Re: [asterisk-users] Echo problem in the analoge lines
On Tue, Aug 16, 2011 at 05:38:37PM -0700, bilal ghayyad wrote: The current dahdi version is: PBX-FF*CLI dahdi show version DAHDI Version: 2.4.1.2 Echo Canceller: Well, the output of the dahdi_cfg as shown below, it declares there is invalid argument. But, really I tried to change the configuration in the systems.conf from fxoks=1-16 to fxsks=1-16 but did not work at all !! I know that FXO ports needs FXS signaling .. But I do not know why this message appears with me: [root@PBX-FF /]# dahdi_cfg DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span This message is appearing because, as you pointed out above, you need fxsks=1-16 signalling specified for the FXO modules that are installed on your card. What is the output of dahdi_cfg when the signalling is configured properly? Also, if you're not familiar with dahdi_genconf, you might want to give that a try: $ rm /etc/dahdi/system.conf $ modprobe -r wctdm24xxp $ modprobe wctdm24xxp $ dahdi_genconf system you should have a resonable configuration in /etc/dahdi/system.conf now... $ dahdi_cfg -vvf Because dahdi_cfg is detecting an error in your configuration file, it is *not* attaching the mg2 echo canceller to your channel. I would make sure that dahdi_cfg runs without any errors before starting Asterisk. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Echo problem in the analoge lines
OK, I can buy echo canceller from Digium and how will be installed in the digium card? Or it is a hardware? Currently I am reading a message at the consol that Unable to enable the echo canceller .. does this means that Digium card that I have is not supporting? This is the output of the dahdi_scan: [root@PBX-FF asterisk]# dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM2400P Board 1 name=WCTDM/0 manufacturer=Digium devicetype=Wildcard TDM2400P location=PCI Bus 16 Slot 05 basechan=1 totchans=24 irq=20 type=analog port=1,FXO port=2,FXO port=3,FXO port=4,FXO port=5,FXO port=6,FXO port=7,FXO port=8,FXO port=9,FXO port=10,FXO port=11,FXO port=12,FXO port=13,FXO port=14,FXO port=15,FXO port=16,FXO And thanks in advance for the help. Regards Bilal - To overcome the echo problem... Digium sells 'High Performance Echo Cancellation' http://www.digium.com/en/products/software/hpec.php Also, the 'Oslec Echo Canceller' http://www.rowetel.com/blog/?page_id=454 is supposed to be pretty good stuff. [un]Fortunately, I've never had the need to try either. -- -- _ -- 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] Echo problem in the analoge lines
On Tue, Aug 16, 2011 at 05:34:58AM -0700, bilal ghayyad wrote: OK, I can buy echo canceller from Digium and how will be installed in the digium card? Or it is a hardware? If you're using a TDM2400 you can buy a hardware echocan module or HPEC (which is a proprietary software echocan) if you desire (or need), however... Currently I am reading a message at the consol that Unable to enable the echo canceller .. does this means that Digium card that I have is not supporting? ...either of those options won't resolve the Unable to enable the echo canceller. After you set 'echocanceller=mg2,1-24' in your /etc/dahdi/system.conf file, did you run dahdi_cfg? Also, what is the output of 'cat /proc/dahdi/1'? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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] Echo problem in the analoge lines
The current dahdi version is: PBX-FF*CLI dahdi show version DAHDI Version: 2.4.1.2 Echo Canceller: Well, the output of the dahdi_cfg as shown below, it declares there is invalid argument. But, really I tried to change the configuration in the systems.conf from fxoks=1-16 to fxsks=1-16 but did not work at all !! I know that FXO ports needs FXS signaling .. But I do not know why this message appears with me: [root@PBX-FF /]# dahdi_cfg DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) Selected signaling not supported Possible causes: FXO signaling is being used on a FXO interface (use a FXS signaling variant) RBS signaling is being used on a E1 CCS span Signaling is being assigned to channel 16 of an E1 CAS span Also I have the following lines in the systems.conf: echocanceller=mg2,1-16 fxoks=1-16 And I have the following lines in the chan_dahdi.conf: context=IncomingPSTN signalling=fxs_ks rxgain=0.0 txgain=0.0 channel = 1-16 group=1 channel = 1-16 The output of the command 'cat /proc/dahdi/1' is: [root@PBX-FF asterisk]# cat /proc/dahdi/1 Span 1: WCTDM/0 Wildcard TDM2400P Board 1 (MASTER) 1 WCTDM/0/0 FXSKS (In use) 2 WCTDM/0/1 FXSKS (In use) 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 FXSKS (In use) 5 WCTDM/0/4 FXSKS (In use) RED 6 WCTDM/0/5 FXSKS (In use) RED 7 WCTDM/0/6 FXSKS (In use) RED 8 WCTDM/0/7 FXSKS (In use) RED 9 WCTDM/0/8 FXSKS (In use) RED 10 WCTDM/0/9 FXSKS (In use) RED 11 WCTDM/0/10 FXSKS (In use) RED 12 WCTDM/0/11 FXSKS (In use) RED 13 WCTDM/0/12 FXSKS (In use) RED 14 WCTDM/0/13 FXSKS (In use) RED 15 WCTDM/0/14 FXSKS (In use) RED 16 WCTDM/0/15 FXSKS (In use) RED 17 WCTDM/0/16 Reserved 18 WCTDM/0/17 Reserved 19 WCTDM/0/18 Reserved 20 WCTDM/0/19 Reserved 21 WCTDM/0/20 Reserved 22 WCTDM/0/21 Reserved 23 WCTDM/0/22 Reserved 24 WCTDM/0/23 Reserved So what do u advise? Regards Bilal OK, I can buy echo canceller from Digium and how will be installed in the digium card? Or it is a hardware? If you're using a TDM2400 you can buy a hardware echocan module or HPEC (which is a proprietary software echocan) if you desire (or need), however... Currently I am reading a message at the consol that Unable to enable the echo canceller .. does this means that Digium card that I have is not supporting? ...either of those options won't resolve the Unable to enable the echo canceller. After you set 'echocanceller=mg2,1-24' in your /etc/dahdi/system.conf file, did you run dahdi_cfg? Also, what is the output of 'cat /proc/dahdi/1'? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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
