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

[Damon Estep]

I get the impression the complaint is NO audio, not poor audio. This
points more to NAT than QoS.

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Derek
Lee-Wo
Sent: Wednesday, May 17, 2006 1:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Audio problems 50% of the time.

This is most likely your upload speed.  I have Comcast supposedly with
384KB upload, but I have a hard time using VoIP unless I use a
low-bandwidth codec like GSM.  For g711, it's a crap shoot as to
whether it works or not.

I can always hear the other person clearly since I have a ton of
download bandwidth available, but they have a hard time hearing me and
I tend to break up a lot.

Derek


On 5/17/06, kurt x <kurtwp at gmail.com> wrote:
> 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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