Re: [asterisk-users] Call Quality Measuring
Hi Patrick, You are welcome to try our tools out for active and passive voice quality measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP metrics analysis (like G.107 E-model and other metrics). You can read more at http://www.sevana.biz or older site http://www.sevana.fi On Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont < p.beaum...@hatsoffsoftware.co.uk> wrote: > Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor > in the near future. > > So can I assume from the lack of discussion nobody is using the “sip show > channelstats” stuff? > > Regards, > Patrick. > > On 31/03/2015 08:23, "Olivier" wrote: > > >Some SIP hardphones (Polycom) or softphones (Counterpath) embed a > >module that metter MOS. > > > > > >Regards > > > >2015-03-25 14:21 GMT+01:00 Patrick Beaumont > >: > >> Hi everyone. > >> > >> We regularly get customers complaining about call quality issues. Most > >>of > >> the time it turns out to be their own broadband. Very occasionally > >>server > >> load. Does anyone have any advice or links to advice on measuring call > >> quality? > >> > >> I’ve been playing around with “sip show channelstats” but can’t other > >>than > >> measuring the packet loss I don’t really know what I’m supposed to be > >> looking for in order to say “ah ha! that’s the problem!”. I also don’t > >> know what it’s limits are. Will the stats in “sip show channelstats” > >>show > >> a customer using a torrent client and saturating their own broadband > >> connection? > >> > >> Regards, > >> Patrick. > >> > >> -- > >> _ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> New to Asterisk? Join us for a live introductory webinar every Thurs: > >>http://www.asterisk.org/hello > >> > >> 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 -- > >New to Asterisk? Join us for a live introductory webinar every Thurs: > > http://www.asterisk.org/hello > > > >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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring
Thanks for the suggestions guys. I’ll try to have a play with Voipmonitor in the near future. So can I assume from the lack of discussion nobody is using the “sip show channelstats” stuff? Regards, Patrick. On 31/03/2015 08:23, "Olivier" wrote: >Some SIP hardphones (Polycom) or softphones (Counterpath) embed a >module that metter MOS. > > >Regards > >2015-03-25 14:21 GMT+01:00 Patrick Beaumont >: >> Hi everyone. >> >> We regularly get customers complaining about call quality issues. Most >>of >> the time it turns out to be their own broadband. Very occasionally >>server >> load. Does anyone have any advice or links to advice on measuring call >> quality? >> >> I’ve been playing around with “sip show channelstats” but can’t other >>than >> measuring the packet loss I don’t really know what I’m supposed to be >> looking for in order to say “ah ha! that’s the problem!”. I also don’t >> know what it’s limits are. Will the stats in “sip show channelstats” >>show >> a customer using a torrent client and saturating their own broadband >> connection? >> >> Regards, >> Patrick. >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >>http://www.asterisk.org/hello >> >> 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 -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring
Some SIP hardphones (Polycom) or softphones (Counterpath) embed a module that metter MOS. Regards 2015-03-25 14:21 GMT+01:00 Patrick Beaumont : > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call > quality? > > I’ve been playing around with “sip show channelstats” but can’t other than > measuring the packet loss I don’t really know what I’m supposed to be > looking for in order to say “ah ha! that’s the problem!”. I also don’t > know what it’s limits are. Will the stats in “sip show channelstats” show > a customer using a torrent client and saturating their own broadband > connection? > > Regards, > Patrick. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring
Hi Markus, Sounds interesting to me too... However my google-fu is letting me down today - I found VOIPmonitor at Sourceforge http://sourceforge.net/projects/voipmonitor/ but this looks like you'll need a license. Any chance you have a link to voipmon? Cheers .. Brendan Ord OntheNet - Network Engineer P 07 5553 9222 F 07 5593 3557 Level One, 165 Varsity Parade Varsity Lakes Qld 4227 (Map) www.OntheNet.com.au NOTICE: This e-mail and any attachments are private and confidential and may contain privileged information. If you are not an authorised recipient, the copying or distribution of this e-mail and any attachments is prohibited and you must not read, print or act in reliance on this e-mail or attachments. Any pricing information supplied via email is an estimate or indicative only and may require a formal quotation to verify full terms and conditions. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler Sent: Thursday, 26 March 2015 7:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Quality Measuring Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most > of the time it turns out to be their own broadband. Very occasionally > server load. Does anyone have any advice or links to advice on > measuring call quality? > > I’ve been playing around with “sip show channelstats” but can’t other > than measuring the packet loss I don’t really know what I’m supposed > to be looking for in order to say “ah ha! that’s the problem!”. I also > don’t know what it’s limits are. Will the stats in “sip show > channelstats” show a customer using a torrent client and saturating > their own broadband connection? > > Regards, > Patrick. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring
Hi Patrick, try voipmon, there it's free and you can even track MOS. Markus Am 25.03.2015 um 14:21 schrieb Patrick Beaumont: Hi everyone. We regularly get customers complaining about call quality issues. Most of the time it turns out to be their own broadband. Very occasionally server load. Does anyone have any advice or links to advice on measuring call quality? I’ve been playing around with “sip show channelstats” but can’t other than measuring the packet loss I don’t really know what I’m supposed to be looking for in order to say “ah ha! that’s the problem!”. I also don’t know what it’s limits are. Will the stats in “sip show channelstats” show a customer using a torrent client and saturating their own broadband connection? Regards, Patrick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring (Laszlo)
Have you tried using tcpdump? Then analyze the pcap on wireshark? Marlon Araujo > On Mar 25, 2015, at 13:00, asterisk-users-requ...@lists.digium.com wrote: > > 1. Re: Call Quality Measuring (Laszlo) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Measuring
On Wed, Mar 25, 2015 at 2:21 PM, Patrick Beaumont < p.beaum...@hatsoffsoftware.co.uk> wrote: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have any advice or links to advice on measuring call > quality? > > I’ve been playing around with “sip show channelstats” but can’t other than > measuring the packet loss I don’t really know what I’m supposed to be > looking for in order to say “ah ha! that’s the problem!”. I also don’t > know what it’s limits are. Will the stats in “sip show channelstats” show > a customer using a torrent client and saturating their own broadband > connection? > > Regards, > Patrick. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users You can try voipmonitor (http://voipmonitor.org) free for 30 days, hopefully it's enough for finding and fixing the call quality issues. (I'm not affiliated with voipmonitor) -- -- Kind regards, Laszlo Bekesi http://voipfreak.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality - how to debug
Hi Steve, Mainly because, if it were a CPU utilisation issue, then putting an extra load on the server because of tcpdump isn't going to help. If I go that route then I'll port mirror on the switch. But thanks for the reply, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 16:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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] Call quality - how to debug
Hi Dave, You're quite right, it's a dedicated down and uplink to my ISP, and Gradwell also has fibre connection into that ISP (so short hop to them) The reason I don't think it's the fiber link, is that Asterisk recorded the conversation as two channels. IN (from Gradwell), and OUT (from the Cisco phone, that's on the same LAN as the asterisk server). And I hear distortion on both sides, at the same time. As thats what asterisk "hears", and that part of the call is a same-LAN RTP stream, pre-ISP, then that's why I don't think it's the IAX link. That said, I've not got complaints from users making internal calls. So my thinking was maybe its an IAX/SIP conversion thing As a test, I've switched my account, and the problem account to inbound SIP, to see if that makes a difference. That makes it 100% SIP. Next step, memory upgrade and the A*k upgrade. Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 16:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? So theres no web browsing etc on that 2mb circuit. In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: h
Re: [asterisk-users] Call quality - how to debug
Unless I've misunderstood and you're not running ANYTHING but voice over that internet uplink? So theres no web browsing etc on that 2mb circuit. In which case, I stand corrected and you don't need QOS. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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 ___ -- 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 ___ -- 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] Call quality - how to debug
I think you're overlooking your internet uplink, which is what I'm talking about: Inbound calls are via in IAX trunk from Gradwell. You certainly DO need QOS to maintain call quality over the INTERNET link. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 10:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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 ___ -- 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/lis
Re: [asterisk-users] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > I’m at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I’m not sure what that will achieve. Why not just tcpdump on the asterisk box then load it into wireshark? ___ -- 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] Call quality - how to debug
On Tue, 2009-06-02 at 15:49 +0100, Adrian Marsh wrote: > However - my question would still stand, how exactly would I be able to > debug whats going on in the RTP stream? And why its stuttering > (sometimes halfway through a call). > > Any tips or tricks for actually debugging within Asterisk ? Wireshark has a lot of RTP tools for looking at the latency and jitter and dropped packets on the line, which are the most common problems I find when helping people diagnose poor audio connections. It won't tell you what is *causing* the problem, but it will help you know what the problem actually is. >From there, you can start to track down the source of the problem one network segment at a time. For example... is the poor audio being caused by network problems between the phone and Asterisk, or between Asterisk and your upstream provider. -- Jared Smith Training Manager Digium, Inc. ___ -- 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] Call quality - how to debug