[asterisk-users] Echo problem in the analoge lines
Hi All; To overcome the echo problem, what mainly I have to do in the configuration other than the following line in the system.conf under dahdi directory? echocanceller=mg2,1-16 1) How can I know if the digium card supporting echo cancellator? 2) If I am getting a message in the consol that unable to enable the echo cancelator, then what does it means? The hardware is not supporting echo cancellation or there is a software problem? Regards Bilal -- _ -- 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] Echo problem in the analoge lines
On Sat, 13 Aug 2011, bilal ghayyad wrote: To overcome the echo problem... Digium sells 'High Performance Echo Cancellation' http://www.digium.com/en/products/software/hpec.php Also, the 'Oslec Echo Canceller' http://www.rowetel.com/blog/?page_id=454 is supposed to be pretty good stuff. [un]Fortunately, I've never had the need to try either. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Echo problem in VoIP-calls
Hello Gareth, echo also appears when making calls with a SIP phone. These are outgoing calls. Another site now also gives feedback on echo, telling they sometimes also have echo on outgoing calls and if they recall right then sometimes also on incoming calls (coming from a queue). This one site that now also gives feedback on echo has a fiber optic internet connection, so I don't think the latency plays a role here. I will now turn off the buffer in sip.conf and see how this goes... I hope I can resolve this echo-problem. Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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
[asterisk-users] Echo problem in VoIP-calls
Hello list, this is the setup : analogue phone -- Grandstream GXW4008 -- Linksys WAG160N -- Asterisk-server (public) and Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public) When calling with an analogue phone + Grandstream GXW and also when calling with the Zoiper softphone, we experience echo on both calling parties. Because the echo is there with the analogue phone AND with the Zoiper, I conclude that it is not the Grandstream GXW4008 gateway that is causing the echo. Could it be the router ??? These are the VoIP speed test results : VoIP test statistics Jitter: you -- server: 4.2 ms Jitter: server -- you: off Packet loss: you -- server: 0.0 % Packet loss: server -- you: off Packet discards: 0.0 % Packets out of order: 0.0 Kind regards, Jonas. -- _ -- 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] Echo problem in VoIP-calls
Jonas Kellens wrote: Hello list, this is the setup : analogue phone -- Grandstream GXW4008 -- Linksys WAG160N -- Asterisk-server (public) and Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public) When calling with an analogue phone + Grandstream GXW and also when calling with the Zoiper softphone, we experience echo on both calling parties. Because the echo is there with the analogue phone AND with the Zoiper, I conclude that it is not the Grandstream GXW4008 gateway that is causing the echo. Could it be the router ??? These are the VoIP speed test results : VoIP test statistics Jitter: you -- server: 4.2 ms Jitter: server -- you: off Packet loss: you -- server: 0.0 % Packet loss: server -- you: off Packet discards: 0.0 % Packets out of order: 0.0 Kind regards, Jonas. Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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] Echo problem in VoIP-calls
Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. I have the same Grandstream GXW 4008 gateway with 5 analoge phones attached in another environment and there, there are no echo-problems. Can't say the analogue phones that are being used there are top of the bill, rather cheap stuff actually. When calling through the analogue phone line, there is no echo (and it seems therefore that the analogue phones that are being used meet the quality standards). The only network-element that is different in the 2 environments is the router... Jonas. On 06/30/2010 11:06 AM, Gareth Blades wrote: Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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] Echo problem in VoIP-calls
Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. Jonas Kellens wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. I have the same Grandstream GXW 4008 gateway with 5 analoge phones attached in another environment and there, there are no echo-problems. Can't say the analogue phones that are being used there are top of the bill, rather cheap stuff actually. When calling through the analogue phone line, there is no echo (and it seems therefore that the analogue phones that are being used meet the quality standards). The only network-element that is different in the 2 environments is the router... Jonas. On 06/30/2010 11:06 AM, Gareth Blades wrote: Echo cannot be caused by a router. The zoipher softphone is probably being used with a headset and I suspect the microphone is picking up the sounds from the earphones resulting in echo. Try turning down the earphone volume to see if this helps. If it does invest in some better headphone preferably ones where the microphone has built in background noise cancelation. For the analogue phone it could be a similar issue. Some phones are better than others. Cant you use a proper SIP phone? They work so much better. -- _ -- 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] Echo problem in VoIP-calls
On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. It will present it self on the analogue phone when it is introduced in Zoiper. As the orignal respondent said, routers dont introduce echo. -- _ -- 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] Echo problem in VoIP-calls
