The audio issue you mention is likely due to resources. There is no doubt the ACD is a resource hog. Specifically CPU. Which odd as the rest of sipx seems to be more RAM constrained. We used to have issues with the choppy audio until we standardized our platform with an ODM 3-4 years ago. The equipment we have used for the last 3-4 years has a Dual Core Core2 w/ 8GB of ram and an enterprise Intel SSD. Its starting to get dated but even our largest call centers run well on that.
I have not had any of the other issues you mentioned. But I will note we never use the "presence" with the ACD. For us, its not needed. In a call center, the people manning the phones are only their to answer ACD calls. So unless they are on a current queue call....they shouldnt be on the phone. The ACD then doesn't have to worry about subscribing to the presence of each phone. It know when the queue call starts/ends. Other than the ACD has been pretty straight forward for us. Only once in a great while do we get the call thats stuck in the call stats and we have to bounce the CDR. I'd say once every 6 months. But most of our customers get a non-acd stuck call in the CDR about once every six months anyways. They only consistent change we make to the acd to make it stable is to set a local subnet under "internet" for 127.0.0.2 (in fact now we add the entire 127.0.0.0/24 subnet as local) I think 127.0.0.1 is in by default but we cant even pick up queued calls without 127.0.0.2 listed. We discovered that a few years back when 4.2 came out....should be a thread on it. Traces showed the ACD would reference 127.0.0.2 and not 127.0.0.1 and it would think the call is natted and throw the public natted ip in the response. But I'm not sure if that is unique to us or not. We maintain our own builds based on SLES so it may be in how we compile. -M >>> On 8/30/2012 at 07:30 PM, in message >>> <CAEpO7ZwUGFuXu=2UKuwmVLuC1ekia+2L=wgidkts9lu5cqs...@mail.gmail.com>, >>> Melcon Moraes <[email protected]> wrote: I am quite impressed with your success stories. :) I used to have an ACD running pretty well on a 4.0.2 box. Now, I have some 4.4 boxes and all kinds of issues. Have you ever faced some of them? - Audio issues. People complaining about chopping voice and noise. Indeed, if I call the extension directly, without passing through ACD, there's a noticeable difference in the audio quality. - Realtime statistics - some calls that enter the queue and got picked up never "leave" the queue. On sipxacd_events.log one can see the events "enter-queue", "pick-up" and "terminate". Sometimes there is the "transfer" event as well. Some calls never got the "terminate" event written on that log. Then you have on Calls Statistics page calls with +70min long, when the real call took only 3min. - Routing to agent: still on the above example, ACD Presence shows the user as idle but the ACDServer "thinks" the user is still not free. That agent will never get a call again until you restart ACDServer. - sipxacd.log is full of "2012-08-30T18:44:02.139782Z":934039:KERNEL:NOTICE:sip.example.com:MpMedia:B6BB1B90:sipxacd:"OsMsgQShared::doSendCore message queue 'mpStartUp::MpMisc.pSpkQ' is over half full - count = 12, max = 14" +1 on what Todd Hodgen said about the best practices/tips. Sorry to hijack the thread. - MM On Wed, Aug 29, 2012 at 9:29 AM, Matt White <[email protected]> wrote: I've often heard it cited that the old ACD cant transfer out of the queue. I think this is based on very old info....pre 4.2 days. We have no less than hundreds of calls (if not thousands) across several customers sites sending calls into queues and back out to non ACD extensions. ACD has preformed very well for us and is very stable (i cant think of one ACD related crash/issue). When it comes to transferring out of the queue, I'd say our heaviest call center customer that has 23 agents and processes no less than 800 calls a day will transfer 50% of the calls from and agent out of the queue to a regular extension. They even transfer ACD calls back out to external numbers (hairpins). In fact, we have a customer that has some pretty complex call flow in and out of the ACD. Calls come into the queue. If its not answered it overflows out to an AA where they get custom options to keep holding etc. they then get transferred back into a second ACD queue. This happens multiple times. Siptraces are quite long because each call stays anchored in the queue which is great for reporting. Hunt groups do not provide was call centers want. Call center managers need to run reports based on when an agent signed in/out; how long did they talk; how long did they ring; how long are customers waiting in queues and how many customers are stacking up in queues.... The distinctive tone as noted in this original thread and modified caller id are also part of that need. Hunt groups just dont preform these functions (if they did, they would be an ACD and not a hunt group). So for us, the bulk of our installations are call centers. As VOIP/SIP becomes more common place its increasingly difficult to sell something that just does call routing and voicemail...there is just so many cheap and hosted solutions if thats all you need. Which is why are installations for the last few years have moved up the stack to customers that have more specialized needs. So for us, our dilemma is get sipXacd maintained internally (we have an elance dev working on that now), or look for a new platform. -M >>> Michael Picher <[email protected]> 08/29/12 6:53 AM >>> The problem with the old ACD, every time I tried to use it, was that it was fragile. Do this, don't ever do that, etc... Any time I tried to use it in a situation the customer (and then ultimately me) had a bad experience. If they can make it stable, and make it so you can transfer calls out of queue that would be awesome. Most people don't need a real ACD (even though they think they want an ACD). What they need are fancy hunt groups which is exactly what the 4.4 ACD does. Work has begun on a new hunt group app that will hopefully alleviate the need for an ACD in cases where it really isn't needed. This will be based around some new code and involve a B2BUA that can 'own' the call and then hunt out. This will be in contrast to how hunt groups work now where it's a SIP messaging nightmare. Circular hunt groups, linear hunt groups, being able to ring the same extension at more than one point in a hunt group are all envisioned. At some point I'd even like to see users be able to (from a phone or user portal) login/out of hunt groups, or only ring certain users for different days/hours in a hunt group. With the current workload I wouldn't expect anything until 4.8 though. Thanks, Mike On Wed, Aug 29, 2012 at 6:21 AM, Matt White <[email protected]> wrote: Funny, the beep is one reason my customers wont move to openACD. OpenACD for my customers that need a true ACD. The old ACD worked well despite its bad rap. We have a developer working on get sipXacd ported over to 4.6 so there is hope yet. As to your specific issue, I don't think the tone is an audio file, so you would need to create a patch to remove it. -m >>> Ali Dashti <[email protected]> 08/28/12 9:14 AM >>> Thanks Tony, one thing that made me go back to 4.4 ACD was the CallerID and DNID! In 4.4 ACD when a call arrives; an agent would see a Queue name and the Line extention in CallerID but in OpenACD even the DNID changes to agent extension number; therefore prevents me from knowing what number was dialed!! This was my primary result, do you see this as well? On Tue, Aug 28, 2012 at 5:29 PM, Tony Graziano <[email protected]> wrote: fyi - If you are referring to the existing ACD (not openacd) I don't think any resources are going into it as it is being removed in favor of the openacd integration starting up in 4.6. On Tue, Aug 28, 2012 at 8:55 AM, Ali Dashti <[email protected]> wrote: This could be a very old issue! I was wondering if there is a way to remove the beep heard a short moment after an agent picks up a call on ACD!! Thanks. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 ( tel:434.984.8430 ) sip: [email protected] Fax: 434.465.6833 ( tel:434.465.6833 ) ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ Using or developing for sipXecs from SIPFoundry? 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