Yeah, I know, but when I last tried an upgrade to 1.4.18 it broke the whole IAX connectivity and I was forced to drop back. I'll go: 1) Memory upgrade first 2) Clone the machine, and upgrade to latest 1.4.x However - my question would still stand, how exactly would I be able to debug whats going on in the RTP stream? And why its stuttering (sometimes halfway through a call). Any tips or tricks for actually debugging within Asterisk ? Thanks, Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Darrick Hartman Sent: 02 June 2009 15:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: > Hi, > > It's a 2mb dedicated leased fibre line, with<50% utilisation. > My first thoughts were the internet link, but that wouldn't explain why > the client transmit (other channel), which is on the same LAN as the > server, would have the same problem at the same time. > > Gut feeling is that A*k was CPU overloaded, or the local LAN was, but > none of the stats show that going on at all, although my CPU stats are > 5min samples - so that might hide a 60s of intense CPU activity. > > It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. > Only runs Asterisk. > > Adrian > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Howes > Sent: 02 June 2009 14:23 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call quality - how to debug > > > On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > >> Hi All, >> >> I've a 1.4.15 A*k server supporting several users (approx 80 total, >> but<10 sim calls usually). I've one user who complains of >> intermittent bad calls, though I suspect the bad calls are across >> the board, but intermittent. >> >> Inbound calls are via in IAX trunk from Gradwell. CPU stats say that >> Asterisk never uses more than 4-5% cpu, systems idle besides that. >> Memory seems ok too. Network utilisation is< 300kbps. The voice >> network (clients + server) sit on their own dedicated 100Mb >> switches. Stats from the switch say its lightly loaded. >> >> I've turned on voicefile recording. What we hear, when there is a >> bad call, is stuttered speech, from BOTH sides (so local SIP client, >> and remote IAX inbound call). >> Debug from asterisk just shows the call inbound, answered and then >> hung up as per normal. >> >> I'm at a loss of how to debug the voice issue further, without >> putting a wireshark PC on the switch, port-mirroring the server and >> then capturing all of the traffic in a round-robin-type capture and >> even then I'm not sure what that will achieve. >> >> I'm going to switch from IAX to SIP for the inbound calls for that >> user and see if that helps. >> >> Any ideas welcome, ___ -- 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] Call quality - how to debug
Do you have any idea the number of bugs that have been fixed since 1.4.15? Upgrade to 1.4.25 (or 1.4.26-rc1) before attempting to debug this. On 06/02/2009 08:58 AM, Adrian Marsh wrote: > Hi, > > It's a 2mb dedicated leased fibre line, with<50% utilisation. > My first thoughts were the internet link, but that wouldn't explain why > the client transmit (other channel), which is on the same LAN as the > server, would have the same problem at the same time. > > Gut feeling is that A*k was CPU overloaded, or the local LAN was, but > none of the stats show that going on at all, although my CPU stats are > 5min samples - so that might hide a 60s of intense CPU activity. > > It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. > Only runs Asterisk. > > Adrian > > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve > Howes > Sent: 02 June 2009 14:23 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Call quality - how to debug > > > On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > >> Hi All, >> >> I've a 1.4.15 A*k server supporting several users (approx 80 total, >> but<10 sim calls usually). I've one user who complains of >> intermittent bad calls, though I suspect the bad calls are across >> the board, but intermittent. >> >> Inbound calls are via in IAX trunk from Gradwell. CPU stats say that >> Asterisk never uses more than 4-5% cpu, systems idle besides that. >> Memory seems ok too. Network utilisation is< 300kbps. The voice >> network (clients + server) sit on their own dedicated 100Mb >> switches. Stats from the switch say its lightly loaded. >> >> I've turned on voicefile recording. What we hear, when there is a >> bad call, is stuttered speech, from BOTH sides (so local SIP client, >> and remote IAX inbound call). >> Debug from asterisk just shows the call inbound, answered and then >> hung up as per normal. >> >> I'm at a loss of how to debug the voice issue further, without >> putting a wireshark PC on the switch, port-mirroring the server and >> then capturing all of the traffic in a round-robin-type capture and >> even then I'm not sure what that will achieve. >> >> I'm going to switch from IAX to SIP for the inbound calls for that >> user and see if that helps. >> >> Any ideas welcome, ___ -- 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] Call quality - how to debug
Scratch that, my inventory tool says the system has 256Mb not 1Gb. I wonder if a memory upgrade would help it out... -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: 02 June 2009 14:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
I don't need QoS. The voice network here is seperated from the PC LAN physically (2 separate switches), by design. So theres no web browsing etc on that 2mb circuit. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Gibbons Sent: 02 June 2009 15:09 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality - how to debug 2mb is small potatoes... unless you mean MegaBytes instead of Megabits... I am assuming you've already implemented QOS? That is likely the problem if the intermittent quality issue is only on calls between internal and external parties. If someone tries to access the yahoo homepage while someone else is on the phone, without QOS, they are really going to be fighting for that bandwidth. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adrian Marsh Sent: Tuesday, June 02, 2009 9:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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 ___ -- 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] Call quality - how to debug
Hi, It's a 2mb dedicated leased fibre line, with <50% utilisation. My first thoughts were the internet link, but that wouldn't explain why the client transmit (other channel), which is on the same LAN as the server, would have the same problem at the same time. Gut feeling is that A*k was CPU overloaded, or the local LAN was, but none of the stats show that going on at all, although my CPU stats are 5min samples - so that might hide a 60s of intense CPU activity. It's a physical machine, Centos 4.6, 3ghz CPU Celeron, 1gb ram. Only runs Asterisk. Adrian -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: 02 June 2009 14:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality - how to debug On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I've a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I've one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I've turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I'm at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I'm not sure what that will achieve. > > I'm going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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] Call quality - how to debug
- "Steve Howes" wrote: > On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > > > Hi All, > > > > I’ve a 1.4.15 A*k server supporting several users (approx 80 total, > > but <10 sim calls usually). I’ve one user who complains of > > intermittent bad calls, though I suspect the bad calls are across > > the board, but intermittent. > > > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > > Asterisk never uses more than 4-5% cpu, systems idle besides that. > > Memory seems ok too. Network utilisation is < 300kbps. The voice > > network (clients + server) sit on their own dedicated 100Mb > > switches. Stats from the switch say its lightly loaded. > > > > I’ve turned on voicefile recording. What we hear, when there is a > > bad call, is stuttered speech, from BOTH sides (so local SIP client, > > and remote IAX inbound call). > > Debug from asterisk just shows the call inbound, answered and then > > hung up as per normal. > > > > I’m at a loss of how to debug the voice issue further, without > > putting a wireshark PC on the switch, port-mirroring the server and > > then capturing all of the traffic in a round-robin-type capture and > > even then I’m not sure what that will achieve. > > > > I’m going to switch from IAX to SIP for the inbound calls for that > > user and see if that helps. > > > > Any ideas welcome, > > > > What internet connection do you have... > ___ > > Physical or virtualised server ? Best Regards, -- SplatNIX IT Services :: Innovation through collaboration ___ -- 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] Call quality - how to debug
On 2 Jun 2009, at 14:14, Adrian Marsh wrote: > Hi All, > > I’ve a 1.4.15 A*k server supporting several users (approx 80 total, > but <10 sim calls usually). I’ve one user who complains of > intermittent bad calls, though I suspect the bad calls are across > the board, but intermittent. > > Inbound calls are via in IAX trunk from Gradwell. CPU stats say that > Asterisk never uses more than 4-5% cpu, systems idle besides that. > Memory seems ok too. Network utilisation is < 300kbps. The voice > network (clients + server) sit on their own dedicated 100Mb > switches. Stats from the switch say its lightly loaded. > > I’ve turned on voicefile recording. What we hear, when there is a > bad call, is stuttered speech, from BOTH sides (so local SIP client, > and remote IAX inbound call). > Debug from asterisk just shows the call inbound, answered and then > hung up as per normal. > > I’m at a loss of how to debug the voice issue further, without > putting a wireshark PC on the switch, port-mirroring the server and > then capturing all of the traffic in a round-robin-type capture and > even then I’m not sure what that will achieve. > > I’m going to switch from IAX to SIP for the inbound calls for that > user and see if that helps. > > Any ideas welcome, > What internet connection do you have... ___ -- 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] Call quality issue across VPN-> POTS vs SIP
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: > It's conceivable, but how would I verify this and how would I change > it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some disallow and allow statements. If they are there, that will tell Asterisk what codecs to use on that trunk. 2) You could also check the codec that is in use during a call by looking at the sip channel. From the asterisk CLI, start with "show channel SIP/" and tab it out to complete the command showing the trunk between your two systems. I believe the codecs are listed here as "NativeFormats" and "ReadFormat". You could check this under both of your scenarios to see if there is a different codec in use. 3) If you'd like to try and force the use of a compressed codec such as GSM between your two sites, you would just need to make sure that both sides had the following lines in the definition for the trunk in sip.conf and then do a 'reload chan_sip.so" from the Asterisk CLI: disallow=all allow=gsm ___ -- 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] Call quality issue across VPN-> POTS vs SIP
Just as an interesting follow-up/additional information, if I place a call to Site 2 on a POTS line, someone at Site 2 answers the call (using one of the Cisco phones) and then transfers it to me across the VPN the call sounds fine. So I think Bob's question was on the right track with it being a CODEC issue, but I'm not sure how I need to deal with that for the ZAP channel type. Thanks again, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lincoln King-Cliby Sent: Monday, November 03, 2008 1:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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 ___ -- 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] Call quality issue across VPN-> POTS vs SIP
Bob, It's conceivable, but how would I verify this and how would I change it if that was the problem? The site that those calls terminate at is using an Asterisk Appliance so most of the config is "done for you" but it is possible to tweak the underlying configuration files (and I also have SSH access so I can do asterisk -v -r) -- If I know what I need to tweak. Thanks, Lincoln -- Lincoln King-Cliby, CTS Applications Engineer ControlWorks Consulting, LLC Crestron Authorized Independent Programmer -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob Pierce Sent: Monday, November 03, 2008 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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] Call quality issue across VPN-> POTS vs SIP
On Mon, 2008-11-03 at 11:14 -0500, Lincoln King-Cliby wrote: > Any ideas why the audio quality would be so markedly different when > the only thing that seems to be different is where the call is > originating from (POTS line vs. SIP phone)? Is it possible that calls from your POTS line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- 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] Call quality