Hello, I stated in my first post that both ends hear an echo when one speaks to the other... The only place where echo cancellation is being applied is in the Asterisk server. I have the following in sip.conf : ;-- JITTER BUFFER CONFIGURATION -- jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. jbforce = no; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to no. ;--- Thank you for your replies. Kind regards. Jonas. On 06/30/2010 11:36 AM, Gareth Blades wrote: Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. -- _ -- 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] Echo problem in VoIP-calls
Thats the jitter buffer. It has no effect on echo. So you get echo when calling from the softphone to the analogue phone? What about when one of those calls somewhere else? What if they call a regular telephone number? How do you connect in order to send calls to normal phone numbers? Jonas Kellens wrote: Hello, I stated in my first post that both ends hear an echo when one speaks to the other... The only place where echo cancellation is being applied is in the Asterisk server. I have the following in sip.conf : ;-- JITTER BUFFER CONFIGURATION -- jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a ; SIP channel. Defaults to no. An enabled jitterbuffer will ; be used only if the sending side can create and the receiving ; side can not accept jitter. The SIP channel can accept jitter, ; thus a jitterbuffer on the receive SIP side will be used only ; if it is forced and enabled. jbforce = no; Forces the use of a jitterbuffer on the receive side of a SIP ; channel. Defaults to no. ;--- Thank you for your replies. Kind regards. Jonas. On 06/30/2010 11:36 AM, Gareth Blades wrote: Routers wont cause echo. In order for them to do so they would have to store the outbound voice traffic, delay it and then mix it into the inbound voice. Telephones inherently cause echo. For domestic calls the audio path is normally so short that any echo arrives back so quick the human ear does not detect it. For international calls the telco uses expensive echo cancelation technology. When you switch to voip you are often suddenly introducing a much larger delay so any excho which was present before but not noticed suddenly becomes noticable. You need to analyse the audio path your calls are taking, where the delays are being introduced and where echo cancelation is being applied. You also havent stated which end of the conversation is hearing the echo. -- _ -- 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] Echo problem in VoIP-calls
Hello, I did not say that the analogue phone calls the Zoiper softphone or vica versa. Calls are made to from the Zoiper to an external number like a cellphone. Calls are also made from the analogue phone to external numbers like an international number in Holland... Jonas. On 30 June 2010 10:28, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I also thought about echo because the Zoiper softphone is used with a headset. But that didn't explain why the echo also appeared on the analogue phone + gateway. It will present it self on the analogue phone when it is introduced in Zoiper. As the orignal respondent said, routers dont introduce echo. -- _ -- 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] Echo problem in VoIP-calls
On 06/30/2010 12:20 PM, Gareth Blades wrote: So you get echo when calling from the softphone to the analogue phone? From softphone to analogue phone is echo. What if they call a regular telephone number? Calling to a cellphone number or a fixed number on another Telco-network : echo How do you connect in order to send calls to normal phone numbers? The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks So basically, there's always an echo. Jonas. -- _ -- 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] Echo problem in VoIP-calls
Jonas Kellens wrote: On 06/30/2010 12:20 PM, Gareth Blades wrote: So you get echo when calling from the softphone to the analogue phone? From softphone to analogue phone is echo. What if they call a regular telephone number? Calling to a cellphone number or a fixed number on another Telco-network : echo How do you connect in order to send calls to normal phone numbers? The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks So basically, there's always an echo. Jonas. By ITSP do you mean a SIP provider? -- _ -- 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] Echo problem in VoIP-calls
Internet Telephony Service Provider = SIP provider. The company that connects the Asterisk-server via a SIP trunk with the other networks like GSM, analogue carriers... Jonas. By ITSP do you mean a SIP provider? -- _ -- 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] Echo problem in VoIP-calls
On 30 Jun 2010, at 13:48, Gareth Blades wrote: By ITSP do you mean a SIP provider? ITSP: Internet Telephony Service Provider S -- _ -- 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] Echo problem in VoIP-calls
Hi! The network setup is : analogue+GXW / softphone -- Linksys WAG160N -- Asterisk server -- ITSP -- other networks Do it step-by-step: Take the Asterisk server out of the equation, i.e. call the destination directly with your softphone or the Grandstream ATA and see if that removes the echo. That fact that both sides are hearing echo is a bit unusual - especially when calling a mobile destination things should be different. Check twice that the analog devices in the setup are ok, and replace them for a test if you can. You could also test with a destination that is run by a different operator (or is located in a different country). Another test: Use the Echo() application on Asterisk and call it from both sides. Also: You could capture the traffic and look at it with Wireshark, the delay/latency in particular. Philipp P.S.: I do think a jitter buffer matters for echo, simply because it introduces an additional delay. However the Asterisk server should not use its jitter buffer because jbforce is set to no and the Asterisk server is not the final endpoint (it only sits in between). -- _ -- 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] Echo problem in VoIP-calls
