Asterisk 13. Writing Call Quality Parameters To CDR. How?

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Hello.

Voice quality when calling - this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.

Because the overall quality of a call can only be determined upon completion, I did it in the HangUp handler and wrote in custom fields of CDR. This worked well in asterisk 11.

In asterisk 13 I did not find a handler after the call, but before finalizing the CDR. I tried to call the AGI and there to update the CDR record by unique identifiers. But faced with the fact that there are no…

Asterisk Users 6 months ago 2 Answers

Digium IP Phones D40

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Hi All; Any one used Digium IP Phones D40? I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium. Regards
Bilal

Asterisk Users 3.3 years ago 6 Answers

Transcoding degradation G711iLBC

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On 04/15/2012 07:26 PM, Patrick Lists wrote:
> On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
>> Is it a good idea to use asterisk transcoding from G711 to iLBC or
>> should I find out any other solution not involving transcoding (f.e.
>> using G.729 that is supported in both sides). I'm worried about voice
>> quality and trying to avoid paying for G.729 licensing.
>>
>> Anybody with experience or quantitative measurements of the voice
>> quality degradation in that scenario?
>
> The term that may interest…

Asterisk Users 3.4 years ago 0 Answers

Transcoding degradation G711iLBC

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On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
> Is it a good idea to use asterisk transcoding from G711 to iLBC or
> should I find out any other solution not involving transcoding (f.e.
> using G.729 that is supported in both sides). I'm worried about voice
> quality and trying to avoid paying for G.729 licensing.
>
> Anybody with experience or quantitative measurements of the voice
> quality degradation in that scenario? The term that may interest you is "Mean Opinion Score" and iLBC is quite
good. See

Asterisk Users 3.4 years ago 0 Answers

Transcoding degradation G711iLBC

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Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides). I'm worried about voice quality and trying to avoid paying for G.729 licensing. Anybody with experience or quantitative measurements of the voice quality degradation in that scenario? Regards,
G ________________________________
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Asterisk Users 3.4 years ago 0 Answers

SIP jitter and packlost channel variables

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Hi, A client of ours get lots of problem with there voice quality when the do a lot SIP calls.
In a application I log the rtpqos audio jitter an lost packets. (see Below) Does anybody know what the numbers mean?
If I look at a sample of the channel variables, I see the following number.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
..
..
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 70000
remote_minjitter = 14000
..
..…

Asterisk Users 3.5 years ago 1 Answer

local channels and g729a voice quality

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Hi, We noticed a very sharp drop in voice quality when using digium g729a
codec. The problem seems to happen if the A channel (caller's channel)
is a landline/mobile number contacted using the same outgoing provider
(as a local channel). It sounds like listening to a mono speaker on
low volume. If I use a softphone that is directly registered to our asterisk box
the audio quality improves, the words come out more clearer and
louder. I also asked my provider to test call me using their Cisco as5300
system and g729 codec…

Asterisk Users 3.7 years ago 4 Answers

how to find out one way latency

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Hi All, How can I find out One way latency from my PBX to my SIP Trunk Provider.
My SIP provider recommends a One way latency of 100ms for good Voice
quality. Ping request to their IP Address gives me a response in approx.
260ms.
Will that be good enough for a SIP Trunk. Please help. We are trying to sign up with a new SIP Provider. Thanks,
Najim

Asterisk Users 3.8 years ago 11 Answers

IAX between 1.6 and 1.8 has bad voice quality

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I recently upgraded my office server to 1.8 and since then I have very
bad voice quality when calling another Asterisk server that uses 1.6.
The links is via IAX2 and I have tried using g729 and ulaw but I still
have the same problem although ulaw has a slight better result. Any changes that need to me made to the IAX2 trunk for it to work?

Asterisk Users 4.7 years ago 1 Answer