Hello, this is the case. Idle goes to 0% and IRQ goes to 100%. I have a Junghanns ISDN card (bristuff) card. And I guess it is using that Echo Canceler. Best regards, Loic Didelot. On Thu, 2008-07-03 at 14:52 +1200, Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Loic Didelot wrote: > > Hi, > > I am using g711a everywhere. > > > > I checked on a completely idle system (no calls at all) and idle CPU is > > dropping from 100% to 0% more than once per minute. > > If you run top, and the idle goes to 0% is it the IRQ that is using the > other 100%? > > If so, what echo canceller are you using? > > - -- > Kind Regards, > > Matt Riddell > Director > ___ > > http://www.venturevoip.com (Great new VoIP end to end solution) > http://www.venturevoip.com/news.php (Daily Asterisk News - html) > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.7 (MingW32) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFIbD7dDQNt8rg0Kp4RAql0AJ9hUDFqaNbliJTCLiKvR9BT+rbdNwCgqUjh > tZxWREnPeYuO5h1PgrXxv30= > =pPAe > -END PGP SIGNATURE- > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Loic Didelot wrote: > Hi, > I am using g711a everywhere. > > I checked on a completely idle system (no calls at all) and idle CPU is > dropping from 100% to 0% more than once per minute. If you run top, and the idle goes to 0% is it the IRQ that is using the other 100%? If so, what echo canceller are you using? - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFIbD7dDQNt8rg0Kp4RAql0AJ9hUDFqaNbliJTCLiKvR9BT+rbdNwCgqUjh tZxWREnPeYuO5h1PgrXxv30= =pPAe -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
2008/7/2 Loic Didelot <[EMAIL PROTECTED]>: > Depends on the phone. > > On many devices you can setup buttons to call a url. Thats what I did. Ah, yes. Would be a good thing to implement here. Then you can do anything, like a support ticket etc. Cheers. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hi its me again. Here is the output of zttest of a completely idle system (no calls). Acoording to some documents those values do not seem to be good. The IRQ of my zaptel card is shared with other devices. But not sure if this causes a problem. lspci -v | grep "IRQ 22" -B4 00:0c.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller [HFC-8S] (rev 01) Subsystem: Cologne Chip Designs GmbH Unknown device b55b Flags: medium devsel, IRQ 22 00:0f.0 IDE interface: VIA Technologies, Inc. VIA VT6420 SATA RAID Controller (rev 80) (prog-if 8f [Master SecP SecO PriP PriO]) Subsystem: VIA Technologies, Inc. VIA VT6420 SATA RAID Controller Flags: bus master, medium devsel, latency 32, IRQ 22 00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT82C586/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE Flags: bus master, medium devsel, latency 32, IRQ 22 Opened pseudo zap interface, measuring accuracy... 99.989845% 99.979881% 99.987305% 99.987297% 99.988190% 99.986824% 99.987999% 99.987701% 99.984970% 99.987892% 99.987587% 99.987595% 99.987885% 99.988968% 99.987885% 99.989449% 99.987595% 99.989250% 99.988571% 99.987106% 99.990044% 99.990921% 99.986519% 99.990822% 99.978127% 99.985054% 99.984482% 99.963478% 99.978722% 99.950005% 99.974609% 99.955170% 99.969528% 99.967972% 99.964066% 99.979797% 99.962898% 99.976852% 99.980072% 99.972946% 99.989937% 99.972359% 99.986908% 99.987694% 99.988770% 99.993660% 99.991516% 99.992577% 99.993164% 99.992470% 99.984276% 99.991600% 99.983200% 99.992279% 99.979790% 99.990036% 99.981544% 99.988770% 99.981346% 99.988182% 99.988190% 99.986717% 99.991211% 99.986618% 99.986824% 99.987991% 99.988869% 99.989265% 99.987015% 99.987396% 99.987495% 99.985657% 99.987396% 99.986229% 99.987206% 99.986908% 99.986618% 99.987411% 99.988579% 99.989059% 99.987106% 99.986336% 99.987114% 99.988190% 99.983200% 99.958191% 99.986031% 99.989357% 99.985939% 99.988678% 99.989746% 99.990341% 99.988762% 99.989159% 99.976067% 99.991798% 99.962799% 99.976173% 99.972366% 99.962898% 99.972855% 99.951462% 99.983986% 99.952049% 99.985733% 99.963776% 99.977440% 99.980186% 99.973915% 99.977333% 99.990341% 99.969032% 99.995110% 99.988770% 99.989555% 99.991211% 99.992386% 99.990929% 99.992294% 99.991119% 99.991997% 99.992088% 99.980865% 99.988670% 99.982712% 99.989059% 99.981934% 99.982903% 99.981850% 99.989845% 99.981628% 99.989258% 99.872566% 99.988678% --- Results after 134 passes --- Best: 99.995 -- Worst: 99.873 -- Average: 99.982824, Difference: 99.983075 Loic On Wed, 2008-07-02 at 01:36 +0200, Loic Didelot wrote: > Hi, > I am using g711a everywhere. > > I checked on a completely idle system (no calls at all) and idle CPU is > dropping from 100% to 0% more than once per minute. > > procs ---memory-- ---swap-- -io -system-- cpu > r b swpd free buff cache si sobibo in cs us sy id wa > 1 0 0 891124 4644 4286800 028 4047 85757 0 97 3 0 > 0 0 0 891124 4644 4287600 0 0 4042 68342 0 94 6 0 > 0 0 0 891124 4644 4287600 0 0 4042 72429 0 97 3 0 > 0 0 0 891124 4644 4287600 0 0 4065 158878 0 100 0 > 0 > 0 0 0 891124 4644 4287600 0 0 4033 59033 0 98 2 0 > 0 0 0 891124 4644 4287600 0 0 4012 14464 0 96 4 0 > 0 0 0 891124 4652 4286800 076 4013 19727 0 37 62 1 > 0 0 0 891124 4652 4287600 0 0 4011 20225 0 4 96 0 > 0 0 0 891124 4652 4287600 0 0 4011 23901 0 20 80 0 > 0 1 0 891124 4652 4287600 0 4 4025 21165 0 40 55 5 > 0 0 0 891124 4660 4287600 032 4028 20190 0 1 95 4 > 0 0 0 891124 4660 4287600 0 0 4022 23295 0 0 100 > 0 > 0 0 0 891124 4660 4287600 0 0 4111 20508 0 0 100 > 0 > 0 0 0 891124 4660 4287600 0 0 4102 25239 0 30 70 0 > 0 0 0 891124 4660 4287600 0 0 4112 23148 0 0 100 > 0 > 0 0 0 891124 4668 4286800 052 4116 19031 0 0 100 > 0 > 1 0 0 891124 4668 4287600 0 0 4110 21776 0 0 100 > 0 > 0 0 0 891124 4668 4287600 0 0 4150 20332 0 0 100 > 0 > 0 0 0 891124 4668 4287600 0 0 4114 26285 0 0 100 > 0 > 0 0 0 891124 4668 4287600 032 4118 23029 1 0 99 0 > 0 0 0 891124 4668 4287600 0 0 4121 23284 0 0 100 > 0 > 0 0 0 891124 4676 4286800 060 4112 25232 0 36 64 0 > 0 0 0 891124 4676 4287600 0 0 4134 21583 0 99 1 0 >
Re: [asterisk-users] Call quality
Hi, I am using g711a everywhere. I checked on a completely idle system (no calls at all) and idle CPU is dropping from 100% to 0% more than once per minute. procs ---memory-- ---swap-- -io -system-- cpu r b swpd free buff cache si sobibo in cs us sy id wa 1 0 0 891124 4644 4286800 028 4047 85757 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4042 68342 0 94 6 0 0 0 0 891124 4644 4287600 0 0 4042 72429 0 97 3 0 0 0 0 891124 4644 4287600 0 0 4065 158878 0 100 0 0 0 0 0 891124 4644 4287600 0 0 4033 59033 0 98 2 0 0 0 0 891124 4644 4287600 0 0 4012 14464 0 96 4 0 0 0 0 891124 4652 4286800 076 4013 19727 0 37 62 1 0 0 0 891124 4652 4287600 0 0 4011 20225 0 4 96 0 0 0 0 891124 4652 4287600 0 0 4011 23901 0 20 80 0 0 1 0 891124 4652 4287600 0 4 4025 21165 0 40 55 5 0 0 0 891124 4660 4287600 032 4028 20190 0 1 95 4 0 0 0 891124 4660 4287600 0 0 4022 23295 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4111 20508 0 0 100 0 0 0 0 891124 4660 4287600 0 0 4102 25239 0 30 70 0 0 0 0 891124 4660 4287600 0 0 4112 23148 0 0 100 0 0 0 0 891124 4668 4286800 052 4116 19031 0 0 100 0 1 0 0 891124 4668 4287600 0 0 4110 21776 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4150 20332 0 0 100 0 0 0 0 891124 4668 4287600 0 0 4114 26285 0 0 100 0 0 0 0 891124 4668 4287600 032 4118 23029 1 0 99 0 0 0 0 891124 4668 4287600 0 0 4121 23284 0 0 100 0 0 0 0 891124 4676 4286800 060 4112 25232 0 36 64 0 0 0 0 891124 4676 4287600 0 0 4134 21583 0 99 1 0 0 0 0 891124 4676 4287600 0 0 4105 26029 0 100 0 0 0 0 0 891124 4676 4287600 076 4143 22795 0 25 75 0 0 0 0 891124 4676 4287600 0 0 4118 21418 0 0 54 46 0 0 0 891124 4676 4287600 0 0 4108 25499 0 0 100 0 0 0 0 891124 4684 4286800 052 4081 20778 0 0 100 0 0 0 0 891124 4684 4287600 0 0 4011 25463 0 13 87 0 0 0 0 891124 4684 4287600 0 0 4021 23502 0 86 14 0 0 0 0 891124 4684 4287600 0 0 4015 21693 0 1 99 0 On Tue, 2008-07-01 at 22:28 +0300, Tzafrir Cohen wrote: > On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote: > > Run top along with the tool that indicated the high I/O and see what > > is going on. Are you doing G729 or anything like that? > > vmstat will probably provide more useful data (vmstat 1 etc. for a > continous run). > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Depends on the phone. On many devices you can setup buttons to call a url. Thats what I did. Loic On Tue, 2008-07-01 at 21:19 +0100, Gavin Henry wrote: > What did you do to setup a button for alerts? > > Thanks. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
What did you do to setup a button for alerts? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 03:22:07PM -0400, Steve Totaro wrote: > Run top along with the tool that indicated the high I/O and see what > is going on. Are you doing G729 or anything like that? vmstat will probably provide more useful data (vmstat 1 etc. for a continous run). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Run top along with the tool that indicated the high I/O and see what is going on. Are you doing G729 or anything like that? Thanks, Steve T On Tue, Jul 1, 2008 at 3:06 PM, Loic Didelot <[EMAIL PROTECTED]> wrote: > Yes, > but they get like 10 voicemails per day. That feature isnt really used > alot. > > Loic > > On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote: >> I don't think your issue is the VIA CPU but the I/O of your flash >> drive. Voicemail is what I suspect being the I/O bottleneck. >> >> Thanks, >> Steve T >> >> On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot <[EMAIL PROTECTED]> wrote: >> > Yes, most calls are SIP-PSTN calls. >> > >> > Thanks for your help. >> > >> > I will try a faster box. Are VIA CPUs known to cause problems? >> > >> > Loic >> > >> > >> > On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: >> >> On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: >> >> > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% >> >> > of all alerts are internal calls. >> >> >> >> Any chance that omst of the calls are outgoing SIP->PSTN calls? >> >> >> >> > I had the chance to notice the problem >> >> > once myself but I could never again reproduce. >> >> >> >> So it doesn't happen with Local->PSTN calls (the type you can easily >> >> test remotely if we assume there's no voip access). >> >> >> > >> > >> > ___ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > >> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> > Register Now: http://www.astricon.net >> > >> > 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 -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> 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 -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Yes, but they get like 10 voicemails per day. That feature isnt really used alot. Loic On Tue, 2008-07-01 at 14:58 -0400, Steve Totaro wrote: > I don't think your issue is the VIA CPU but the I/O of your flash > drive. Voicemail is what I suspect being the I/O bottleneck. > > Thanks, > Steve T > > On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot <[EMAIL PROTECTED]> wrote: > > Yes, most calls are SIP-PSTN calls. > > > > Thanks for your help. > > > > I will try a faster box. Are VIA CPUs known to cause problems? > > > > Loic > > > > > > On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: > >> On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: > >> > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% > >> > of all alerts are internal calls. > >> > >> Any chance that omst of the calls are outgoing SIP->PSTN calls? > >> > >> > I had the chance to notice the problem > >> > once myself but I could never again reproduce. > >> > >> So it doesn't happen with Local->PSTN calls (the type you can easily > >> test remotely if we assume there's no voip access). > >> > > > > > > ___ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > 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 -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I don't think your issue is the VIA CPU but the I/O of your flash drive. Voicemail is what I suspect being the I/O bottleneck. Thanks, Steve T On Tue, Jul 1, 2008 at 2:52 PM, Loic Didelot <[EMAIL PROTECTED]> wrote: > Yes, most calls are SIP-PSTN calls. > > Thanks for your help. > > I will try