Jonas Kellens wrote: Internet Telephony Service Provider = SIP provider. The company that connects the Asterisk-server via a SIP trunk with the other networks like GSM, analogue carriers... Jonas. By ITSP do you mean a SIP provider? Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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] Echo problem in VoIP-calls
Gareth, multiple users/SIP-accounts use this asterisk server from many locations. Like I said: in another location with a similar setup, there are no echo-complaints on received or made calls. If you say that it has nothing to do with the Cisco-router, I don't really know what to go looking for... I will take your advise and try with a SIP-phone (snom 320). What do I do if : 1. I also have echo with a SIP-phone ? 2. I do not have echo with a SIP-phone ? Jonas. On 06/30/2010 03:52 PM, Gareth Blades wrote: Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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] Echo problem in VoIP-calls
Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. Jonas Kellens wrote: Gareth, multiple users/SIP-accounts use this asterisk server from many locations. Like I said: in another location with a similar setup, there are no echo-complaints on received or made calls. If you say that it has nothing to do with the Cisco-router, I don't really know what to go looking for... I will take your advise and try with a SIP-phone (snom 320). What do I do if : 1. I also have echo with a SIP-phone ? 2. I do not have echo with a SIP-phone ? Jonas. On 06/30/2010 03:52 PM, Gareth Blades wrote: Thats where I believe the problem lies. You are sending audio to them and they are putting it onto the PSTN network. When the audio comes back from the PSTN it has echo on it. They are not performing echo cancellation. If it is an international call from the ITSP's perspective then teh network operator should be performing echo cancelation anyway. If its a national call then the telco doesnt perform echo cancelation but the ITSP should do it themselves. The only time this is not needed is if the phones have a very low delay to the ITSP but since this is normally not the case echo cancelation must be performed at this point. -- _ -- 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] Echo problem in VoIP-calls
Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] Echo problem in VoIP-calls
The harm in any of these settings is environmentally controlled. What does no harm in one setup can be a deal breaker on a smaller machine or slightly different technology. How harmful or harmless jbenable is depends on your hardware and what your other settings are. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Wednesday, June 30, 2010 9:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo problem in VoIP-calls Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] Echo problem in VoIP-calls
Yes if you have a link where there is a lot of jitter it may affect the call quality. I would try turning it off to see if it cures the problem and if it does then you can restore the setting and implement a workaround. Jonas Kellens wrote: Will turning off the jitter buffer affect the quality of the other calls ?? jbenable = no I must say I'm not really into these jitter-settings in asterisk. I made jbenable=yes as it can do no harm... Jonas. On 06/30/2010 04:24 PM, Gareth Blades wrote: Try the SIP phone. If it is better then you might try looking to see if there are any echo cancelation settings on the softphone or analogue adapter you can change. Try turning echo cancelation off aswell since if there are two running they can interfere with each other and make the situation worse. If you hear echo on that phone then it might be that the network connection from that location has a higher latency making the echo far more noticeable. If the other party you are connecting to hears echo then this could be down to the phone or the jitter buffer. If you start with a small jitter buffer the echo cancelation will train to that but if you get increased jitter the buffer will grow and add an additional delay to the audio. Often echo cancelation only trains at the start of a call. Maybe try disabling the jitter buffer. -- _ -- 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] Echo problem
On 12/20/06, Jason Bachman [EMAIL PROTECTED] wrote: As I understand it, the echo cancelers in Asterisk only work with the Analog cards (FXS/FXO). Not true, echo is caused by any number of things in the voice network, so Asterisk will echo cancel any Zap device. We use it to cancel ISDN2e and ISDN30 E1 lines very successfully. If you are getting echo on a digital line, there is a problem with either a DAC, the T1 clocking, or you are getting bit errors. Again, not true. The echo is (mostly) not caused in or by asterisk, it is caused out there. Even if a call is digital end-to-end, there is the posibility of acoustic echo in the handsets. Of course the above problems might also cause echo, but I expect they would also cause a log full of errors :) You have a Switch in the middle - perhaps the switch is doing doing digital-analog conversions instead of sending the digital data straight through. The cause of the echo could very well be there, and the echo cancelers (even if they worked on a digital line) would not help because the cause of the echo is somewhere else, not at the Digium card. Check your Tadiran switch for any echo cancel options. I'm not familiar with that switch so I am no help to you on that, but I am pretty sure that its not the Digium card or Asterisk. I agree, that is a very good candidate. AD/DA conversions in this device would IMHO make it responsible for cancelling any resultant echo, and the conversions could indeed add significant delay. Regards, --Jason Bachman Scott Gifford wrote: Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. 1.5 to 2 seconds. That is a HUGE delay. echo delay is normally measured in tens or perhaps hundreds of milliseconds, and you are unlikely to find a software EC that can deal with a 1.5 to 2 second delay! This sounds as if there is something very broken in the voice network, causing huge amounts of delay. As suggested above, check the intermediate switch. [snip] We have done loopback tests with the Digium card with a loop plug in it. What were the results? Cheers, Steve ___ --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] Echo problem
