Hi there,
In other words you are maybe on 60ms and they are on 20ms or vice versa.
Do a wireshark trace and see if the codecs and ptime agree on both sides
otherwise you will get grabbled sounds.
On 10/29/2013 02:49 PM, Daniel van den Berg wrote:
> Hi there,
>
> Sounds like codec ptime mismatch..
Hi there,
Sounds like codec ptime mismatch...what codec are you using? If you are
using g729 make sure that you and your provider is giving the same ptime.
On 10/29/2013 11:55 AM, Stelios Koroneos wrote:
> On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote:
>> All,
>>
>>
>> The users in our or
On Mon, 2013-10-28 at 14:29 -0400, Eddie Mikell wrote:
> All,
>
>
> The users in our organization are well, quite frankly, sick of phone
> service that is being provided. The choppy phone calls, and drop outs
> are detrimental to our sales force.
>
>
> I've tried about everything I can think o
On 28/10/2013 4:12 PM, Mark Wiater wrote:
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the
Asterisk - No analogue.
I don't have any problems w
> In my case, I have good incoming quality and terrible quality going out.
> That is, I can hear people perfectly well but they complain that my
> voice drops out and is garbled regardless of who places the call.
This suggests to me that you may have congestion problems in your
"upstream" traffic
On 13-10-28 06:03 PM, Patrick Lists wrote:
On 10/28/2013 07:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about everythin
Steve Edwards wrote:
> What? Why did my bandwidth dive from 800 Mbits/sec to 1 Mbits/sec?
--help shows:
Client specific:
-b, --bandwidth #[KM]for UDP, bandwidth to send at in bits/sec
(default 1 Mbit/sec, implies -u)
Doug
--
Ben Franklin quote:
"Those who woul
On Mon, 28 Oct 2013, Mike wrote:
I found iperf (http://iperf.sourceforge.net/) to be a free and easy
starting point, which actually turned out to be all I needed.
I've used iperf to check bandwidth before, but never looked deeper into
it's features. Thanks for the nudge. Maybe you can help me
On 10/28/2013 07:29 PM, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs
are detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk se
iperf is great. Another essential troubleshooting tool is nfsen/nfdump
(or any netflow/sflow monitoring utility that shows DSCP/TOS tag values).
On 10/28/2013 01:55 PM, Mike wrote:
As stated in previous replies if you haven't already I would certainly
try to isolate the problem, e.g., are exten
On 29/10/2013, at 9:55 am, Mike wrote:
> On Mon, 28 Oct 2013, Eddie Mikell wrote:
>
>> All,
>> The users in our organization are well, quite frankly, sick of phone service
>> that is being provided. The choppy phone
>> calls, and drop outs are detrimental to our sales force.
>> I've tried abo
Ron Wheeler писал 28.10.2013 21:59:
> I have not found any
good tools to track down the causes of poor voice quality.
> In my case,
I have good incoming quality and terrible quality going out.
> That is,
I can hear people perfectly well but they complain that my voice drops
out and is garbled
On Mon, 28 Oct 2013, Eddie Mikell wrote:
All,
The users in our organization are well, quite frankly, sick of phone service
that is being provided. The choppy phone
calls, and drop outs are detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk
Asterisk is a swiss army knife, you should either know how to use it or
rely on ready made software which control routing of calls through variable
bit rates (skype does that very effectively)
So the key here for you to research upon from those several hundred results
is "variable bit rate codec n
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Monday, October 28, 2013 4:00 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Tired of dropouts and garbled phone calls - where
to go next?
I am reaching the same level of frustration.
I have
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No
analogue.
I don't have any problems with IAX, but I hear some do.
We have a ve
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the
Asterisk - No analogue.
We have a very lightly loaded 60 Mbs cable link to the Internet that
tests pretty close to that most of the time.
asterisk-users-boun...@lists.digium.com wrote on 10/28/2013 01:29:13 PM:
> From: Eddie Mikell
> To: asterisk-users@lists.digium.com,
> Date: 10/28/2013 01:29 PM
> Subject: [asterisk-users] Tired of dropouts and garbled phone calls
> - where to go next?
> Sent by: a
All,
The users in our organization are well, quite frankly, sick of phone
service that is being provided. The choppy phone calls, and drop outs are
detrimental to our sales force.
I've tried about everything I can think of.
Moved the asterisk server from VM machine to dedicated machine
More th
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