canuck15
2005-Aug-24 15:33 UTC
[Asterisk-Users] FW: [Asterisk-Users, Andrew] Will Echo problems EVER be solved, I'm scared
Thank you for the suggestions Andrew. I have not come across some of them before and will give them a shot. Based on my reading, changing the motherboard should have minimal impact unless that motherboard and the TDM400P don't get along (aka. IRQ sharing). I have disabled everything that is not needed and I do not believe I have any IRQ problems and I am NEVER wrong ;). Calls are crisp and clear. . No snap, crackle, pop. It would be a beautiful thing if not for the echo. To get the RX/TX levels, run "ztmonitor 1 -vv", dial a telco 1004hz 0dbm test phone # and set the quantitative RX number to around 14500. With 2 lines (which I don't have) you test the TX level by looping out to the other PSTN. Without a second line you do the simple ztmonitor test for the TX levels. http://www.voip-info.org/tiki-index.php?page=Asterisk+zapata+gain+adjustment I was MOST DEFINITELY NOT wildly changing settings. It would require a whole book to explain properly what I did but the end result was that I pretty much covered every possible combination of settings. I have read the white papers and EVERYTHING else I could find on the web to determine the most logical and proper way to go about this. I was NOT approaching this like some back yard six pack scientist. I made a mistake when I used the word "levels" to describe what fxotune does. The bottom line is that it did not change anything. My settings are pretty much default except where I stated otherwise. Network is a Linksys WRT54g (ie. Switch). Asterisk server on port 1, GXP2000 on port 2, 9133i on port 3. I have NO echo between SIP phones! #cat /proc/pci Bus 0, device 0, function 0: Host bridge: Intel Corp. 82815 815 Chipset Host Bridge and Memory Controller Hub (rev 2). Prefetchable 32 bit memory at 0xd0000000 [0xd3ffffff]. Bus 0, device 1, function 0: PCI bridge: Intel Corp. 82815 815 Chipset AGP Bridge (rev 2). Master Capable. Latency=32. Min Gnt=12. Bus 0, device 30, function 0: PCI bridge: Intel Corp. 82801BA/CA/DB/EB PCI Bridge (rev 1). Master Capable. No bursts. Min Gnt=6. Bus 0, device 31, function 0: ISA bridge: Intel Corp. 82801BA ISA Bridge (LPC) (rev 1). Bus 0, device 31, function 1: IDE interface: Intel Corp. 82801BA IDE U100 (rev 1). I/O at 0xf000 [0xf00f]. Bus 0, device 31, function 3: SMBus: Intel Corp. 82801BA/BAM SMBus (rev 1). IRQ 11. I/O at 0x5000 [0x500f]. Bus 1, device 0, function 0: VGA compatible controller: ATI Technologies Inc Rage 128 RF/SG AGP (rev 0). IRQ 10. Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xd4000000 [0xd7ffffff]. I/O at 0x9000 [0x90ff]. Non-prefetchable 32 bit memory at 0xd9000000 [0xd9003fff]. Bus 2, device 1, function 0: Unknown mass storage controller: PCI device 1095:3124 (CMD Technology Inc) (rev 1). IRQ 11. Master Capable. Latency=32. Non-prefetchable 64 bit memory at 0xdb008000 [0xdb00807f]. Non-prefetchable 64 bit memory at 0xdb000000 [0xdb007fff]. I/O at 0xa000 [0xa00f]. Bus 2, device 2, function 0: Communication controller: Tiger Jet Network Inc. Intel 537 (rev 0). IRQ 5. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0xa400 [0xa4ff]. Non-prefetchable 32 bit memory at 0xdb009000 [0xdb009fff]. Bus 2, device 4, function 0: Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 16). IRQ 10. Master Capable. Latency=32. Min Gnt=32.Max Lat=64. I/O at 0xa800 [0xa8ff]. Non-prefetchable 32 bit memory at 0xdb00a000 [0xdb00a0ff]. #cat /proc/interrupts CPU0 0: 1290132 XT-PIC timer 1: 4 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 12878920 XT-PIC wctdm 8: 1 XT-PIC rtc 10: 33866 XT-PIC eth0 12: 41 XT-PIC PS/2 Mouse 14: 17345 XT-PIC ide0 15: 60 XT-PIC ide1 NMI: 0 ERR: 0 -----Original Message----- From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] Sent: Wednesday, August 24, 2005 1:54 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared On Wednesday 24 August 2005 16:37, canuck15 wrote:> As others have recommended, I created a test system with the proposed > production parts. I bought a couple different SIP phones to try and a > Digium TDM01B card. I am using an older PIII 1Ghz system with > 815chipset (PCI Rev2.2) with 256MB for my test system. The only thing > that will be different on a production system is that I will be using > a newer chipset PC with faster processor and 512MB. Probably Intel > 7505, 7210, or 7211 chipsets which seem to be the most compatible withAsterisk. So in other words, everything will be changing on your production system. Not a good way to start.