a faster box. Are VIA CPUs known to cause problems? > > Loic > > > On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: >> On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: >> > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% >> > of all alerts are internal calls. >> >> Any chance that omst of the calls are outgoing SIP->PSTN calls? >> >> > I had the chance to notice the problem >> > once myself but I could never again reproduce. >> >> So it doesn't happen with Local->PSTN calls (the type you can easily >> test remotely if we assume there's no voip access). >> > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Yes, most calls are SIP-PSTN calls. Thanks for your help. I will try a faster box. Are VIA CPUs known to cause problems? Loic On Tue, 2008-07-01 at 18:50 +0300, Tzafrir Cohen wrote: > On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: > > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% > > of all alerts are internal calls. > > Any chance that omst of the calls are outgoing SIP->PSTN calls? > > > I had the chance to notice the problem > > once myself but I could never again reproduce. > > So it doesn't happen with Local->PSTN calls (the type you can easily > test remotely if we assume there's no voip access). > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 05:17:50PM +0200, Loic Didelot wrote: > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% > of all alerts are internal calls. Any chance that omst of the calls are outgoing SIP->PSTN calls? > I had the chance to notice the problem > once myself but I could never again reproduce. So it doesn't happen with Local->PSTN calls (the type you can easily test remotely if we assume there's no voip access). -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Try IOSTAT http://www.linuxquestions.org/linux/articles/Jeremys_Magazine_Articles/Hunting_I_O_Bottlenecks_with_iostat Maybe you can correlate VM and/or emailing of VM to your IO spikes. Have you watched top and the Asterisk CLI when someone hits the panic button? Thanks, Steve T On Tue, Jul 1, 2008 at 11:17 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% > of all alerts are internal calls. I had the chance to notice the problem > once myself but I could never again reproduce. > > Best regards, > Loic Didelot. > > On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote: >> On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: >> > Hello, >> > one of my customers complained about bad voice quality on several calls, >> > so I programmed a button on each phone which users can hit if they have >> > audio drops and echo. >> > >> > I did this to check if there is a common recurrent problem to a given >> > destination or just for one user etc... But till now I could not detect >> > a pattern which could explain the problems >> > >> > This "alert button" is pressed between 7%-10% of all calls. The customer >> > has 25 phones and around 300 calls per day. >> > >> > The SNOM phones are connected to Linksys switches and are totaly split >> > from the computers network. The same goes for the asterisk box. No calls >> > are routed trough the internet. >> > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier >> >> Are the problems in SIP->PSTN calls? SIP->SIP calls? >> PSTN->Local? (echo test, playback, whatever) >> >> SIP->PSTN or PSTN->SIP (what direction is the call)? >> >> 7% is something you have hope of reproducing. Unless you miss the real >> factor. Have you managed to reproduce it yourself? >> >> > >> > The carrier we use is known for his good quality and we never had a >> > problem. It is the historic and most expensive carrier in Luxembourg. >> > >> > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a >> > maximum of 6 concurrent calls. >> > >> > Maybe someone can help me to track down the problem. What should I >> > check, monitor test. Any ideas are welcome. >> > -- > Loïc DIDELOT > MIXvoip S.a. > [EMAIL PROTECTED] > http://www.mixvoip.com > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
The problem appears mostly on outgoing calls SIP-PSTN but not only. 10% of all alerts are internal calls. I had the chance to notice the problem once myself but I could never again reproduce. Best regards, Loic Didelot. On Tue, 2008-07-01 at 18:03 +0300, Tzafrir Cohen wrote: > On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: > > Hello, > > one of my customers complained about bad voice quality on several calls, > > so I programmed a button on each phone which users can hit if they have > > audio drops and echo. > > > > I did this to check if there is a common recurrent problem to a given > > destination or just for one user etc... But till now I could not detect > > a pattern which could explain the problems > > > > This "alert button" is pressed between 7%-10% of all calls. The customer > > has 25 phones and around 300 calls per day. > > > > The SNOM phones are connected to Linksys switches and are totaly split > > from the computers network. The same goes for the asterisk box. No calls > > are routed trough the internet. > > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier > > Are the problems in SIP->PSTN calls? SIP->SIP calls? > PSTN->Local? (echo test, playback, whatever) > > SIP->PSTN or PSTN->SIP (what direction is the call)? > > 7% is something you have hope of reproducing. Unless you miss the real > factor. Have you managed to reproduce it yourself? > > > > > The carrier we use is known for his good quality and we never had a > > problem. It is the historic and most expensive carrier in Luxembourg. > > > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > > maximum of 6 concurrent calls. > > > > Maybe someone can help me to track down the problem. What should I > > check, monitor test. Any ideas are welcome. > -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 01, 2008 at 02:38:25PM +0200, Loic Didelot wrote: > Hello, > one of my customers complained about bad voice quality on several calls, > so I programmed a button on each phone which users can hit if they have > audio drops and echo. > > I did this to check if there is a common recurrent problem to a given > destination or just for one user etc... But till now I could not detect > a pattern which could explain the problems > > This "alert button" is pressed between 7%-10% of all calls. The customer > has 25 phones and around 300 calls per day. > > The SNOM phones are connected to Linksys switches and are totaly split > from the computers network. The same goes for the asterisk box. No calls > are routed trough the internet. > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier Are the problems in SIP->PSTN calls? SIP->SIP calls? PSTN->Local? (echo test, playback, whatever) SIP->PSTN or PSTN->SIP (what direction is the call)? 7% is something you have hope of reproducing. Unless you miss the real factor. Have you managed to reproduce it yourself? > > The carrier we use is known for his good quality and we never had a > problem. It is the historic and most expensive carrier in Luxembourg. > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > maximum of 6 concurrent calls. > > Maybe someone can help me to track down the problem. What should I > check, monitor test. Any ideas are welcome. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Recording the calls may or may not reveal an issue. I have personally done this exact same method of troubleshooting only to find the recordings were perfect but not the actual calls. I think you should try just putting a regular server in place of your "appliance" and then test. I have a feeling the I/O is choking your system, similar to recording many simultaneous calls, which to me would indicate a flash bottleneck. At least put in a real HD and copy over your configs. Thanks, Steve T On Tue, Jul 1, 2008 at 9:15 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > I considered that, > but I fear that this would load the machine even more. So I guess I > should take a more powerful box with a good harddrive (at the moment I > have a solid state flash card) and start recording calls. > > > Best regards, > Loic Didelot. > > > > On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote: >> On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: >> > Maybe someone can help me to track down the problem. What should I >> > check, monitor test. Any ideas are welcome. >> >> If there are no legal reasons not to, consider recording all calls for >> a limited time. It's easier for engineers to debug a voice quality >> problem when they have a recording of exactly what it sounds like. >> It's possible that different people are complaining about different >> perceptions of what they consider a voice quality problem, and that >> the problem might not even be on your end of the conversation. >> >> ___ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >>http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Loïc DIDELOT > MIXvoip S.a. > [EMAIL PROTECTED] > http://www.mixvoip.com > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I considered that, but I fear that this would load the machine even more. So I guess I should take a more powerful box with a good harddrive (at the moment I have a solid state flash card) and start recording calls. Best regards, Loic Didelot. On Tue, 2008-07-01 at 09:10 -0400, David Backeberg wrote: > On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > > Maybe someone can help me to track down the problem. What should I > > check, monitor test. Any ideas are welcome. > > If there are no legal reasons not to, consider recording all calls for > a limited time. It's easier for engineers to debug a voice quality > problem when they have a recording of exactly what it sounds like. > It's possible that different people are complaining about different > perceptions of what they consider a voice quality problem, and that > the problem might not even be on your end of the conversation. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hello, I forgot to include CPU information [EMAIL PROTECTED]:/usr/src/bristuff-0.4.0-RC2# cat /proc/cpuinfo processor : 0 vendor_id : CentaurHauls cpu family : 6 model : 10 model name : VIA Esther processor 1000MHz stepping: 9 cpu MHz : 1000.127 cache size : 128 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge cmov pat clflush acpi mmx fxsr sse sse2 tm nx up pni est tm2 rng rng_en ace ace_en ace2 ace2_en phe phe_en pmm pmm_en bogomips: 2002.19 clflush size: 64 The box is running asterisk, nothing else: - asterisk - postfix just to send out voicemails - no realtime - som AGIS at call setup and call end - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2 - zaptel-1.4.10 Best regards, Loic Didelot. On Tue, 2008-07-01 at 13:54 +0100, Steve Davies wrote: > 2008/7/1 Loic Didelot <[EMAIL PROTECTED]>: > > Hello, > > one of my customers complained about bad voice quality on several calls, > > so I programmed a button on each phone which users can hit if they have > > audio drops and echo. > > > > I did this to check if there is a common recurrent problem to a given > > destination or just for one user etc... But till now I could not detect > > a pattern which could explain the problems > > > > This "alert button" is pressed between 7%-10% of all calls. The customer > > has 25 phones and around 300 calls per day. > > > > The SNOM phones are connected to Linksys switches and are totaly split > > from the computers network. The same goes for the asterisk box. No calls > > are routed trough the internet. > > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier > > > > The carrier we use is known for his good quality and we never had a > > problem. It is the historic and most expensive carrier in Luxembourg. > > > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > > maximum of 6 concurrent calls. > > > > Which version of asterisk/zaptel, and which echo canceler is running in > Zaptel? > > Regards, > Steve > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > Maybe someone can help me to track down the problem. What should I > check, monitor test. Any ideas are welcome. If there are no legal reasons not to, consider recording all calls for a limited time. It's easier for engineers to debug a voice quality problem when they have a recording of exactly what it sounds like. It's possible that different people are complaining about different perceptions of what they consider a voice quality problem, and that the problem might not even be on your end of the conversation. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
