Hi all, We have recently implemented an Asterisk system using Trixbox (asterisk v1.4.4 at the moment, yet to move to 1.4.9) but are getting pressure to switch back to our old key system unless we fix two major issues. So please help me avoid switching back!
An overview: We have about 12 Linksys SPA941 SIP phones connected on a private switched network to our asterisk box which is a highly- specced HP xeon server. This in turn connects to an Epygi gateway ( http://www.epygi.com/quadro-gateway/70/#isdn ), bringing in 4 ISDN BRI lines as a SIP trunk. The issues: Dropouts - by far the most serious issue we've encountered. On most calls (normally anything longer than 1 or 2 minutes), suddenly one end of the call will go silent and not be able to hear the other person. After a few seconds of "I can't hear you!" the audio returns and continues normally. This seems to happen whether it's an internal call between SIP devices or whether it involves a call via our ISDN gateway. At first we believed this was just when we had our phones on 'speakerphone' and that it was an issue with the physical SIP phone itself, but we're now also finding 'dropouts' just using the phone handset aswell. Echos - on a majority of calls we can hear an echo of our own voice, a few milliseconds later (enough to be very annoying). From all I've read regarding echo in a VoIP system, I understood that echo was normally introduced by a non-voip device in the system (in our case the external ISDN lines). However, we are having echo produced on a call between two internal staff members between their respective SIP phones. Can anyone advise what could cause either of these and what we can do to try and investigate them? Thanks, Tom _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
