[olug] Trixbox/Asterisk call quality

[Shawn L. Djernes]

May I ask the stupid question of why there is a Call Manager box in this setup?

If they have the phones and trunks both hung off the CM box then the Asterisk box is pushing 20 calls to that box and the CM may be the issue. 

Best to go and have a listen your self at the end they say is the problem. 

I have managed 15 calls on a Dell 1750 2.6ghz 2gb  7 external to IVR, rest mixed phone to phone internal and external on a Cox 3/1. 

Later

---
Shawn L. Djernes 

On Dec 30, 2010, at 20:36, 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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