Hi, its a new installation in a new office. Customer moved in, so right moment to get a new PBX. The box is running asterisk, nothing else: - asterisk - postfix just to send out voicemails - no realtime - som AGIS at call setup and call end - Asterisk 1.4.19.1-BRIstuffed-0.4.0-RC2 - zaptel-1.4.10 We use a Junghann BRI card and a XORCOMM Analog Astribank. But only one modem and 2 fax devices are connected to the astribank. I did not do a debug on the spans. Anythin special I should look for? Difficult to describe the audio: - basically echo is appearing - audio problems are only one way - audio has cuts when speaking Best regards, Loic Didelot. On Tue, 2008-07-01 at 08:58 -0400, Steve Totaro wrote: > On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > > Hello, > > one of my customers complained about bad voice quality on several calls, > > so I programmed a button on each phone which users can hit if they have > > audio drops and echo. > > > > I did this to check if there is a common recurrent problem to a given > > destination or just for one user etc... But till now I could not detect > > a pattern which could explain the problems > > > > This "alert button" is pressed between 7%-10% of all calls. The customer > > has 25 phones and around 300 calls per day. > > > > The SNOM phones are connected to Linksys switches and are totaly split > > from the computers network. The same goes for the asterisk box. No calls > > are routed trough the internet. > > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier > > > > The carrier we use is known for his good quality and we never had a > > problem. It is the historic and most expensive carrier in Luxembourg. > > > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > > maximum of 6 concurrent calls. > > > > Maybe someone can help me to track down the problem. What should I > > check, monitor test. Any ideas are welcome. > > > > > > Best regards, > > Loic Didelot. > > > > -- > > Loïc DIDELOT > > MIXvoip S.a. > > [EMAIL PROTECTED] > > http://www.mixvoip.com > > Is this a new install or a new problem? > > If it is a new problem, what has changed? > > If it is a new install, I would not rule out the provider, the more > "historic" may or may not be a good thing. Describe the audio when it > is poor, popping, clicking, hissing? > > Have you tried running a debug on the spans? > > Thanks, > Steve T > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Loïc DIDELOT MIXvoip S.a. [EMAIL PROTECTED] http://www.mixvoip.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I/O wait is very suspicious. What is your hardware platform? Is this just a plain Jane PBX or are you doing anything unusual? Thanks, Steve T On Tue, Jul 1, 2008 at 8:57 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > I tried to get a little into cpu utilization and found the following > results. > > Can they help me to come to a conclusion? > > Best regards, > Loic Didelot. > > [EMAIL PROTECTED]:~# mpstat 1 > Linux 2.6.22-14-server (ppsite1)07/01/2008 > > 02:54:40 PM CPU %user %nice%sys %iowait%irq %soft %steal > %idleintr/s > 02:54:41 PM all0.000.001.000.000.000.000.00 > 99.00 4210.00 > 02:54:42 PM all0.000.000.000.000.000.000.00 > 100.00 4207.00 > 02:54:43 PM all0.000.000.000.000.000.000.00 > 100.00 4208.00 > 02:54:44 PM all0.000.000.000.00 45.000.000.00 > 55.00 4127.00 > 02:54:45 PM all0.000.000.000.00 97.000.000.00 > 3.00 4148.00 > 02:54:46 PM all0.000.000.000.00 93.004.000.00 > 3.00 4195.00 > 02:54:47 PM all0.000.000.000.00 92.006.000.00 > 2.00 4175.00 > 02:54:48 PM all0.000.000.001.00 91.002.000.00 > 6.00 4154.00 > 02:54:49 PM all0.000.000.000.00 100.000.000.00 > 0.00 4069.00 > 02:54:50 PM all0.000.000.000.00 23.000.000.00 > 77.00 4125.00 > 02:54:51 PM all2.000.000.000.00 19.000.000.00 > 79.00 4123.00 > 02:54:52 PM all0.000.000.000.000.000.000.00 > 100.00 4236.00 > 02:54:53 PM all0.000.006.002.000.003.000.00 > 89.00 4302.00 > 02:54:54 PM all0.000.005.000.000.003.000.00 > 92.00 4267.00 > 02:54:55 PM all0.000.00 20.000.000.001.000.00 > 79.00 4328.00 > 02:54:56 PM all0.000.000.000.009.000.000.00 > 91.00 4352.00 > 02:54:57 PM all0.000.000.00 49.00 46.000.000.00 > 5.00 4376.00 > 02:54:58 PM all0.000.000.000.000.000.000.00 > 100.00 4350.00 > 02:54:59 PM all 11.000.002.00 36.000.000.000.00 > 51.00 4237.00 > 02:55:00 PM all0.000.000.00 100.000.000.000.00 > 0.00 4221.00 > 02:55:01 PM all1.000.001.00 62.00 36.000.000.00 > 0.00 4318.00 > 02:55:02 PM all0.000.000.002.00 98.000.000.00 > 0.00 4219.00 > 02:55:03 PM all0.000.000.000.00 100.000.000.00 > 0.00 4342.00 > 02:55:04 PM all0.000.000.002.00 98.000.000.00 > 0.00 4236.00 > 02:55:05 PM all 14.000.004.00 20.00 62.000.000.00 > 0.00 4229.00 > 02:55:06 PM all 39.000.003.00 38.00 18.001.000.00 > 1.00 4346.00 > 02:55:07 PM all8.000.008.00 79.003.001.000.00 > 1.00 4240.00 > 02:55:08 PM all1.000.000.00 98.000.000.000.00 > 1.00 4217.00 > 02:55:09 PM all0.000.001.006.000.000.000.00 > 93.00 4167.00 > 02:55:10 PM all0.000.000.000.00 25.000.000.00 > 75.00 4132.00 > 02:55:11 PM all0.000.000.000.00 75.000.000.00 > 25.00 4117.00 > 02:55:12 PM all0.000.000.000.00 53.000.000.00 > 47.00 4130.00 > 02:55:13 PM all0.000.000.000.000.000.000.00 > 100.00 4103.00 > 02:55:14 PM all0.000.000.00 50.000.000.000.00 > 50.00 4124.00 > 02:55:15 PM all0.000.001.000.000.000.000.00 > 99.00 4216.00 > 02:55:16 PM all1.000.000.000.00 32.000.000.00 > 67.00 4214.00 > 02:55:17 PM all0.000.000.000.00 98.000.000.00 > 2.00 4209.00 > 02:55:18 PM all0.000.000.000.00 94.000.000.00 > 6.00 4220.00 > 02:55:19 PM all0.000.000.000.00 58.000.000.00 > 42.00 4216.00 > 02:55:20 PM all1.000.000.000.000.000.000.00 > 99.00 4204.00 > 02:55:21 PM all1.000.000.000.000.000.000.00 > 99.00 4210.00 > 02:55:22 PM all1.000.000.000.000.000.000.00 > 99.00 4234.00 > 02:55:23 PM all0.000.001.000.000.000.000.00 > 99.00 4202.00 > 02:55:24 PM all0.000.001.000.000.000.000.00 > 99.00 4109.00 > 02:55:25 PM all1.000.001.001.00 53.001.000.00 > 43.00 4179.00 > 02:55:26 PM all0.000.000.000.00 35.000.000.00 > 65.00 4213.00 > 02:55:27 PM all
Re: [asterisk-users] Call quality
On Tue, Jul 1, 2008 at 8:38 AM, Loic Didelot <[EMAIL PROTECTED]> wrote: > Hello, > one of my customers complained about bad voice quality on several calls, > so I programmed a button on each phone which users can hit if they have > audio drops and echo. > > I did this to check if there is a common recurrent problem to a given > destination or just for one user etc... But till now I could not detect > a pattern which could explain the problems > > This "alert button" is pressed between 7%-10% of all calls. The customer > has 25 phones and around 300 calls per day. > > The SNOM phones are connected to Linksys switches and are totaly split > from the computers network. The same goes for the asterisk box. No calls > are routed trough the internet. > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier > > The carrier we use is known for his good quality and we never had a > problem. It is the historic and most expensive carrier in Luxembourg. > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > maximum of 6 concurrent calls. > > Maybe someone can help me to track down the problem. What should I > check, monitor test. Any ideas are welcome. > > > Best regards, > Loic Didelot. > > -- > Loïc DIDELOT > MIXvoip S.a. > [EMAIL PROTECTED] > http://www.mixvoip.com Is this a new install or a new problem? If it is a new problem, what has changed? If it is a new install, I would not rule out the provider, the more "historic" may or may not be a good thing. Describe the audio when it is poor, popping, clicking, hissing? Have you tried running a debug on the spans? Thanks, Steve T ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call quality
I tried to get a little into cpu utilization and found the following results. Can they help me to come to a conclusion? Best regards, Loic Didelot. [EMAIL PROTECTED]:~# mpstat 1 Linux 2.6.22-14-server (ppsite1)07/01/2008 02:54:40 PM CPU %user %nice%sys %iowait%irq %soft %steal %idleintr/s 02:54:41 PM all0.000.001.000.000.000.000.00 99.00 4210.00 02:54:42 PM all0.000.000.000.000.000.000.00 100.00 4207.00 02:54:43 PM all0.000.000.000.000.000.000.00 100.00 4208.00 02:54:44 PM all0.000.000.000.00 45.000.000.00 55.00 4127.00 02:54:45 PM all0.000.000.000.00 97.000.000.00 3.00 4148.00 02:54:46 PM all0.000.000.000.00 93.004.000.00 3.00 4195.00 02:54:47 PM all0.000.000.000.00 92.006.000.00 2.00 4175.00 02:54:48 PM all0.000.000.001.00 91.002.000.00 6.00 4154.00 02:54:49 PM all0.000.000.000.00 100.000.000.00 0.00 4069.00 02:54:50 PM all0.000.000.000.00 23.000.000.00 77.00 4125.00 02:54:51 PM all2.000.000.000.00 19.000.000.00 79.00 4123.00 02:54:52 PM all0.000.000.000.000.000.000.00 100.00 4236.00 02:54:53 PM all0.000.006.002.000.003.000.00 89.00 4302.00 02:54:54 PM all0.000.005.000.000.003.000.00 92.00 4267.00 02:54:55 PM all0.000.00 20.000.000.001.000.00 79.00 4328.00 02:54:56 PM all0.000.000.000.009.000.000.00 91.00 4352.00 02:54:57 PM all0.000.000.00 49.00 46.000.000.00 5.00 4376.00 02:54:58 PM all0.000.000.000.000.000.000.00 100.00 4350.00 02:54:59 PM all 11.000.002.00 36.000.000.000.00 51.00 4237.00 02:55:00 PM all0.000.000.00 100.000.000.000.00 0.00 4221.00 02:55:01 PM all1.000.001.00 62.00 36.000.000.00 0.00 4318.00 02:55:02 PM all0.000.000.002.00 98.000.000.00 0.00 4219.00 02:55:03 PM all0.000.000.000.00 100.000.000.00 0.00 4342.00 02:55:04 PM all0.000.000.002.00 98.000.000.00 0.00 4236.00 02:55:05 PM all 14.000.004.00 20.00 62.000.000.00 0.00 4229.00 02:55:06 PM all 39.000.003.00 38.00 18.001.000.00 1.00 4346.00 02:55:07 PM all8.000.008.00 79.003.001.000.00 1.00 4240.00 02:55:08 PM all1.000.000.00 98.000.000.000.00 1.00 4217.00 02:55:09 PM all0.000.001.006.000.000.000.00 93.00 4167.00 02:55:10 PM all0.000.000.000.00 25.000.000.00 75.00 4132.00 02:55:11 PM all0.000.000.000.00 75.000.000.00 25.00 4117.00 02:55:12 PM all0.000.000.000.00 53.000.000.00 47.00 4130.00 02:55:13 PM all0.000.000.000.000.000.000.00 100.00 4103.00 02:55:14 PM all0.000.000.00 50.000.000.000.00 50.00 4124.00 02:55:15 PM all0.000.001.000.000.000.000.00 99.00 4216.00 02:55:16 PM all1.000.000.000.00 32.000.000.00 67.00 4214.00 02:55:17 PM all0.000.000.000.00 98.000.000.00 2.00 4209.00 02:55:18 PM all0.000.000.000.00 94.000.000.00 6.00 4220.00 02:55:19 PM all0.000.000.000.00 58.000.000.00 42.00 4216.00 02:55:20 PM all1.000.000.000.000.000.000.00 99.00 4204.00 02:55:21 PM all1.000.000.000.000.000.000.00 99.00 4210.00 02:55:22 PM all1.000.000.000.000.000.000.00 99.00 4234.00 02:55:23 PM all0.000.001.000.000.000.000.00 99.00 4202.00 02:55:24 PM all0.000.001.000.000.000.000.00 99.00 4109.00 02:55:25 PM all1.000.001.001.00 53.001.000.00 43.00 4179.00 02:55:26 PM all0.000.000.000.00 35.000.000.00 65.00 4213.00 02:55:27 PM all0.000.001.000.000.000.000.00 99.00 4204.00 02:55:28 PM all0.000.001.000.000.000.000.00 99.00 4169.00 02:55:29 PM all0.000.003.960.00 37.620.990.00 57.43 4149.50 02:55:30 PM all0.000.001.000.000.000.000.00 99.00 4208.00 02:55:31 PM all0.000.000.00 16.003.000.000.00
Re: [asterisk-users] Call quality
2008/7/1 Loic Didelot <[EMAIL PROTECTED]>: > Hello, > one of my customers complained about bad voice quality on several calls, > so I programmed a button on each phone which users can hit if they have > audio drops and echo. > > I did this to check if there is a common recurrent problem to a given > destination or just for one user etc... But till now I could not detect > a pattern which could explain the problems > > This "alert button" is pressed between 7%-10% of all calls. The customer > has 25 phones and around 300 calls per day. > > The SNOM phones are connected to Linksys switches and are totaly split > from the computers network. The same goes for the asterisk box. No calls > are routed trough the internet. > Phone -> Local Lan -> Asterisk -> Zaptel (Junghanns BRI card) -> Carrier > > The carrier we use is known for his good quality and we never had a > problem. It is the historic and most expensive carrier in Luxembourg. > > Asterisk is running on a 1GHZ VIA CPU with 1GB RAM box. They have a > maximum of 6 concurrent calls. > Which version of asterisk/zaptel, and which echo canceler is running in Zaptel? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?
