Tag Archives: Sip
Codec Translation I have already mentioned before. In the path caller->called codecs can change several times, and every transformation can introduce distortion and delay. To minimize fax issues all the legs involved must be using the same codec to avoid … Continue reading
Certainly there are huge benefits in adopting VoIP technology (i.e. Asterisk PBX), and certainly there will be many people who will be wise to use Voip PBX trunk instead of traditional phone lines. But using Voip PBX you may start … Continue reading
In this post I’ll show how to create a Sip trunk between Avaya IP Office and Asterisk pbx.
This is a quick and dirty configuration process for Asterisk setup and AudioCodes MP-118 Gateway SIP/Analog FXS.