I just signed up with livevoip for 800 DID and have very choppy audio. From
PSTN to my asterisk is ok but
asterisk to PSTN is terrible. I am using IAX and was assigned to server
iax01.nyc.*. I do not believe it is
a bandwidth problem on my end and I have no problems using iax with
gafachi. I opened a ticket with
livevoip but no response yet. Would I be better off using sip with them? Is
there a server with better
response/bandwidth?
I admit that I am running a cvs head may 2004 prior to 1.x.x release. Could
that be the problem?
Regards,
John
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
