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

[Derek Lee-Wo]

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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>