We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/doc s-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. Checkout the Digium KB: http://kb.digium.com/19/ You will see a suggestion to adjust the gain levels as well. Even though the echo is there, it helps to not make it noticeable to the users. I just found this as well, although they are trying to sell their product at the same time, it helps explain echo and some steps in Asterisk for reducing echo: http://www.xorcom.com/pdfs/AB007_Echo.pdf Michael L. Young ___ --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] Echo problem
Steve Davies [EMAIL PROTECTED] writes: Scott Gifford [EMAIL PROTECTED] writes: [...] 1.5 to 2 seconds. That is a HUGE delay. echo delay is normally measured in tens or perhaps hundreds of milliseconds, and you are unlikely to find a software EC that can deal with a 1.5 to 2 second delay! This sounds as if there is something very broken in the voice network, causing huge amounts of delay. As suggested above, check the intermediate switch. What's interesting is the lines come in via 2 PRI lines, and most calls go out via analog lines to people's desks and a voicemail system. These lines all work fine. So the problem likely isn't in the PSTN and isn't an inherent flaw with the switch, though it could be the T1 card connected to our Asterisk server or its configuration. It seems the problem is either on the Tadiran switch or the Asterisk server. Unfortunately we don't have a good way to determine which, since we don't have another switch to try, or another device to replace the Digium server. [snip] We have done loopback tests with the Digium card with a loop plug in it. What were the results? Oh, sorry, I should have said: These tests were successful. Scott. ___ --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
[asterisk-users] Echo problem
Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. We have done loopback tests with the Digium card with a loop plug in it. We're a bit stumped as to what to try next. Any suggestions or advice? Thanks ---Scott. ___ --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] Echo problem
Hi, Did you try to increase echotraining ?? echo training = 800 .. @++ Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. We have done loopback tests with the Digium card with a loop plug in it. We're a bit stumped as to what to try next. Any suggestions or advice? Thanks ---Scott. ___ --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] Echo problem
As I understand it, the echo cancelers in Asterisk only work with the Analog cards (FXS/FXO). If you are getting echo on a digital line, there is a problem with either a DAC, the T1 clocking, or you are getting bit errors. You have a Switch in the middle - perhaps the switch is doing doing digital-analog conversions instead of sending the digital data straight through. The cause of the echo could very well be there, and the echo cancelers (even if they worked on a digital line) would not help because the cause of the echo is somewhere else, not at the Digium card. Check your Tadiran switch for any echo cancel options. I'm not familiar with that switch so I am no help to you on that, but I am pretty sure that its not the Digium card or Asterisk. Regards, --Jason Bachman Scott Gifford wrote: Hello, We're in the process of setting up an Asterisk server, and are having echo problems. We have a Digium TE110P, and have tried the MG and MARK2 AGGRESSIVE echo cancellers, with a variety of gain levels and training times, and with both trunk and 1.2 branch versions of Zaptel, Libpre, and Asterisk. In all cases, callers from the PSTN hear their own voice echoed back after 1.5-2 seconds; none of these adjustments made a difference, except adjusting gain made the echo quieter. We followed these instructions in trying to eliminate echo: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html Our lines come in from the telco in a PRI, then connect to a Tadiran switch which hands the lines off to Asterisk over a T1 card. We have done loopback tests with the Digium card with a loop plug in it. We're a bit stumped as to what to try next. Any suggestions or advice? Thanks ---Scott. ___ --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] Echo problem
pixiesfr [EMAIL PROTECTED] writes: Hi, Did you try to increase echotraining ?? echo training = 800 .. Yes, I tried 800, 1200, and 2000; none seemed to make any difference. Thanks! ---Scott. ___ --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
[asterisk-users] Echo problem with TDM440P and ADSL Line
Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of them have a ADSL splitter. The location are Spain, i load the tdm driver with opermode SPAIN, i read that this will set the correct impedance for the line. I compile first time zaptel with ECHO_CAN_MARK, but also i probe MARK2, MARK3, KB1 and ECHO_CAN_MG2, with no result for the Line that have the ADSL. The aggresive make good with the adsl line, but get really extrange sounds across the conversation. I don't know the milliwat test line here on spain to setup the gains correctly (If any body knowsm send it, :) ), so i call from one line to the anoter one with a millivat app, so i notice that when i connect a analog phone to the line 1 press 5 to get a clear line, and talk something i hear me but very very low, but when i make this on the line 2 i hear me more high. So i use the fxotune that come with zaptel, with no results for this line, so i get the lastest from CVS that comes with a patch that can dump the waves to see what happend, i get 1-3 % of echo on line 1, and arround 30-25 % on the Line 2, i read that if i have more than 10% of echo the soft echo canceller can't make her work due to it can't determine the echo signal. So i read (all googling) that with ADSL there is a some tricky to solve, but no information. It's posible to solve this problem? i suspect that i not alone with this conf, analog +adsl? Any help is apreciated, i sepnt a lot of hours to solve this with no success. Excuse my english. Regards. ___ --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] Echo Problem with T411P