'iax2 show channels'maybe I have a feeling this is going to be one of those ugly ones where it's going to be a pain to troubleshoot... Offhand - have you tested 'trunk=yes' vs 'trunk=no'? PaulH On Wed, 2007-12-12 at 17:00 +1100, Daniel Cole wrote: > Hi Paul, > > Where abouts exactly is the best place to get these figures from? > > I have been checking iax2 show netstats, which does give some figures. > These appear not to be accurate though, as when there are multiple > inter-site calls, the result for one channel of audio can show no > jitter or latency, but another will have some jitter and latency. Or > is this a weird way for the problem to show its head? > > Thanks, > > Daniel Cole (CCNA) > > > P Please consider the environment before you print this e-mail or any > attachments. > > > > > > __ > From: Paul Hales [mailto:[EMAIL PROTECTED] > Sent: Wednesday, 12 December 2007 4:40 PM > To: Daniel Cole > Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - > Router Issue? > > > > > Hmmm..wierd > > Are you getting an weird jitter/latency figures in the CLI? > > PaulH > > > On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote: > > G729 All Around. > > Daniel Cole (CCNA) > > > > P Please consider the environment before you print this e-mail or > > any attachments. > > > > > > > > > > ________ > > > > From: Paul Hales [mailto:[EMAIL PROTECTED] > > Sent: Wednesday, 12 December 2007 4:10 PM > > To: Daniel Cole > > Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - > > Router Issue? > > > > > > > > > > What codec are you using? > > > > PaulH > > > > > > On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: > > > Hello Everyone, > > > > > > We have recently installed a pair of Trixbox servers in for a > > > client of our. They have two locations, with one server each. The > > > servers terminate 3 standard POTS lines into a Sangoma A200D card. > > > The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, > > > Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 > > > all around. > > > > > > Each site has two (2) 512k/512k ADSL connections terminating into > > > a Cisco 877W router (using an additional 'dumb' modem in a > > > separate VLAN for the extra dsl connection). Using policy based > > > routing, all Voice Data goes over one DSL connection (the one that > > > terminates directly into the router), and all other traffic (e.g. > > > Web and VPN) goes out the second connection (the bridged dumb dsl > > > modem). > > > > > > We are also the ISP for this client, and as thus we have full > > > monitoring of our Layer 2 and Layer 3 networks. From our analysis, > > > it doesn't appear that there is any issue in these networks. We > > > have other customers using the VoIP service, who have not > > > complained of these issues. > > > > > > Now for the Fun part! > > > The client is complaining of issues with inter-site calls. They > > > are reporting issues with crackly and broken speech, and horrible > > > jitter (or packet loss). This presents a huge issues, because they > > > have one receptionist answering all calls for both sites. So if a > > > call comes in from the other site, it automatically an inter-site > > > call, and the quality falls out of it. If the call is then > > > transfered back to the originating site, the audio 'bounces' > > > between the two sites, which add to the call quality degradation. > > > > > > We have been monitoring the router while these incidents have been > > > reported, and it does not appear to be a bandwidth issue. The DSL > > > tail used for Voice gets to no more then 120k in each direction > > > (we have tested the links, and can pull data at 53k/s between > > > sites). CPU usage floats at around 20-25% under load. The router > > > has only shows major packet loss (that we can tell) when REALLY > > > pushing it in testing (e.g. 10+ calls between sites). > > > We have enabled the SIP jitter buffer, as well as the IAX jitter > > > buffer, which appeared to make a huge difference, but the issue is > > > sti
Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue?
Hi Paul, Where abouts exactly is the best place to get these figures from? I have been checking iax2 show netstats, which does give some figures. These appear not to be accurate though, as when there are multiple inter-site calls, the result for one channel of audio can show no jitter or latency, but another will have some jitter and latency. Or is this a weird way for the problem to show its head? Thanks, Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:40 PM To: Daniel Cole Subject: RE: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? Hmmm..wierd Are you getting an weird jitter/latency figures in the CLI? PaulH On Wed, 2007-12-12 at 16:37 +1100, Daniel Cole wrote: G729 All Around. Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:10 PM To: Daniel Cole Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? What codec are you using? PaulH On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues With 2 Trixbox's - Router Issue?
G729 All Around. Daniel Cole (CCNA) Technical Support [http://www.hugonet.com.au/clients/hugonet.gif] Ph: 1800 424 683 Fax: 03 5221 7659 e: [EMAIL PROTECTED]<mailto:[EMAIL PROTECTED]> w: hugonet.com.au<http://www.hugonet.com.au/> --- The information transmitted is the property of HugoNet and is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Statements and opinions expressed in this e-mail may not represent those of the company. Any review, retransmission, dissemination and other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender immediately and delete the material from any computer. P Please consider the environment before you print this e-mail or any attachments. From: Paul Hales [mailto:[EMAIL PROTECTED] Sent: Wednesday, 12 December 2007 4:10 PM To: Daniel Cole Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - Router Issue? What codec are you using? PaulH On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues With 2 Trixbox's - Router Issue?
What codec are you using? PaulH On Wed, 2007-12-12 at 13:00 +1100, Daniel Cole wrote: > Hello Everyone, > > We have recently installed a pair of Trixbox servers in for a client > of our. They have two locations, with one server each. The servers > terminate 3 standard POTS lines into a Sangoma A200D card. The servers > are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon > Processors). We are using Trixbox 2.2, and G729 all around. > > Each site has two (2) 512k/512k ADSL connections terminating into a > Cisco 877W router (using an additional 'dumb' modem in a separate VLAN > for the extra dsl connection). Using policy based routing, all Voice > Data goes over one DSL connection (the one that terminates directly > into the router), and all other traffic (e.g. Web and VPN) goes out > the second connection (the bridged dumb dsl modem). > > We are also the ISP for this client, and as thus we have full > monitoring of our Layer 2 and Layer 3 networks. From our analysis, it > doesn't appear that there is any issue in these networks. We have > other customers using the VoIP service, who have not complained of > these issues. > > Now for the Fun part! > The client is complaining of issues with inter-site calls. They are > reporting issues with crackly and broken speech, and horrible jitter > (or packet loss). This presents a huge issues, because they have one > receptionist answering all calls for both sites. So if a call comes in > from the other site, it automatically an inter-site call, and the > quality falls out of it. If the call is then transfered back to the > originating site, the audio 'bounces' between the two sites, which add > to the call quality degradation. > > We have been monitoring the router while these incidents have been > reported, and it does not appear to be a bandwidth issue. The DSL tail > used for Voice gets to no more then 120k in each direction (we have > tested the links, and can pull data at 53k/s between sites). CPU usage > floats at around 20-25% under load. The router has only shows major > packet loss (that we can tell) when REALLY pushing it in testing (e.g. > 10+ calls between sites). > We have enabled the SIP jitter buffer, as well as the IAX jitter > buffer, which appeared to make a huge difference, but the issue is > still ongoing. > > These issues have also been reported with some outbound VoIP calls. > Internal calls, and calls directly in or out of the Sangoma card are > clear, with no issues reported. > > Does anyone have any thoughts on what could be causing these issues? > We have been racking our brains here, and have tried everything that > we can think of. These system is a million times better then what is > what when it was first installed, but it is still not where it should > be in terms of quality. > > Any thoughts/ideas are most welcome. > > Thank you > > > > Daniel Cole (CCNA) > > > > > P Please consider the environment before you print this e-mail or any > attachments. > > > > > ___ > --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] Call Quality Issues With 2 Trixbox's - RouterIssue?
The two boxes are labeled as per the town they are in: Leongatha and Korumburra. The receptionist is in Korumburra. When a call comes in off the PSTN in Leongatha, the first number in the call queue is the receptionist. If she answers it, then the media flow looks like this: PSTN -> Leongatha -> IAX Trunk -> Korumburra -> Receptionist Phone If she then transfers the call back to a Leongatha extension, the media path looks like this: PSTN -> Leongatha -> IAX Trunk -> Korumburra -> IAX Trunk -> Leongatha -> IP Phone I believe that it is possible to stop this 'bouncing' of the call from happening by using the re-invite feature. However, taking the Trixbox's out of the media path is undesirable, as they client needs to be able to record calls. Also, doing this does not 'fix' the underlying problem. They are also having some issues with outbound calls from Leongatha (over VoIP), and they are having no real issues at Korumburra. Many Thanks, Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, 12 December 2007 3:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue? How are the calls being transferred from Box A to Box B? On what box is the receptionist registered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Cole Sent: Tuesday, December 11, 2007 9:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue? Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues With 2 Trixbox's - Router Issue?
Do an RTP analysis with Wireshark of a sample call. That could probably narrow down the source of the problem. I would suspect you will either see some jitter or packets out of order. Daniel Cole wrote: > Hello Everyone, > > We have recently installed a pair of Trixbox servers in for a client > of our. They have two locations, with one server each. The servers > terminate 3 standard POTS lines into a Sangoma A200D card. The servers > are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon > Processors). We are using Trixbox 2.2, and G729 all around. > > Each site has two (2) 512k/512k ADSL connections terminating into a > Cisco 877W router (using an additional 'dumb' modem in a separate VLAN > for the extra dsl connection). Using policy based routing, all Voice > Data goes over one DSL connection (the one that terminates directly > into the router), and all other traffic (e.g. Web and VPN) goes out > the second connection (the bridged dumb dsl modem). > > We are also the ISP for this client, and as thus we have full > monitoring of our Layer 2 and Layer 3 networks. From our analysis, it > doesn't appear that there is any issue in these networks. We have > other customers using the VoIP service, who have not complained of > these issues. > > Now for the Fun part! > The client is complaining of issues with inter-site calls. They are > reporting issues with crackly and broken speech, and horrible jitter > (or packet loss). This presents a huge issues, because they have one > receptionist answering all calls for both sites. So if a call comes in > from the other site, it automatically an inter-site call, and the > quality falls out of it. If the call is then transfered back to the > originating site, the audio 'bounces' between the two sites, which add > to the call quality degradation. > > We have been monitoring the router while these incidents have been > reported, and it does not appear to be a bandwidth issue. The DSL tail > used for Voice gets to no more then 120k in each direction (we have > tested the links, and can pull data at 53k/s between sites). CPU usage > floats at around 20-25% under load. The router has only shows major > packet loss (that we can tell) when REALLY pushing it in testing (e.g. > 10+ calls between sites). > We have enabled the SIP jitter buffer, as well as the IAX jitter > buffer, which appeared to make a huge difference, but the issue is > still ongoing. > > These issues have also been reported with some outbound VoIP calls. > Internal calls, and calls directly in or out of the Sangoma card are > clear, with no issues reported. > > Does anyone have any thoughts on what could be causing these issues? > We have been racking our brains here, and have tried everything that > we can think of. These system is a million times better then what is > what when it was first installed, but it is still not where it should > be in terms of quality. > > Any thoughts/ideas are most welcome. > > Thank you > > > > *Daniel Cole **(CCNA)** * > > // > > > P Please consider the environment before you print this e-mail or any > attachments. > > > > >___ >--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 > -- Andres Technical Support http://www.telesip.net ___ --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] Call Quality Issues With 2 Trixbox's - RouterIssue?
