Re: [asterisk-users] Dropouts and echo
Tom Lanyon wrote: Hi all, Can I ask that you please keep my personal address in the To: or CC: in this thread as for some reason I'm only getting half of the list emails coming through, and they're not showing up on the digium pipermail archive either. The list archive on http://marc.info seems to have the whole thread though. Have you tried changing the RTP packet size on the phones from .30(default I believe) to .20? The RTP packet size is currently 0.030 which is the default (I wasn't aware we could change it). Would changing to 0.020 help? Why don't you try it? Echo and drop outs usually require trying many different things (especially echo). That is a place to start. Make sure you document your changes and so you can easily roll them back. Finding your actual problem will most likely be a trial and error process, so start trying. Turn OFF CDP on the phones. The phones don't support CDP as far as I can find; I know Linksys is a subdivision of Cisco, but these phones are actually made by Sipura. As for Echo Canceling, that is the job of the device that does VoIP/PSTN gateway functions. As mentioned before, this is SIP - SIP, so the echo isn't introduced by the PSTN. I'll keep experimenting with volume levels and environmental issues to try and fix the echo. Depending on how you define echo, you can try different things with the phones such as pressing mute, stuffing the handset with something to dampen the sound, have both parties speak softly and then try loud. Again, document your results. 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. The switch is indeed a poor quality one. My next step is to replace it with something decent and see if it helps. I wasn't sure whether this could be the cause so it's good to have your input. A switch could cause drop outs but I doubt echo so much. While the first thing you should do is have decent quality gear on your network, I would also look at timing issues on your box and change RTP from packet size to 20. Make sure ZAPTEL or ZTDUMMY is loaded, make sure your system is using RTC and no HPET. Thanks, Steve ___ --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
Re: [asterisk-users] Dropouts and echo
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 [EMAIL PROTECTED] 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. ___ --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
Re: [asterisk-users] Dropouts and echo
Hi all, Can I ask that you please keep my personal address in the To: or CC: in this thread as for some reason I'm only getting half of the list emails coming through, and they're not showing up on the digium pipermail archive either. The list archive on http://marc.info seems to have the whole thread though. Have you tried changing the RTP packet size on the phones from .30(default I believe) to .20? The RTP packet size is currently 0.030 which is the default (I wasn't aware we could change it). Would changing to 0.020 help? Turn OFF CDP on the phones. The phones don't support CDP as far as I can find; I know Linksys is a subdivision of Cisco, but these phones are actually made by Sipura. As for Echo Canceling, that is the job of the device that does VoIP/PSTN gateway functions. As mentioned before, this is SIP - SIP, so the echo isn't introduced by the PSTN. I'll keep experimenting with volume levels and environmental issues to try and fix the echo. 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. The switch is indeed a poor quality one. My next step is to replace it with something decent and see if it helps. I wasn't sure whether this could be the cause so it's good to have your input. 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
Re: [asterisk-users] Dropouts and echo
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 ___ --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
Re: [asterisk-users] Dropouts and echo
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 ___ --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
Re: [asterisk-users] Dropouts and echo
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. ___ --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
Re: [asterisk-users] Dropouts and echo
Drat! That also could be a a major cause of the issues he is having. Dan Austin wrote: 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 ___ --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 ___ --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
