Re: [Asterisk-Users] echo / delay problem
I'm in the US, using cards bought direct from Digium. I have lowered the rxgain and txgain to -8 and that seems to be helping futher. I wish I could understand why? The problem with more time is that I can hear myself in the headset of the std. phone as well as the party on the other end. The other person just hears me. The problem then is somewhere internal. as I can talk into the mouthpiece and hear myself in the speaker??? Thanks Barry ___ 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] echo / delay problem
-Original Message- From: Barry FAWTHROP [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 22, 2005 7:12 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] echo / delay problem I'm in the US, using cards bought direct from Digium. I have lowered the rxgain and txgain to -8 and that seems to be helping futher. I wish I could understand why? Consider the following bad ASCII diagram of the amplitude of the signal as it travels from your handset to the remote end, becomes an echo and comes back to your handset: 1.00| | | 0.75| | (effect of txgain -8.0) | | 0.50|--^- | | 0.25| | (effect of rxgain -8.0) | | 0.00| |- ||||| ||||\Echo level arriving at local handset |||\Echo level arriving from remote end ||| Signal level arriving at remote end | \-Signal level leaving to the PSTN \-Signal level leaving local handset Now consider the amplitude of the legitimate speech originating at the remote end: 1.00| - | | 0.75|| (effect of rxgain -8.0) | | 0.50| | 0.25| | 0.00| |-- || | \Echo level arriving at handset \-Signal level leaving remote handset It follows then that a signal that is attenuated going out and then further attenuated coming back in to your system (ie. the echo) will be relatively quieter than a signal that is just attenuated once (ie. the remote end conversation). Remember that a gain expressed in Db is a logarithmic measure so two passes are more than double the attenuation. This is known as a 'loss plan' in telco spheres and was traditionally used in analog systems to make echo (and sideton) more managable. Theres some interesting blurb on loss planning at: http://telecom.tbi.net/lossplan.htm or consult Google. It's also important to understand that the training algorithms in many echo cancellers rely on being able to differentiate between the echo and legitimate far end signal and one of the mechanisms used is to measure the average relative signal levels. Thus an outgoing signal that is too 'hot' (ie. has excessive amplitude) may not be properly echo-supressed as the signal cannot be differentiated from the far end signal, thus getting your gains correct is an important part of PSTN interfacing. Hope that helps. Kris Boutilier Information Services Coordinator Sunshine Coast Regional District ___ 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] echo / delay problem
I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried and tried for 2 weeks it is rather annoying and irating to hear this delay/echo I would call it a delay since you can hear the end of the sentence repeat over and over. Also every now and again it sounds like a underwater submarine with ping and all. Thanks in advance Barry [channels] echocancel= 16 echocancelwhenbridged = yes echotraining = no ;; yes rxgain= -2.0 txgain= -2.0 If you are in the US and using a digium x100p, then try the following: echotraining=800 echocancel=yes (The values you are showing above aren't even valid parameters.) If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. ___ 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] echo / delay problem
On Sat, Mar 19, 2005 at 07:19:39AM -0600, Rich Adamson wrote: If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. Regards, Mikael Magnusson ___ 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] echo / delay problem
If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. The x100p does not have a programable chipset for impedance matching. That parameter was intended for the TDM card. ___ 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] echo / delay problem
On Sat, Mar 19, 2005 at 09:22:22AM -0600, Rich Adamson wrote: If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. The x100p does not have a programable chipset for impedance matching. That parameter was intended for the TDM card. Ok, maybe it isn't programmable, but there seems to be two different DAA's used in the clones[1]. Si3012 for the US market and Si3014 for the global market. I assume this has to do with the line interface. I bought a clone in Sweden that has the Si3014. It should support the line impedance used in Sweden I think, since it's marked with the following label: Clas Ohlson AB Modell/Malli: AMI-IE92 Art.nr/nro:32-2055 CE For use in Sweden, Norway and Finland. To be connected to the public switched telefone network. Hereby, Clas Ohlson, declares that this modem is in complience with the essential requirements and other relevant provisions of Directive 1999/5/EC. [1]http://www.intel.com/design/modems/linecard.htm Regards Mikael Magnusson ___ 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] echo / delay problem
If you are outside the US, there isn't much you can do since the x100p card was specifically designed to operate with US 600 ohm impedance pstn lines. If you have a x100p clone, it is likely the problem. Replace it with something capable of matching the pstn impedance for whatever country you are located in. Loading wcfxo with option opermode=1 should select CTR21 mode, which according to wcfxo.c (in zaptel) should be used in Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and UK. The default is to use FCC mode, which is used in the US. The x100p does not have a programable chipset for impedance matching. That parameter was intended for the TDM card. Ok, maybe it isn't programmable, but there seems to be two different DAA's used in the clones[1]. Si3012 for the US market and Si3014 for the global market. I assume this has to do with the line interface. I bought a clone in Sweden that has the Si3014. It should support the line impedance used in Sweden I think, since it's marked with the following label: Clas Ohlson AB Modell/Malli: AMI-IE92 Art.nr/nro:32-2055 CE For use in Sweden, Norway and Finland. To be connected to the public switched telefone network. Hereby, Clas Ohlson, declares that this modem is in complience with the essential requirements and other relevant provisions of Directive 1999/5/EC. [1]http://www.intel.com/design/modems/linecard.htm So, does that apply to the OP (I think he posted from Germany, but I could be wrong)? ___ 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] echo / delay problem
[channels] echocancel= 16 echocancelwhenbridged = yes echotraining = no ;; yes rxgain= -2.0 txgain= -2.0 If you are in the US and using a digium x100p, then try the following: echotraining=800 echocancel=yes (The values you are showing above aren't even valid parameters.) Why not? I don't see what wrong with those values? Can you explain please? ___ 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] echo / delay problem
I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried and tried for 2 weeks it is rather annoying and irating to hear this delay/echo I would call it a delay since you can hear the end of the sentence repeat over and over. Also every now and again it sounds like a underwater submarine with ping and all. Thanks in advance Barry [channels] language = en context = inbound signalling= fxs_ks usecallerid = yes hidecallerid = no callwaiting = yes usecallingpres= yes callwaitingcallerid = yes threewaycalling = yes echocancel= 16 echocancelwhenbridged = yes echotraining = no ;; yes rxgain= -2.0 txgain= -2.0 musiconhold = default channel = 1 context = intern signalling= fxo_ks callwaiting = yes usecallerid = yes echotraining = no ;; yes echocancel= 16 channel = 2 ___ 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
