[asterisk-users] Dropouts and echo

Eric "ManxPower" Wieling eric at fnords.org
Tue Jul 31 19:16:04 CDT 2007


Turn OFF CDP on the phones.  I don't know if those phones support CDP, 
but since CDP is the Cisco Discovery Protocol and those Linksys is owned 
by Cisco....  As for Echo Canceling, that is the job of the device that 
does VoIP/PSTN gateway functions.

Tom Lanyon wrote:
> Hi all,
> 
> We have recently implemented an Asterisk system using Trixbox  
> (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting  
> pressure to switch back to our old key system unless we fix two major  
> issues. So please help me avoid switching back!
> 
> An overview:  We have about 12 Linksys SPA941 SIP phones connected on  
> a private switched network to our asterisk box which is a highly- 
> specced HP xeon server. This in turn connects to an Epygi gateway  
> ( http://www.epygi.com/quadro-gateway/70/#isdn ), bringing in 4 ISDN  
> BRI lines as a SIP trunk.
> 
> The issues:
> 	Dropouts - by far the most serious issue we've encountered. On most  
> calls (normally anything longer than 1 or 2 minutes), suddenly one  
> end of the call will go silent and not be able to hear the other  
> person. After a few seconds of "I can't hear you!" the audio returns  
> and continues normally. This seems to happen whether it's an internal  
> call between SIP devices or whether it involves a call via our ISDN  
> gateway. At first we believed this was just when we had our phones on  
> 'speakerphone' and that it was an issue with the physical SIP phone  
> itself, but we're now also finding 'dropouts' just using the phone  
> handset aswell.
> 
> 	Echos - on a majority of calls we can hear an echo of our own voice,  
> a few milliseconds later (enough to be very annoying). From all I've  
> read regarding echo in a VoIP system, I understood that echo was  
> normally introduced by a non-voip device in the system (in our case  
> the external ISDN lines). However, we are having echo produced on a  
> call between two internal staff members between their respective SIP  
> phones.
> 
> Can anyone advise what could cause either of these and what we can do  
> to try and investigate them?
> 
> Thanks,
> Tom
> 
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