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

[Eric Wieling]

Does using SIP to your ITSP make any difference?    I stopped using IAX2 and switched to SIP around 2003 when I experienced similar problems, never looked back.  If you insist on using IAX2, then Google for iax2 audio problems

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ron Wheeler
Sent: Monday, October 28, 2013 4:00 PM
To: asterisk-users at 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 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 <mailto: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