For two years and on every installation I have ever touched there is one nagging annoyance that is very disappointing & have never been able to figure out via Google/Voip-info/whirlpool/etc.
Transferring a Zap call to another Zap channel (bridging), conferencing a Cisco SIP extension with a couple Zap channels, MeetMe conferencing a couple Zap channels, etc. ---- all drop volume once the second+ lines are added to the point frequently of not being able to hear without yelling.
HELP please, myself and many installations of Asterisk still suffer from this.
Environment: All installs running latest Asterisk & Zaptel, all have TDM400P’s (1 or more quad FXO’s), none have echo issues, all have Cisco 7960G handsets running SIP 7.4.
Experiencing the same issue. Can anyone shed some light on this one?
No audio issues when people call in and reach SIP users and vice versa but none of the users on the ZAP channels can hear each other. I’ve noticed this with MeetMe and DISA too, where the audio is extremely faint when I dial in on a ZAP line and try to dial back out on another ZAP line or on an outbound SIP connection.
I hope someone’s got a clue!
After reading another forum, I tried this:
My the two ZAP channels could hear each other at last!
Can you give it a try and post your results?
this article spends more time regarding echo and volume setting but it does a good job explaining how to use zttool to measure and adjust your rx and tx gains.