How are the calls being transferred from Box A to Box B? On what box is the receptionist registered too? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Cole Sent: Tuesday, December 11, 2007 9:00 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call Quality Issues With 2 Trixbox's - RouterIssue? Hello Everyone, We have recently installed a pair of Trixbox servers in for a client of our. They have two locations, with one server each. The servers terminate 3 standard POTS lines into a Sangoma A200D card. The servers are IBM x3250 1RU servers (1GB Ram, Raid 1 160GB HDD, Dual Core Xenon Processors). We are using Trixbox 2.2, and G729 all around. Each site has two (2) 512k/512k ADSL connections terminating into a Cisco 877W router (using an additional 'dumb' modem in a separate VLAN for the extra dsl connection). Using policy based routing, all Voice Data goes over one DSL connection (the one that terminates directly into the router), and all other traffic (e.g. Web and VPN) goes out the second connection (the bridged dumb dsl modem). We are also the ISP for this client, and as thus we have full monitoring of our Layer 2 and Layer 3 networks. From our analysis, it doesn't appear that there is any issue in these networks. We have other customers using the VoIP service, who have not complained of these issues. Now for the Fun part! The client is complaining of issues with inter-site calls. They are reporting issues with crackly and broken speech, and horrible jitter (or packet loss). This presents a huge issues, because they have one receptionist answering all calls for both sites. So if a call comes in from the other site, it automatically an inter-site call, and the quality falls out of it. If the call is then transfered back to the originating site, the audio 'bounces' between the two sites, which add to the call quality degradation. We have been monitoring the router while these incidents have been reported, and it does not appear to be a bandwidth issue. The DSL tail used for Voice gets to no more then 120k in each direction (we have tested the links, and can pull data at 53k/s between sites). CPU usage floats at around 20-25% under load. The router has only shows major packet loss (that we can tell) when REALLY pushing it in testing (e.g. 10+ calls between sites). We have enabled the SIP jitter buffer, as well as the IAX jitter buffer, which appeared to make a huge difference, but the issue is still ongoing. These issues have also been reported with some outbound VoIP calls. Internal calls, and calls directly in or out of the Sangoma card are clear, with no issues reported. Does anyone have any thoughts on what could be causing these issues? We have been racking our brains here, and have tried everything that we can think of. These system is a million times better then what is what when it was first installed, but it is still not where it should be in terms of quality. Any thoughts/ideas are most welcome. Thank you Daniel Cole (CCNA) P Please consider the environment before you print this e-mail or any attachments. ___ --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] Call Quality Issues with IAX?
Funny you mention this because I've run into some voice degradation problems with IAX2 myself recently... When I have an external call come in on a DiD I frequently have to send it back out to the PSTN (i.e. to a cell phone). When this happens I don't want my server in the media path, I want to hand it off to my ITSP and let them handle both ends of the call. I couldn't get it to work with SIP through the provider I'd been working with so I moved to a new ITSP and I switched from SIP to IAX2 at the same time. I had much better success transferring the call back to my ITSP using IAX2 - I could see the handshakes in the CLI and I could phyically disconnect my * server from the Ethernet once the call had been established. Unfortunately the call quality suffered terribly and was unacceptable. I had much better quality using SIP on my old ITSP, even with the media passing through my Asterisk box. So I'm curious whether this is an IAX2 problem or whether my new ITSP is simply not that good. Any thoughts? I don't think the problem can possibly be on my server given that the call is completly handed off but could I be missing something? Thanks, H On 11/5/06, Aaron J. Angel <[EMAIL PROTECTED]> wrote: Hey all, I recently got a message from my provider about IAX: > We do not recommend the use of IAX. It is a lossy protocol that is > known to cause crackling, loss of audio and other issues. You can > use IAX if you want, but we will not assist with any issues you may > encounter. Does anyone else know about these "known" problems? I'm not sure where this provided got this information, but it sounds like a crock. I've never experienced any of the above issues with IAX. I am concered about the reference to a "lossy protocol". How is a protocol lossy? I've heard of lossy compression, which has nothing to do with the protocol used to trasmit compressed data...but I've never heard of a lossy "protocol". Thoughts? Thanks, Aaron -- http://www.aaronjangel.us/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Quality / Echo / Problems
Try running the echo test from both the house side and the co (outside) side. That will let us know where the problem is. Post results. Alex > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Barry Fawthrop > Sent: Monday, October 02, 2006 6:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Call Quality / Echo / Problems > > Hi all > I'm having a problem getting usable quality from my Asterisk setup. > > *SETUP* > 2 Ghz PC with 1 GB Ram with TDM 400p 1 x FXS to route to analog phones > in the house and 2 x FXO to receive calls and in the future faxes. > Gentoo Linux > > > Here is what I've done so far > (1) Moved theTDM 400p (FXS, , FXO, FXO) to it's own interrupt (It was > sharing in the past) > cat /proc/interrupts >CPU0 > 0: 10236724 XT-PIC timer > 1:486 XT-PIC i8042 > 2: 0 XT-PIC cascade > 5: 40694267 XT-PIC wctdm<== > 10: 196233 XT-PIC eth0 > 12:225 XT-PIC i8042 > 14: 247177 XT-PIC ide0 > 15: 26 XT-PIC ide1 > NMI: 0 > LOC: 0 > ERR: 0 > MIS: 0 > > (2) I'm running the latest kernel > uname -r > 2.6.17- > > (3) I am running the latest Asterisk > Asterisk-1.2.12.1 Libpri-1.2.3 zaptel-1.2.8 > I compiled Zaptel with make clean ; make linux26 ; make install > > (4) ztmonitor has become my friend > the Ring sends the VU meter off the chart, but the voice is below half > way. > I have tried changing the rxgain, txgain but that doesn't improve > much. It raises the Volume and I can heard better. > But the "feedback" (What I hear of myself in my analog headset) is off > tone , too loud and poor > > (5) ztspeed reports > Count: 254114 (Not sure if that is good or bad) ??? > > (6) zttest reports > --- Results after 474 passes --- > Best: 100.00 -- Worst: 92.578125 -- Average: 99.713476 > > (7) I have even run ./fxotune > which generated /etc/fxotune.conf > 3=3,0,0,0,0,0,0,0,0 > 4=4,0,0,0,0,0,0,0,0 > Found > http://www.voip-info.org/wiki/view/Asterisk+fxotune > But the -d -b 3 doesn't work only -i and -s are allowed. > > > PROBLEM > The Call tone has a tin can sound (too much highs and not enough lows, > for those with musical backgrounds) > The Volume has improved. It did sound like I was talking behind my hand > in front of my mouth, but not anymore. > > The is a static HISSS that randomly comes and goes and gets so loud that > it drowns out whatever the calling party is trying to say. > It can be heard on both ends but very Loud on the FXS connected phone > > Any Ideas What I can try next > > Thanks All > > Barry > > > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call quality statistics?
try "iax2 show netstats" On 6/23/06, Dr. Michael J. Chudobiak <[EMAIL PROTECTED]> wrote: Is it possible to set up some sort of call-quality statistics reporting/logging for IAX2 calls? Something that can keep track of dropped packet / jitter trends? (I know "iax2 show channels" shows this info for active calls.) Suggestions appreciated! - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: I think the only time you need a timing source is if you are mixing audio streams, i.e. meetme, MOH. In which case you'd probably need to run ztdummy. Yes , ztdummy is running. I'm going to (temporarily) put a TDM card in the system just to eliminate that possibility. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: I'm not on site, but I remember 1.6.4. I had in place 1.6.2, and had way to many problems with it. I reverted back to 1.5.2 and things cleared up. Is the phone (or Asterisk) performing echo suppression that drops the last part of the tone? I believe the phone does some E.C. along with Asterisk. Also, there are no ZAP cards in the system. What timing source does SIP use to play the incoming media stream? I think the only time you need a timing source is if you are mixing audio streams, i.e. meetme, MOH. In which case you'd probably need to run ztdummy. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: Doug Lytle wrote: Michael Welter wrote: The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet I've noted from employees that the volumes levels on the phones themselves, when set too high will cause crackling. Does the crackling coincide with talking on the local side? What firmware are you running on the Polycoms? I'm not on site, but I remember 1.6.4. It's not really crackling or popping that's the problem. The problem is with dropouts. It also seems that the trailing edge of each word will sometimes be lost (possibly a dropout). If you're familiar with the WWV time signal (303-499-7111), for the first 45 minutes of each hour there is a tone interrupted by a click every second (during the last 15 minutes it's just the clicks). When I listen to this on the Asterisk system, the tone only lasts for a fraction of a second and then silence until the next click. Is the phone (or Asterisk) performing echo suppression that drops the last part of the tone? Also, there are no ZAP cards in the system. What timing source does SIP use to play the incoming media stream? Thanks for your comments, Doug. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: Doug Lytle wrote: Michael Welter wrote: The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet I've noted from employees that the volumes levels on the phones themselves, when set too high will cause crackling. Does the crackling coincide with talking on the local side? What firmware are you running on the Polycoms? Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Doug Lytle wrote: Michael Welter wrote: I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. RX Gains too high IRQ sharing of the of the ZAP device There is no ZAP device (it is a SIP-only implementation) and there are no interrupts being shared. High load of the machine The machine is totally idle. The T1 vendor noticed 2% packet loss during a ping flood originating from outside. We changed the Cisco IAD, and there is no longer packet loss, but we still have the clipping. Asterisk 1.2.4. -- Michael Welter Telecom Matters Corp. Denver, Colorado US +1.303.414.4980 [EMAIL PROTECTED] www.TelecomMatters.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problems
Michael Welter wrote: I'm having difficulty with an Asterisk system. The external party has very good call quality, but the internal party hears clipping and drop outs. RX Gains too high IRQ sharing of the of the ZAP device High load of the machine Are a few that come to mind. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality problem when using lan
On 03:57, Sat 20 Aug 05, Ing. Marlo R. Beltran G wrote: > Hi > > i just implemented asterisk and is such a grate solution...i am using > polycom 301 and 501 phoneson lan a iam using g.711 and i have a > 16 port linksys switch... > > the problem come when somebody inside the network is making a call to > other extension (in the same network) and is sending an e > mail trough internet the quality goes down...it hears rally bad... > > i am on a 10/100 network with cat5e on wire, and switches...what can > i do to have an excelent voice quality inside my network??? > > The e mail doesn't go trough the asterisk computer. Hi, Maybe QoS is the answer to this problem. We have a lot more traffic on our lan, and we made all terminals QoS aware. We are giving the voice packets priority and even a ping -f cannot influence call quality. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALL QUALITY PROBLEM...
