[asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

[Mitul Limbani]

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 negotiations"

Mitul Limbani
www.facebook.com/enterux
www.facebook.com/entvoice
 On Oct 29, 2013 1:30 AM, "Ron Wheeler" <rwheeler at artifact-software.com>
wrote:

>  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.
>
> 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 regardless of who places the call.
> As a result,  I use Skype for all of my calls and if someone calls me, I
> call them back on Skype if they have any problems.
> I don't understand why Skype works so well and Asterisk works so poorly on
> the same environment.
>
> Googling "Asterisk poor audio quality" return several hundred thousand
> references
>
> Ron
> On 28/10/2013 2:29 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 server from VM machine to dedicated machine
>
>  More than enough bandwidth
>
>  Setting 802.1p = 7
>
>  Set Dedicated voice traffic 35% of bandwidth.
>
>  Not sure what option would be the best
>
>
>   Put analog lines in the conference room to avoid the dropouts - leave
> the sip lines in place for day to day use
>
>  Hire a consultant
>
>  Ditch the system and buy a pre-packaged system - RingCentral or some
> such.
>
>  There are no local asterisk professionals who can help, and we are a
> little leery of opening up our system to outside consultants.
>
>  Anyone else face the above, and finally abandoned Asterisk for a
> commercial system?
>
>  We have 167 users.
> I use Grandstream GXP 2100 on the desktop and Polycom ip6000 for the
> conference rooms.
>
>  Suggestions welcome.
>
>  Best
>
>  Eddie
> --
> Eddie H. Mikell
> Senior Systems Engineer
> RKG
>
>  Office: 434.970.1010 x 124
> Email: emikell at rimmkaufman.com
>
>  <http://www.rimmkaufman.com>
> <http://twitter.com/rimmkaufman>  <http://www.linkedin.com/company/85385> <http://plus.google.com/104980442218952272663/posts>
>   <http://www.facebook.com/rimmkaufman>  <http://www.RKGblog.com>
>
>
>
>
>
>
> --
> Ron Wheeler
> President
> Artifact Software Inc
> email: rwheeler at artifact-software.com
> skype: ronaldmwheeler
> phone: 866-970-2435, ext 102
>
>
> --
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