> My problem is that I cannot eliminate echo no matter what I try. I > seriously doubt that a newer chipset faster PC with more memory will > eliminate or even reduce my echo problems based on what I have read. Iam> not about to drop more cash to try and find out. Essentially, my > findings are that Asterisk is NOT production capable for my > configuration which is via FXO and PSTN. That is probably THE most > common configuration so if it is not production capable like that it > isn't production capable period as far as I'm concerned. What adisappointment :(. Most of us don't have any trouble.> *Buy latest TDM400P with latest FXO module *Ensure copper connection > to analog telco lines and telco are not causing problems including > running a separate shielded line to the demarc AND having the telco > guy come out and test the levels, impedance etc.I'd be damn curious to know what you got out of this -- most telco guys will do a basic metallic check, throw on a butt-set and say "yup, I got dialtone." -- hardly a real check but that's neither here nor there. I'm also in Canada (1.5hrs from Toronto, ON) so I'm *really* curious who you got on the line to do a real line test with you. I have resorted to buying my own telco test equipment off ebay and using that, even though our techs here are excellent.> *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor > method and by using the detailed Ztmonitor method via a Telco > 102milliwatt test phone #. The end result was RX=8.0, TX=-1.0. Since > I still have echo problems I have tried all sort of other settings withoutsuccess. Ok good. Can you detail exactly what you did to reach these numbers? I'm curious.> *After ALL of the above, try every possible combination of all of the > following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, > 64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 2 > (default, aggressive, CVS head developments, bugs.digium.com patches, > adjust threshold level as per wiki etc. etc.)I'd posted something earlier that basically says this: Without measured, controlled tests, you're just pissing up a rope. Wildly changing settings and hoping for the best does nothing but cost you time and energy.> *Run fxotune which did not find a need to adjust the FXO levels > (1=0,0,0,0,0,0,0,0)fxotune doesn't adjust FXO levels, it adjusts a very simple FIR filter which is part of the DAA in the FXO module. IMO it helps with audio quality but not much with echo.> Still have echo. Aggressive mode helps a bit but then the other > persons voice get's cut off a lot especially when I talk and the > cutting in and out of the canceller is more noticeable and > objectionable in general than if Aggressive is turned off.Agressive mode turns the phone line into a half-duplex environment. When your voice energy is detected it mutes the receive audio.> I have two SIP phones. An Aastra 9133i and a Grandstream GXP2000. > Echo problem is the same on both phones.Do you have echo between the two phones? What about when calling out to a VOIP provider, dialing a DID you own that comes back in and hits the other phone?> Any comments and/or suggestions would be greatly appreciated as I am > pretty much out of ideas and ready to give up on Asterisk as a > suitable traditional small business phone system replacement.I haven't seen your zconfig.h nor your zaptel Makefile, and you didn't tell us anything about your network (network card, switch, etc.). My general advice for zaptel is to do the following: zaptel Makefile: underneath the comments about zconfig.h add KFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc) CFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc) and in zconfig.h - enable XLAW (optimize for small # of zap channels) - enable MMX - MARK2, no agressive mode. Whenever I've done that my echo has largely disappeared. Have you also tried flipping tip and ring going into the TDM card? -A.
