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 <[EMAIL PROTECTED]> 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
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