More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! Okay, to be complete in my answers: No I do not get more than 128ms delay caused by European routing (I only threw that in as an example anyway), but asterisk's software cancellers only cancel 16ms, any more than that seems fairly buggy, and eats CPU. On the other hand, if that is a satellite link on span 3, you could easily get latency in excess of 1 second, which it should be the provider's responsibility to cancel, not the end user's IMHO. I also agree that the sangoma EC is excellent :) Do we know what E1/T1 hardware is in use here, and whether hardware EC is available? Steve ___ --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] Echo Problem with T411P
Hi Steve, Thank you for your answers. First of all span 3 is not a satellite link and no echo occurs when I connect this line to another pbx with HW EC feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I have to do something to enable EC for this card ? Idris -Original Message- From: Steve Davies [mailto:[EMAIL PROTECTED] Sent: Monday, June 19, 2006 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo Problem with T411P More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! Okay, to be complete in my answers: No I do not get more than 128ms delay caused by European routing (I only threw that in as an example anyway), but asterisk's software cancellers only cancel 16ms, any more than that seems fairly buggy, and eats CPU. On the other hand, if that is a satellite link on span 3, you could easily get latency in excess of 1 second, which it should be the provider's responsibility to cancel, not the end user's IMHO. I also agree that the sangoma EC is excellent :) Do we know what E1/T1 hardware is in use here, and whether hardware EC is available? Steve ___ --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] Echo Problem with T411P
On 6/19/06, Idris AVCI [EMAIL PROTECTED] wrote: Hi Steve, Thank you for your answers. First of all span 3 is not a satellite link and no echo occurs when I connect this line to another pbx with HW EC feature. I use TE411P with hardware EC and asterisk version 1.2.5. Do I have to do something to enable EC for this card ? :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. Best of luck. Steve ___ --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] Echo Problem with T411P
- Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. -- Kevin P. Fleming Senior Software Engineer Digium, Inc. ___ --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] Echo Problem with T411P
Kevin P. Fleming wrote: - Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. How can you detect if the HW echo can is enabled? Is it console output during module load or something else? ___ --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] Echo Problem with T411P
On 6/19/06, Mike Fedyk [EMAIL PROTECTED] wrote: Kevin P. Fleming wrote: - Steve Davies [EMAIL PROTECTED] wrote: :) Now you've defeated me. I imagine that you need to do something to enable EC on that card, but it is not a card I know, so I'll leave it to someone who knows the card to offer any suggestions. The only requirement is that 'echocancel=yes' is present in zapata.conf for those channels. If the hardware echo canceler is present and enabled, then it will be used instead of the software canceler for those channels. How can you detect if the HW echo can is enabled? Is it console output during module load or something else? Yes. You'll see messages about a VPM (Voice Processing Module) getting initialized. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Echo Problem with T411P
On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote: Steve Davies wrote: We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. Steve ___ --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] Echo Problem with T411P
I forgot to mention one thing. I don't know if it changes anything. Internal users are connected to another PBX which connects to asterisk over SIP. Echo is always at internal user side. External user never hears echo. External User -- PSTN -- Asterisk -- SIP -- CIC -- Internal User -Original Message- From: Steve Davies [mailto:[EMAIL PROTECTED] Sent: Friday, June 16, 2006 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Echo Problem with T411P On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote: Steve Davies wrote: We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. Steve ___ --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] Echo Problem with T411P
Steve Davies wrote: On 6/15/06, Mike Fedyk [EMAIL PROTECTED] wrote: Steve Davies wrote: We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? 128 = 128 taps = 16ms of 8KHz audio, so no, not more that 128ms, but definitely more than 16ms. No, 128ms = 1024 taps Like what sangoma offers. Ding, Ding, Ding, Ding! ___ --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
[Asterisk-Users] Echo Problem with T411P
