, and I don't want to burden the inboxes of the subscribers to Asterisk lists inappropriately. You might also choose to respond privately with your concerns; in any case I'd be happy to post the resolution of this issue more widely once we've worked out together exactly what that will be. For the meanwhile, if you're concerned about this issue, and considering using OrderlyCalls with call queues, please don't be scared, and do just ask! Many thanks, Matt King, M.A. Oxon. Managing Director, Orderly
[Asterisk-Users] Re: New JAVA application server for Asterisk - OrderlyCalls
Hello Adam, Matt King [EMAIL PROTECTED] 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 license must not restrict anyone from making use of the program in a specific field of endeavor. For example, it may not restrict the program from being used in a business... http://www.opensource.org/docs/definition.php - a Well the full text of section 6 reads as follows: 6. No Discrimination Against Fields of Endeavor The license must not restrict anyone from making use of the program in a specific field of endeavor. For example, it may not restrict the program from being used in a business, or from being used for genetic research. Rationale: The major intention of this clause is to prohibit license traps that prevent open source from being used commercially. We want commercial users to join our community, not feel excluded from it. I believe that 'field of endeavour' means quite a broad spectrum of activity, such as 'business' or 'genetic research'. It's certainly *not* our intent to discriminate in this way, and I don't think the very specific usage requirements in the licence file could be taken to mean that we're discriminating against any particular field of endeavour. We're certainly *not* intending to prohibit commercial use of OrderlyCalls - indeed we have chosen the LGPL specifically to *encourage* commercial use. We are open to suggestion on this issue, so if you've got a way forward I'd love to hear about it, but in the meantime I'd like to repeat my request to move this discussion off the Asterisk lists and onto a more appropriate forum (such as [EMAIL PROTECTED]), as I feel like we're bugging the readers here with unnecessary detail. This will therefore be my last post to the Asterisk lists on this issue until a way forward has been agreed. Respectfully yours, Matt King, M.A. Oxon. Managing Director, Orderly Software Ltd. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
