[Asterisk-Users] Audio problems 50% of the time.

kurt x kurtwp at gmail.com
Wed May 17 13:02:30 MST 2006


I would agree, if I also experience choppy voice.  Over the last month
I had one spike of 893k over my T1.  My average usually is
223k.  I carved out 640k for voice QOS on the WAN router.  At most I
would have 4 calls up at once.

The call comes in, the phone rings,  50% of the time I can have a
conversations.  50% of time I can not.  Maybe I should complain to my
SIP service provider.

Kurt
-----------------------------------------------------------------------------------------------

if your connection is also used for web, email, and the worst, p2p, you
better to have qos on your router.

just be aware that g711 will use 80Kb up and down...
gsm and g729  wil use 30/40Kb

then :
disallow all
allow = gsm
allow = g729



Olivier

kurt x a écrit :
> I have an Asterisk server that I use at work.  I have a phone that is
> at home that logs into
> the Asterisk server at work.  My home phone is hooked up via DSL
> through a Linksys router. You can see the my sip.conf for the phone
> blow.
>
> The problem is each time the phone rings I can hear/be heard 50% of
> the time.
>
> Any suggestion on what to look for.
>
> I do have my reg time set for 180 seconds on the cisco ATA186.
>
> [72459]
> type=friend
> username=XXXXXX
> secret=XXXXX
> host=dynamic
> context=voice-mail
> dtmfmode=rfc2833
> ;canreivet=yes
> nat=yes
> qualify=yes
>
> Kurt



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