I have the following case. A customer is a heavy Meetme/audio conference user. He is equiped with Polycom SS2W (DECT SoundStation 2W audio conference station). Users complain they “often do not hear the other party loud enough”.
The setup is then:
Remote party <--- PSTN/ISDN---> Asterisk <---SIP---> Kirk300 <---DECT--->
My questions are:
1. How can I measure audio strength/loudness/quality and strip social/psychological interferences off ?
2. Is there any builtin mechanism inside Asterisk (this setup is 1.6.1 but upgrade is possible) that can change call volume ?
3. Given my setup is purely digital, could it be the source of calls not being loud enough ?
4. Suggestions ? Comments