Our office is about to switch from an analog pbx to asterisk. The last thing we are trying to resolve is the echo. We have a TDM400 with 4 FXS cards installed. No echo on other sip extensions but when making or receiving calls over the outside line there is a bad echo. We only have one phone line plugged into the asterisk box now and 9 other lines to our old pbx. I tried everything in the zapata.conf with the gains and echo cancel but nothing works. Today just to see what would happen I unplugged one of our other lines from our old pbx and plugged it into the asterisk box. No echo. So with one standard line I have a ton of echo and with another line I have no echo. I’m assuming the problem is with the lines not asterisk? Is this because there is more of a delay on one line then the other? When I plug in a standard anolog phone to the line that gives me echo on asterisk there is no echo. Is the delay to small to notice with the anolog phone but when going through voip it’s more noticeable? Also, if thats the case is this something I could call the phone company about and have them fix? Or are they going to say it works fine with an analog phone so it’s not there problem?
I would try some of the other 9 and see what results you get.
If they all work good too, then it has to be in the line.
Make sure that the line with echo is ran the same place as the others. Interferance could be something to do with it…
Have you run ‘fxotune’ to set up the TDM400 line matching?
This can make an incredible difference on line quality.
Description here:
voip-info.org/wiki/view/Asterisk+fxotune
If (like me) you are not in the USA:
Check what number you need to dial to get a quiet line. fxotune -i 1 seems to work OK in the UK. (Dialling ‘1’ on a normal exchange line gives 20 seconds of silence).
Also, make sure you set the correct parameter for the Zap modules, like
options wctdm opermode=UK
before the
install wctdm … line in modprobe.conf
(And for anyone using other analog zap hardware, also
options wctdm24xxp opermode=UK
options wctdm8xxp opermode=UK
before the respective install lines)
Echo is a serious problem, the phone company won’t help you, as they’ll say it’s fine with an ordinary phone. If you have nine lines, I would recommend going digital (ISDN) You’ll probably never get rid of the echo issue, and your users will hate you otherwise.
Howdy,
If you’ve four FXS modules on your TDM400P, what are you using for PSTN termination?
Do you perhaps have an FXO module on that TDM400P, or maybe four?
Give Digium support a ring about HPEC. HPEC is a commercially-licensed, toll-quality echo canceller available at no-charge to in-warranty analog Digium hardware customers.
support@digium.com
Cheers.
fxotune worked. When I start asterisk I get a huge echo. After setting fxotune with /usr/src/zaptel-1.4.4/fxotune -s the echo is gone. Do you find that you have to re-tune often? Do you thing that with the echo corrected with fxotune that the system would be reliable enough for use in an office?
Also we are testing out phone to see what one we like. We have tried the grandstream gxp-2000 and really like the blf features and the ard line buttons, but the sound is not that great. We have also tried the linksys spa922 and it sounds a lot better but is hard to use when putting calls on hold and resuming them with the soft buttons, and no blf. Any suggestions on a phone the sounds good, is easy to use and has the blf that will work with asterisk? and $100-$150 each.
Stay away from the gxp2000’s for an office envoirenment, the handsets will induce echo and you’ll have complaints that other people appear to be on the line. To be honest, you can’t beat Aastra’s for office work horses.
Your suspicion about the added delay is quite possibly true. The perception of echo is very much dependent upon the round-trip delay through the telephone network. When making calls local to an Asterisk system using analog phones, the delay is probably not enough that one would perceive echo. But when you connect to the public telephone network, the delay increases and it depends upon where you are calling.
One thing that I didn’t quite understand from your posting is this - which end is hearing the echo - the Asterisk user or the person at the opposite end of the outside phone call?
If you are hearing the echo on the local phone, I have to ask - do you have echo cancellation turned on for the FXO port?
Regardless, I highly recommend using the High Performance Echo Canceller (HPEC), which can be obtained from Digium support. It is a carrier-class echo canceller. One of the benefits that might be helpful in your situation is that it handles echo tails much longer than those handled by the default Asterisk echo canceller.
Your suspicion about the added delay is quite possibly true. The perception of echo is very much dependent upon the round-trip delay through the telephone network. When making calls local to an Asterisk system using analog phones, the delay is probably not enough that one would perceive echo. But when you connect to the public telephone network, the delay increases and it depends upon where you are calling.
One thing that I didn’t quite understand from your posting is this - which end is hearing the echo - the Asterisk user or the person at the opposite end of the outside phone call?
If you are hearing the echo on the local phone, I have to ask - do you have echo cancellation turned on for the FXO port?
Regardless, I highly recommend using the High Performance Echo Canceller (HPEC), which can be obtained from Digium support. It is a carrier-class echo canceller. One of the benefits that might be helpful in your situation is that it handles echo tails much longer than those handled by the default Asterisk echo canceller.
Your suspicion about the added delay is quite possibly true. The perception of echo is very much dependent upon the round-trip delay through the telephone network. When making calls local to an Asterisk system using analog phones, the delay is probably not enough that one would perceive echo. But when you connect to the public telephone network, the delay increases and it depends upon where you are calling.
One thing that I didn’t quite understand from your posting is this - which end is hearing the echo - the Asterisk user or the person at the opposite end of the outside phone call?
If you are hearing the echo on the local phone, I have to ask - do you have echo cancellation turned on for the FXO port?
Regardless, I highly recommend using the High Performance Echo Canceller (HPEC), which can be obtained from Digium support. It is a carrier-class echo canceller. One of the benefits that might be helpful in your situation is that it handles echo tails much longer than those handled by the default Asterisk echo canceller.
Your suspicion about the added delay is quite possibly true. The perception of echo is very much dependent upon the round-trip delay through the telephone network. When making calls local to an Asterisk system using analog phones, the delay is probably not enough that one would perceive echo. But when you connect to the public telephone network, the delay increases and it depends upon where you are calling.
One thing that I didn’t quite understand from your posting is this - which end is hearing the echo - the Asterisk user or the person at the opposite end of the outside phone call?
If you are hearing the echo on the local phone, I have to ask - do you have echo cancellation turned on for the FXO port?
Regardless, I highly recommend using the High Performance Echo Canceller (HPEC), which can be obtained from Digium support. It is a carrier-class echo canceller. One of the benefits that might be helpful in your situation is that it handles echo tails much longer than those handled by the default Asterisk echo canceller.
The echo effect is from where the ‘hybrid’ in the card converts separate incoming & outgoing audio and the two-way audio on the phone line.
If everything works perfectly, the hybrid cancels the outgoing audio so it does’nt appear on the receive side, while passing full-level audio inward from the line.
It relies on matching the impedance of the line over the full range of frequencies, which is not easy.
If the matching is not perfect, some outgoing audio comes back on the incoming side.
fxotune optimises the match at the hardware level, by setting up the best complex impedance match. This should not change as long as the line itself is not changed at any point.
The reason echo is such a problem with VoIP is due to the processing delays both ways between the phone and the FXO port.
With a simple analog phone, you can hear yourself in the earpiece to some extent but it’s in sync with your own speech and your brain is used to handling that.
With VoIP system echo, there is a significant delay and you hear it completely apart from your own speech, which is extremely annoying and distracting.