Rich Adamson
2005-Aug-25 04:33 UTC
[Asterisk-Users] Re: [Asterisk-Users, Andrew] Will Echo problems EVER be solved, I'm scared
I'll jump in here with one comment. I worked with an individual in Canada that could not get rid of the echo (some time ago with an x100p). As a very experienced telephony engineer and two years of asterisk experience, I logged into his system and tried many many changes without impacting the problem. He swapped his motherboard from a rather current one to an older slower P3. Echo disappeared. Same distro, same * code, same card. We had no idea what the root cause was but guessed it had something to do with the pci bus and/or interrupt structure that we could not quantify with anything that we had available at the time. Would have been very interesting to have had the original motherboard in a captive lab environment to jack with. ------------------------> Thank you for the suggestions Andrew. I have not come across some of them > before and will give them a shot. Based on my reading, changing the > motherboard should have minimal impact unless that motherboard and the > TDM400P don't get along (aka. IRQ sharing). I have disabled everything that > is not needed and I do not believe I have any IRQ problems and I am NEVER > wrong ;). Calls are crisp and clear. . No snap, crackle, pop. It would > be a beautiful thing if not for the echo. > > To get the RX/TX levels, run "ztmonitor 1 -vv", dial a telco 1004hz 0dbm > test phone # and set the quantitative RX number to around 14500. With 2 > lines (which I don't have) you test the TX level by looping out to the other > PSTN. Without a second line you do the simple ztmonitor test for the TX > levels. > http://www.voip-info.org/tiki-index.php?page=Asterisk+zapata+gain+adjustment > > I was MOST DEFINITELY NOT wildly changing settings. It would require a > whole book to explain properly what I did but the end result was that I > pretty much covered every possible combination of settings. I have read the > white papers and EVERYTHING else I could find on the web to determine the > most logical and proper way to go about this. I was NOT approaching this > like some back yard six pack scientist. > > I made a mistake when I used the word "levels" to describe what fxotune > does. The bottom line is that it did not change anything. > > My settings are pretty much default except where I stated otherwise. > Network is a Linksys WRT54g (ie. Switch). Asterisk server on port 1, > GXP2000 on port 2, 9133i on port 3. > > I have NO echo between SIP phones! > > #cat /proc/pci > Bus 0, device 0, function 0: > Host bridge: Intel Corp. 82815 815 Chipset Host Bridge and Memory > Controller Hub (rev 2). > Prefetchable 32 bit memory at 0xd0000000 [0xd3ffffff]. > Bus 0, device 1, function 0: > PCI bridge: Intel Corp. 82815 815 Chipset AGP Bridge (rev 2). > Master Capable. Latency=32. Min Gnt=12. > Bus 0, device 30, function 0: > PCI bridge: Intel Corp. 82801BA/CA/DB/EB PCI Bridge (rev 1). > Master Capable. No bursts. Min Gnt=6. > Bus 0, device 31, function 0: > ISA bridge: Intel Corp. 82801BA ISA Bridge (LPC) (rev 1). > Bus 0, device 31, function 1: > IDE interface: Intel Corp. 82801BA IDE U100 (rev 1). > I/O at 0xf000 [0xf00f]. > Bus 0, device 31, function 3: > SMBus: Intel Corp. 82801BA/BAM SMBus (rev 1). > IRQ 11. > I/O at 0x5000 [0x500f]. > Bus 1, device 0, function 0: > VGA compatible controller: ATI Technologies Inc Rage 128 RF/SG AGP (rev > 0). > IRQ 10. > Master Capable. Latency=32. Min Gnt=8. > Prefetchable 32 bit memory at 0xd4000000 [0xd7ffffff]. > I/O at 0x9000 [0x90ff]. > Non-prefetchable 32 bit memory at 0xd9000000 [0xd9003fff]. > Bus 2, device 1, function 0: > Unknown mass storage controller: PCI device 1095:3124 (CMD Technology > Inc) (rev 1). > IRQ 11. > Master Capable. Latency=32. > Non-prefetchable 64 bit memory at 0xdb008000 [0xdb00807f]. > Non-prefetchable 64 bit memory at 0xdb000000 [0xdb007fff]. > I/O at 0xa000 [0xa00f]. > Bus 2, device 2, function 0: > Communication controller: Tiger Jet Network Inc. Intel 537 (rev 0). > IRQ 5. > Master Capable. Latency=32. Min Gnt=1.Max Lat=128. > I/O at 0xa400 [0xa4ff]. > Non-prefetchable 32 bit memory at 0xdb009000 [0xdb009fff]. > Bus 2, device 4, function 0: > Ethernet controller: Realtek Semiconductor Co., Ltd. > RTL-8139/8139C/8139C+ (rev 16). > IRQ 10. > Master Capable. Latency=32. Min Gnt=32.Max Lat=64. > I/O at 0xa800 [0xa8ff]. > Non-prefetchable 32 bit memory at 0xdb00a000 [0xdb00a0ff]. > > > #cat /proc/interrupts > CPU0 > 0: 1290132 XT-PIC timer > 1: 4 XT-PIC keyboard > 2: 0 XT-PIC cascade > 5: 12878920 XT-PIC wctdm > 8: 1 XT-PIC rtc > 10: 33866 XT-PIC eth0 > 12: 41 XT-PIC PS/2 Mouse > 14: 17345 XT-PIC ide0 > 15: 60 XT-PIC ide1 > NMI: 0 > ERR: 0 > > > > -----Original Message----- > From: Andrew Kohlsmith [mailto:akohlsmith-asterisk@benshaw.com] > Sent: Wednesday, August 24, 2005 1:54 