[olug] Trixbox/Asterisk call quality

[Brian Roberson]

Are the agent handsets sccp? Is this a CCME system? Just trying to
understand the call flow a little better... So the call comes in TDM, sip to
asterisk, sip back to the ccm gateway, then to the agent? If thats the
scenario, are other calls from the handsets garbled when the queue hits the
magical number 10? if not, then you can narrow it down to the the asterisk
call legs...

This sounds like a system I've setup in a number of small businesses in
town, if this is a client of yours, and not your own setup, hit me off the
list and I may be able to filter additional detail to you :)

Sean is right that the vanilla app_queue is usually the same version of the
core asterisk system, however, none of the hundred I've setups I've done
are. I've found that the app_queue shipping in all versions of trixbox are
broke in one way or another, so I tend to keep my own versions.

"strings /usr/lib/asterisk/modules/app_queue.so | grep -i roberson"

That will let you know if it is one of my custom ones.




On Thu, Dec 30, 2010 at 8:36 PM, jay swackhamer <reboottheuser at gmail.com>wrote:

> This is not going over the internet(as far as I know, will need to get more
> details on the env), its supposed to be cox voice trunk->cisco call
> manager->trixbox.........
> On Dec 30, 2010 8:05 PM, "Shawn L. Djernes" <shawn at djernes.org> wrote:
> > Your app_queue version is the same as Asterisk.
> >
> > The usual cause for this problem is to due with your upstream bandwidth
> from your Internet provider.
> >
> > If you ate using G.711 that it roughly uses 80k per call between the data
> and packet header. Ten calls plus other traffic can overwhelm a 1M upstream
> pipe from Cox. Using QOS will improve things, but the best option is to
> increase the upstream connection speed.
> >
> > If audio quality is not very important, you can see if your carrier
> supports smaller codecs like GSM or G.729. G.729 requires paying a license
> fee but it's sound quality is midrange between G.711 and GSM. It uses about
> 16k per call.
> >
> > Hope this helps.
> >
> > ---
> > Shawn L. Djernes
> >
> > On Dec 30, 2010, at 16:12, jay swackhamer <reboottheuser at gmail.com>
> wrote:
> >
> >> Asterisk 1.6.0.9-samy-r27
> >>
> >> Where do I find the app_queue version?
> >>
> >> Garbled in music-on-hold is what I've been told
> >>
> >> SIP
> >>
> >> Someone on-site that has experienced the problem is listening to the
> clips
> >> to relay the information about what they hear.
> >>
> >>
> >> On Thu, Dec 30, 2010 at 3:49 PM, Brian Roberson <roberson at olug.org>
> wrote:
> >>
> >>> we need details!
> >>>
> >>> 1) version of asterisk?
> >>> 2) version of app_queue?
> >>> 3) Garbled to agent or to caller or both?
> >>> 4) codec in use?
> >>> 5) ingress call leg is TDM or SIP to asterisk?
> >>> 6) are you *sure* it is garbled audio? please listen to these clips to
> make
> >>> sure which exact symptom you have:
> >>>
> >>>
> >>>
>
> http://www.cisco.com/en/US/tech/tk652/tk698/technologies_white_paper09186a00801545e4.shtml
> >>>
> >>>
> >>>
> >>>
> >>> On Thu, Dec 30, 2010 at 3:33 PM, jay swackhamer <
> reboottheuser at gmail.com
> >>>> wrote:
> >>>
> >>>> Does anyone have experience with the call audio being garbled once 10+
> >>>> callers are queued up?
> >>>>
> >>>> Hardware = Dell 2650, 2x3.06Ghz 3GB ram, 2.6GB used, Hyperthreading
> >>>> disabled.
> >>>>
> >>>> When Hyperthreading was turned on, the call quality after 10 was the
> >>> same,
> >>>> but top/sar showed minimal usage. With Hyperthreading disabled It
> shows
> >>> up
> >>>> to 100% utilized.
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