Hi Some basic mailing lists ethics: 1. writing in CAPITAL LETTERS usually indicates SHOUTING. Please don't do that. 2. when you want to start a new message to the list, write a new message, and don't just reply to an existing list message. 3. Proper English is also preffered, so readers spend less time on trying to understand your English and more on trying to help you. (/me no native English speaker and I know it shows well on my messages. I try, though). For the convinience of the readers, quoted message was converted to small caps (gu). See reply below, On Fri, Aug 19, 2005 at 07:18:44PM -0700, Ing. Marlo R. Beltran G wrote: > hi > > i just implemented asterisk and is such a grate solution...i am using it > polycom 301 and 501 phoneson lan a iam using g.711 and i have a 16 port > linksys switch... > > the problem is when somebody inside the network is making a call to other > extension (in the same network) and for example is sending an e mail to the > internet the quality goes down...it hears rally bad... > > i am on a 10/100 network with cat5e on cable, and switches...what can i do > to have an excelent voice quality inside my network??? Do you actually use a 100Mbit full-duplex network? mii-tool is the simplest way to check that on Linux. If your card does not support it, maybe the messages in dmesg will tell. Also: does the relevant mail go through the computer that runs Asterisk? If so, all the switching may be irrelevant. In that case this may also be due to CPU usage issues rather than a network issue. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality degradation after time
Thanks for the reply, Adam. If this is the case, it would seem to me (because the degradation happens only after a period of time, and quite suddenly) that the issue lies with digium's implementation of g729. As an interesting note, I had the same problems using ulaw -> ulaw over the local network (from internal phone to internal phone) with a much shorter period of 'good voice' before degradation--which was my reason for switching to g729, which seemed to solve the problem. If there is no other current solution, it would seem to me the best thing to do would be to force g729 for non PSTN connections, and ulaw (im in the US) for calls going out the PRI. Has anyone done this..? I'm still hoping to be able to stick with g729; anyone else experience this kind of issue? -a Adam Goryachev wrote: On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote: I'm using Polycom 501's; with stable1.0.8, g729 and a very decent machine; we have a PRI interface to a T1. Many users complain that after a given amount of time, say, 30 or 40 minutes on a call, the outside party complains that their sound keeps 'cutting in and out'. I believe that the incoming sound quality remains fine. I had read that there may be some soft of memory issue with the Polycom's... what else could be causing this? Will auto-rebooting the phones once every few days fix this problem? We had this problem initially, except it happened from the beginning of the call. The fix was to tell the phones to prefer alaw and to tell asterisk to prefer alaw (and not allow anything else). This meant that we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It solved the above symptom. PS, this is in Australia and was with CVS stable 1.0.x from about 12 months ago. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality degradation after time
Thanks for the reply, Adam. If this is the case, it would seem to me (because the degradation happens only after a period of time, and quite suddenly) that the issue lies with digium's implementation of g729. As an interesting note, I had the same problems using ulaw -> ulaw over the local network (from internal phone to internal phone) with a much shorter period of 'good voice' before degradation--which was my reason for switching to g729, which seemed to solve the problem. If there is no other current solution, it would seem to me the best thing to do would be to force g729 for non PSTN connections, and ulaw (im in the US) for calls going out the PRI. Has anyone done this..? I'm still hoping to be able to stick with g729; anyone else experience this kind of issue? -a Adam Goryachev wrote: On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote: I'm using Polycom 501's; with stable1.0.8, g729 and a very decent machine; we have a PRI interface to a T1. Many users complain that after a given amount of time, say, 30 or 40 minutes on a call, the outside party complains that their sound keeps 'cutting in and out'. I believe that the incoming sound quality remains fine. I had read that there may be some soft of memory issue with the Polycom's... what else could be causing this? Will auto-rebooting the phones once every few days fix this problem? We had this problem initially, except it happened from the beginning of the call. The fix was to tell the phones to prefer alaw and to tell asterisk to prefer alaw (and not allow anything else). This meant that we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It solved the above symptom. PS, this is in Australia and was with CVS stable 1.0.x from about 12 months ago. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality degradation after time
On Thu, 2005-07-21 at 15:56 -0400, Adam Dobrin wrote: > I'm using Polycom 501's; with stable1.0.8, g729 and a very decent > machine; we have a PRI interface to a T1. > > Many users complain that after a given amount of time, say, 30 or 40 > minutes on a call, the outside party complains that their sound keeps > 'cutting in and out'. I believe that the incoming sound quality remains > fine. > > I had read that there may be some soft of memory issue with the > Polycom's... what else could be causing this? Will auto-rebooting the > phones once every few days fix this problem? We had this problem initially, except it happened from the beginning of the call. The fix was to tell the phones to prefer alaw and to tell asterisk to prefer alaw (and not allow anything else). This meant that we were using alaw from PSTN -> asterisk -> SIP phone ie, everywhere. It solved the above symptom. PS, this is in Australia and was with CVS stable 1.0.x from about 12 months ago. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Quality Detail Record
At 3:32 PM +0200 on 3/17/05, Calin Serbanescu wrote: Hello, I need some help setting up statistics per call. I need to store in a database call quality details such as jitter, packets lost and other informations. Is there any way to do this? I'd really appreciate some links or any other kind of info on this. Thanks, Calin. If you're using Cisco 79xx devices with reasonably new SIP images, or VOIP Inc. devices, you might have some luck since both of those support the Tx and Rx header stats out of SIP BYE packets. http://lists.digium.com/pipermail/asterisk-dev/2004-May/004174.html Since that post, there are now built-in routines to extract SIP headers in the dialplan - see "show application SIPGetHeader". You may be able to get that to work instead of making your own patches. There is also now some stuff in the IAX2 statistics area (as of yesterday?) but I don't know if it's reference-able from the dialplan or anywhere else. Maybe someone can fill me in on that? You're on your own for making a database of the information, though. My JT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call quality monitoring
Chris Icide <[EMAIL PROTECTED]> writes: > Satellite links can be pretty tough to troubleshoot. It sounds like > you are running into a uplink buffer issue. On heavily loaded > uplinks, the input buffers can get quite large, and if the satellite > provider isn't using some form of buffer handling that prioritizes udp > traffic, it may be that most of your voice packets are falling on the > floor of the uplink facility... Yeah, I thought about that too, which is why I set up tools to monitor packet loss and measure jitter. It uses a non-conflicting UDP port near the one that IAX2 uses. My tests indicate very little loss and the same jitter in both directions. What I really want is a way to get asterisk to yelp if it notices that its about to make the call sound bad due to late/missing packets or for whatever reason. Right now it seems that the network is functioning normally but still one direction of the calls sounds intermittently awful. -- Matt Ranney - [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call quality monitoring
mjr, Satellite links can be pretty tough to troubleshoot. It sounds like you are running into a uplink buffer issue. On heavily loaded uplinks, the input buffers can get quite large, and if the satellite provider isn't using some form of buffer handling that prioritizes udp traffic, it may be that most of your voice packets are falling on the floor of the uplink facility... -Chris On 10 Sep 2004 11:37:33 -0700, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: > I need to debug a call quality issue with remote users on the other > end of a satellite link. The symptoms are: we here on the Internet > side can hear them just fine. On their end, things work sorta OK most > times, but they often suffer from severe dropouts and digital > warbling, both of which I attribute to them missing packets. Often > times they can't make out a word we are saying while we can hear them > crystal clearly. > > Various pings and other network tests indicate that the underlying > network is functioning as well as can be expected for a sat link. In > fact, the overall jitter seems to be pretty low (avg 20ms). Packet > loss is around 1-2%, and latency is around 700ms on average. > > I'm left to assume that the jitter buffer on that end isn't > functioning properly. Both ends of the call have the same jitter > buffer settings. The call is carried by IAX2 and encoded with ILBC. > > The iax.conf files on each end start like this: > > >[general] > >trunk=no > >notransfer=yes > >iaxcompat=no > > > >bandwidth=low > > > >disallow=all > >allow=ilbc > > > >jitterbuffer=yes > >dropcount=3 > >maxjitterbuffer=500 > >maxexcessbuffer=150 > >minexcessbuffer=40 > >jittershrinkrate=1 > > Of course, perhaps the jitter buffer isn't to blame, but given that > one side of the call sounds perfect, I can't think of anything else > obvious that would cause this. > > Is there any way to extract from asterisk some idea of why it thinks > the calls sound bad? For example, when the jitter buffer notices that > packets are discarded because they are too late, when excessive > packets are completely missing, etc. > > I've been collecting a giant debug log for a while now, so I could > pretty easily sift through it if there's something good to look for. > > Thanks. > -- > Matt Ranney - [EMAIL PROTECTED] > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call quality questions
Lane Hoskins wrote: > Our basic system is as follows: > > P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS > several weeks ago, working OK for routing, VM, and AA, calls in on > separate PSTN lines to Adtran TSU 600, into * server through T100P > card. The hardware is not taxed at all with little over 20% proc > utilization ever, low mem use, etc. All Phones are SNOM 200's with > various firmware revisions from 2.2t to 2.3o. I have finally gotten my Snom 200's to work. I upgraded them all to the 2.3o. > Any help is appreciated. > > 1. We have horrible sound quality regardless of the codec we use in > the phone or specify in *. Has anyone else run into this early on and > found a software fix? I had the same problem with sound. It happen to be my switch hub that I had. I changed it out to a good switch and for internal network I use alaw. For outside network (Over the internet) I use gsm which has sometimes has sound problems. > 2. Speakerphone will not work for playing VM messages, it chops the > message into unintelligible fragments of audio. Any ideas? Same problem as above. > 3. Initially we have horrible introduction of background noise into > the handset earpiece which seems to quiet after there is audio on the > other end. Ideas? I did not have this problem. > 4. Sound quality to called parties outside our system is > intermittently horrible: static filled and raspy where we have to ask > people to repeat themselves many times. Could this be related to > powerline noise or something like that? > If you want I can call you to explain more on what we did with our settings. > > Lane Hoskins, MCP > Network Engineer > Automated Horizons Inc. > Direct - 540.767.7626 > Main - 540.767.7600 > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call quality questions
Thanks... -Original Message- From: mattf [mailto:[EMAIL PROTECTED] Sent: Fri 1/30/2004 5:16 PM To: '[EMAIL PROTECTED]' Cc: Subject: RE: [Asterisk-Users] Call quality questions Hello, Did you set the flag in the makefile for zaptel for SMP kernels? Yes, we are using the SMP settings 1. I have a couple Snom200 phones on my system running redhat with a P4 HT and haven't had any issues with horrible sound quality using 711ulaw. Have tried everything - alaw, ulaw, gsm - currently using gsm 2. As for the speakerphone cutout, that's to be expected, The snom200s are just half-duplex speakerphones. If you want a good speakerphone get a Polycom. wish I'd known that before I recommended we buy 12 of them :-( :-) 3. If you have silence suppression turned on anywhere I would turn it off. We had the same problem with our Sipura adapters until we turned off the silence supression. You could also mess with the level of echo supression and see if that makes a difference. this I need to look into 4. do you have your T1 set in zaptel.conf to be the primary timing source for your card? yes Hope that helps, MATT--- -Original Message- From: Lane Hoskins [mailto:[EMAIL PROTECTED] Sent: Friday, January 30, 2004 4:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call quality questions Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions from 2.2t to 2.3o. Any help is appreciated. 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users <>
RE: [Asterisk-Users] Call quality questions
Hello, Did you set the flag in the makefile for zaptel for SMP kernels? 1. I have a couple Snom200 phones on my system running redhat with a P4 HT and haven't had any issues with horrible sound quality using 711ulaw. 2. As for the speakerphone cutout, that's to be expected, The snom200s are just half-duplex speakerphones. If you want a good speakerphone get a Polycom. 3. If you have silence suppression turned on anywhere I would turn it off. We had the same problem with our Sipura adapters until we turned off the silence supression. You could also mess with the level of echo supression and see if that makes a difference. 4. do you have your T1 set in zaptel.conf to be the primary timing source for your card? Hope that helps, MATT--- -Original Message- From: Lane Hoskins [mailto:[EMAIL PROTECTED] Sent: Friday, January 30, 2004 4:22 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call quality questions Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions from 2.2t to 2.3o. Any help is appreciated. 1. We have horrible sound quality regardless of the codec we use in the phone or specify in *. Has anyone else run into this early on and found a software fix? 2. Speakerphone will not work for playing VM messages, it chops the message into unintelligible fragments of audio. Any ideas? 3. Initially we have horrible introduction of background noise into the handset earpiece which seems to quiet after there is audio on the other end. Ideas? 4. Sound quality to called parties outside our system is intermittently horrible: static filled and raspy where we have to ask people to repeat themselves many times. Could this be related to powerline noise or something like that? Lane Hoskins, MCP Network Engineer Automated Horizons Inc. Direct - 540.767.7626 Main - 540.767.7600 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users