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Will Echo problems EVER be solved, I'm scared > > On Wednesday 24 August 2005 16:37, canuck15 wrote: > > As others have recommended, I created a test system with the proposed > > production parts. I bought a couple different SIP phones to try and a > > Digium TDM01B card. I am using an older PIII 1Ghz system with > > 815chipset (PCI Rev2.2) with 256MB for my test system. The only thing > > that will be different on a production system is that I will be using > > a newer chipset PC with faster processor and 512MB. Probably Intel > > 7505, 7210, or 7211 chipsets which seem to be the most compatible with > Asterisk. > > So in other words, everything will be changing on your production system. > Not a good way to start. > > > My problem is that I cannot eliminate echo no matter what I try. I > > seriously doubt that a newer chipset faster PC with more memory will > > eliminate or even reduce my echo problems based on what I have read. I > am > > not about to drop more cash to try and find out. Essentially, my > > findings are that Asterisk is NOT production capable for my > > configuration which is via FXO and PSTN. That is probably THE most > > common configuration so if it is not production capable like that it > > isn't production capable period as far as I'm concerned. What a > disappointment :(. > > Most of us don't have any trouble. > > > *Buy latest TDM400P with latest FXO module *Ensure copper connection > > to analog telco lines and telco are not causing problems including > > running a separate shielded line to the demarc AND having the telco > > guy come out and test the levels, impedance etc. > > I'd be damn curious to know what you got out of this -- most telco guys will > do a basic metallic check, throw on a butt-set and say "yup, I got > dialtone." > -- hardly a real check but that's neither here nor there. I'm also in > Canada (1.5hrs from Toronto, ON) so I'm *really* curious who you got on the > line to do a real line test with you. I have resorted to buying my own > telco test equipment off ebay and using that, even though our techs here are > excellent. > > > *Adjust RX/TX levels as per Asterisk Wiki using the quick Ztmonitor > > method and by using the detailed Ztmonitor method via a Telco > > 102milliwatt test phone #. The end result was RX=8.0, TX=-1.0. Since > > I still have echo problems I have tried all sort of other settings without > success. > > Ok good. Can you detail exactly what you did to reach these numbers? I'm > curious. > > > *After ALL of the above, try every possible combination of all of the > > following on Asterisk v1.0.9: echocancel (off, on, 128, 256, 16, 32, > > 64), echowhenbridged (on, off), echotraining (off, on, 800), Mark 2 > > (default, aggressive, CVS head developments, bugs.digium.com patches, > > adjust threshold level as per wiki etc. etc.) > > I'd posted something earlier that basically says this: Without measured, > controlled tests, you're just pissing up a rope. Wildly changing settings > and hoping for the best does nothing but cost you time and energy. > > > *Run fxotune which did not find a need to adjust the FXO levels > > (1=0,0,0,0,0,0,0,0) > > fxotune doesn't adjust FXO levels, it adjusts a very simple FIR filter which > is part of the DAA in the FXO module. IMO it helps with audio quality but > not much with echo. > > > Still have echo. Aggressive mode helps a bit but then the other > > persons voice get's cut off a lot especially when I talk and the > > cutting in and out of the canceller is more noticeable and > > objectionable in general than if Aggressive is turned off. > > Agressive mode turns the phone line into a half-duplex environment. When > your voice energy is detected it mutes the receive audio. > > > I have two SIP phones. An Aastra 9133i and a Grandstream GXP2000. > > Echo problem is the same on both phones. > > Do you have echo between the two phones? What about when calling out to a > VOIP provider, dialing a DID you own that comes back in and hits the other > phone? > > > Any comments and/or suggestions would be greatly appreciated as I am > > pretty much out of ideas and ready to give up on Asterisk as a > > suitable traditional small business phone system replacement. > > I haven't seen your zconfig.h nor your zaptel Makefile, and you didn't tell > us anything about your network (network card, switch, etc.). > > My general advice for zaptel is to do the following: > zaptel Makefile: underneath the comments about zconfig.h add > KFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc) > CFLAGS+=-march=pentium4 (or pentium3 or pentiumpro, use the exact proc) > > and in zconfig.h > - enable XLAW (optimize for small # of zap channels) > - enable MMX > - MARK2, no agressive mode. > > Whenever I've done that my echo has largely disappeared. > > Have you also tried flipping tip and ring going into the TDM card? > > -A.