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
Tom Wrote:> 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!Have you tried changing the RTP packet size on the phones from .30(default I believe) to .20? That may help the cut-out issue. I wouldn't bet on it helping with the echo issue, which I would approach by tweaking the phone volume levels and seeing if environmental issues my play a part. Dan
Turn OFF CDP on the phones. I don't know if those phones support CDP, but since CDP is the Cisco Discovery Protocol and those Linksys is owned by Cisco.... As for Echo Canceling, that is the job of the device that does VoIP/PSTN gateway functions. Tom Lanyon wrote:> 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 >
But the OP stated this was SIP <-> SIP calls -- As Dan mentioned, check environmental issues -- hard walls, poor handset quality, noisy desks, volume levels too high? Eric "ManxPower" Wieling wrote:> by Cisco.... As for Echo Canceling, that is the job of the device that > does VoIP/PSTN gateway functions.
dave cantera
2007-Aug-01 00:39 UTC
[asterisk-users] check out the cursor movement on this website!!!
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What kind of switch are you connecting the phones to? I've seen that behaviour with cheap Repotec switches (24+2Gigabit). Just replacing it with a different one fixed the problem. Julian J. M. On 7/31/07, Tom Lanyon <tom at netspot.com.au> wrote:> 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.