Hello, There are 3 PRIs connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callerid=asreceived echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 group = 1 signalling=pri_cpe context=default channel = 1-15 channel = 17-31 group = 2 signalling=pri_cpe context=default channel = 32-46 channel = 48-62 group = 3 signalling=pri_cpe context=Satelco channel = 63-77 channel = 79-93 group = 4 signalling=pri_cpe context=default channel = 94-108 channel = 110-124 zaptel.conf-- span = 1,1,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 span = 2,1,0,ccs,hdb3 bchan = 32-46,48-62 dchan = 47 span = 3,1,0,ccs,hdb3,crc4 bchan = 63-77,79-93 dchan = 78 span = 4,1,0,ccs,hdb3 bchan = 94-108,110-124 dchan = 109 loadzone = nl defaultzone = nl ___ --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] Echo Problem with T411P
Title: Re: [Asterisk-Users] Echo Problem with T411P Try setting echocancelwhenbridged=yes. Also, in your zaptel, you only need to define one span as the clock master, so should be like zaptel.conf-- span = 1,1,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 span = 2,0,0,ccs,hdb3 bchan = 32-46,48-62 dchan = 47 span = 3,0,0,ccs,hdb3,crc4 bchan = 63-77,79-93 dchan = 78 span = 4,0,0,ccs,hdb3 bchan = 94-108,110-124 dchan = 109 loadzone = nl defaultzone = nl Thanks, James On 6/15/06 6:29 AM, Idris AVCI [EMAIL PROTECTED] wrote: Hello, There are 3 PRIs connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Zapata.conf -- [channels] context=default switchtype=euroisdn usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes callerid=asreceived echocancel=yes echotraining=yes rxgain=0.0 txgain=0.0 group = 1 signalling=pri_cpe context=default channel = 1-15 channel = 17-31 group = 2 signalling=pri_cpe context=default channel = 32-46 channel = 48-62 group = 3 signalling=pri_cpe context=Satelco channel = 63-77 channel = 79-93 group = 4 signalling=pri_cpe context=default channel = 94-108 channel = 110-124 zaptel.conf-- span = 1,1,0,ccs,hdb3,crc4 bchan = 1-15,17-31 dchan = 16 span = 2,1,0,ccs,hdb3 bchan = 32-46,48-62 dchan = 47 span = 3,1,0,ccs,hdb3,crc4 bchan = 63-77,79-93 dchan = 78 span = 4,1,0,ccs,hdb3 bchan = 94-108,110-124 dchan = 109 loadzone = nl defaultzone = nl ___ --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] Echo Problem with T411P
On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote: Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Have you verified that the provider on span 3 is not using some high latency routing? The configuration line context=Satelco suggests a satellite company? They should do the echo cancelling on your behalf if they have high latency routes as the asterisk EC will never keep up under those circumstances. We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. Cheers, Steve ___ --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] Echo Problem with T411P
Steve Davies wrote: On 6/15/06, Idris AVCI [EMAIL PROTECTED] wrote: Hello, There are 3 PRI's connected to the card each from different operators. Especially echo occured on span 3 is really annoying. Configuration files are as follows. Is there something wrong in conf ? Have you verified that the provider on span 3 is not using some high latency routing? The configuration line context=Satelco suggests a satellite company? They should do the echo cancelling on your behalf if they have high latency routes as the asterisk EC will never keep up under those circumstances. We have even experienced problems within Europe where providers route national calls via international routes to save money. This adds significant latency and makes any echo so heavily delayed that asterisk cannot remove it. More than 128ms? ___ --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] echo problem + choppy sound
Look also at AudioFrames setting on your phone. I read that it needs to match 20ms packet size of Asterisk packets and it depends from codec you use. Mimmus ___ --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] echo problem + choppy sound
Have you tried with apic turned off? And, on another note, our system had bad sound (you might describe it as choppy) with acpi enabled. Do you have access to a milliwatt test line? Moj sdgesa gaeharth wrote: thanks for the info. it is not sharing an irq: 0: 59840409 59803082IO-APIC-edge timer 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd:usb1 14:21418512143209IO-APIC-edge ide0 177: 111558 111273 IO-APIC-level aic7xxx 185: 15 0 IO-APIC-level aic7xxx 193: 736328 748953 IO-APIC-level eth0 201: 239290099 239259220 IO-APIC-level wctdm NMI: 0 0 LOC: 119645889 119645888 ERR: 0 MIS: 0 I checked the switch. The net connection is running at full duplex: FastEthernet0/15 is up, line protocol is up (connected) Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f) MTU 1500 bytes, BW 10 Kbit, DLY 100 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulatio n ARPA, loopback not set Keepalive set (10 sec) Full-duplex, 100Mb/s, media type is 100BaseTX input flow-control is unsupported output flow-control is unsupported ARP type: ARPA, ARP Timeout 04:00:00 Last input never, output 00:00:03, output hang never Last clearing of show interface counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: fifo Output queue: 0/40 (size/max) 5 minute input rate 31000 bits/sec, 15 packets/sec 5 minute output rate 32000 bits/sec, 15 packets/sec 679924 packets input, 225898296 bytes, 0 no buffer Received 3803 broadcasts (0 multicast) 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored 0 watchdog, 5 multicast, 0 pause input 0 input packets with dribble condition detected 689110 packets output, 145860377 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 babbles, 0 late collision, 0 deferred 0 lost carrier, 0 no carrier, 0 PAUSE output 0 output buffer failures, 0 output buffers swapped out */Rich Adamson [EMAIL PROTECTED]/* wrote: Based only on what I see below (from previous posts), it sounds like you have two separate issues going on: 1) echo, and, 2) choppy sound. Those should be analyzed as two problems (not one). You will find plenty of posts in the archives relative to both. In general terms, the choppy audio m ost often is caused by shared IRQ's when using a x100p or TDM400 card, and sometimes from a misconfigured ethernet nic on the asterisk machine. For the nic card, ensure