I’m willing to pay a consultant for some answer to simple questions…unless someone is feeling generous.
I have an Asterisk server set up based on AsteriskNOW.
What are the bandwidth guidelines for Asterisk? I’m stuck with ADSL. I don’t quite understand how to debug my call quality issues. Is it based on bandwidth limitations or a misconfiguration? My SIP provider is a company called voipvoip.com.
The problem I’m experiencing currently is poor quality on the external caller side. The initiating caller sounds great on my end but I sound terrible on his end.
I should also explain that I’m attempting to use Asterisk for a radio call in show. I have my softphone’s output routed to my broadcast console’s dedicated caller channel.
Any help or pointers would be much appreciated. Again, if you lead me to a solution, I’m willing to pay for your time.