[olug] Trixbox/Asterisk call quality

[Shawn L. Djernes]

What do you to app_queue to customize it?

I have custom versions of app_swift to make it multi-lingual. 

---
Shawn L. Djernes 

On Dec 30, 2010, at 22:50, Brian Roberson <roberson at olug.org> wrote:

> 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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