you are running full duplex on the nic and whatever the nic is plugged in to. Both need to be the same (half duplex will work in a low usage environment, but full duplex is preferred.) For the IRQ issue (and we are all assuming you are using a TDM04b card since you really didn't say), do a 'cat /proc/interrupts' and make sure your TDM card is on its own IRQ. If it is shared with other devices, it is likely the cause for choppy audio. You'll see the TDM driver wctdm on that list. If it is shared, then move the TDM card from one pci slot to another to get it on its own IRQ. The echo problem is going to be almost aways related to too high of gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as others have already noted. Try reducing those to 0 and restart asterisk. Then increase the values (if needed) by increments of 2 until you find a balance between low volume and echo. I'd suggest doing that after resolving the IRQ/choppy audio issues. I have done this but I still get choppy sound and echo on some calls thanks */Giovanni Miano /* wrote: Of course, Echo is 2 types: electric and ambiental. If u gain rx o tx more than you need, its return in recive and gen echo Try to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth : Can you explain why? */Giovanni Miano /* wrote: g t; rxgain=10.0 txgain=10.0 Maybe this is a problem 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] : I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help? We are all using: Polycom 501 -- asterisk -- PSTN zapata.conf: [channels] group = 1 language=en context=incoming gt; signalling=fxs_ks switchtype=national usecallerid=yes
Re: [Asterisk-Users] echo problem + choppy sound
I have done this but I still get choppy sound and echo on some callsthanksGiovanni Miano [EMAIL PROTECTED] wrote: Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]: Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote:rxgain=10.0 txgain=10.0 Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=encontext=incoming signalling=fxs_ksswitchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yestransfer=yes canpark=yescancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.c om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.c om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Travel Find great deals to the top 10 hottest destinations!___ --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] echo problem + choppy sound
Based only on what I see below (from previous posts), it sounds like you have two separate issues going on: 1) echo, and, 2) choppy sound. Those should be analyzed as two problems (not one). You will find plenty of posts in the archives relative to both. In general terms, the choppy audio most often is caused by shared IRQ's when using a x100p or TDM400 card, and sometimes from a misconfigured ethernet nic on the asterisk machine. For the nic card, ensure you are running full duplex on the nic and whatever the nic is plugged in to. Both need to be the same (half duplex will work in a low usage environment, but full duplex is preferred.) For the IRQ issue (and we are all assuming you are using a TDM04b card since you really didn't say), do a 'cat /proc/interrupts' and make sure your TDM card is on its own IRQ. If it is shared with other devices, it is likely the cause for choppy audio. You'll see the TDM driver wctdm on that list. If it is shared, then move the TDM card from one pci slot to another to get it on its own IRQ. The echo problem is going to be almost aways related to too high of gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as others have already noted. Try reducing those to 0 and restart asterisk. Then increase the values (if needed) by increments of 2 until you find a balance between low volume and echo. I'd suggest doing that after resolving the IRQ/choppy audio issues. I have done this but I still get choppy sound and echo on some calls thanks */Giovanni Miano [EMAIL PROTECTED]/* wrote: Of course, Echo is 2 types: electric and ambiental. If u gain rx o tx more than you need, its return in recive and gen echo Try to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: Can you explain why? */Giovanni Miano [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]/* wrote: rxgain=10.0 txgain=10.0 Maybe this is a problem 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] : I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help? We are all using: Polycom 501 -- asterisk -- PSTN zapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 ___ --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] echo problem + choppy sound
thanks for the info.it is not sharing an irq: 0: 59840409 59803082 IO-APIC-edge timer 8: 1 0 IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd:usb1 14: 2141851 2143209 IO-APIC-edge ide0 177: 111558 111273 IO-APIC-level aic7xxx 185: 15 0 IO-APIC-level aic7xxx 193: 736328 748953 IO-APIC-level eth0 201: 239290099 239259220 IO-APIC-level wctdm NMI: 0 0 LOC: 119645889 119645888 ERR: 0 MIS: 0 I checked the switch. The net connection is running at full duplex:FastEthernet0/15 is up, line protocol is up (connected) Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f) MTU 1500 bytes, BW 10 Kbit, DLY 100 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulatio n ARPA, loopback not set Keepalive set (10 sec) Full-duplex, 100Mb/s, media type is 100BaseTX input flow-control is unsupported output flow-control is unsupported ARP type: ARPA, ARP Timeout 04:00:00 Last input never, output 00:00:03, output hang never Last clearing of "show interface" counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: fifo Output queue: 0/40 (size/max) 5 minute input rate 31000 bits/sec, 15 packets/sec 5 minute output rate 32000 bits/sec, 15 packets/sec 679924 packets input, 225898296 bytes, 0 no buffer Received 3803 broadcasts (0 multicast) 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored 0 watchdog, 5 multicast, 0 pause input 0 input packets with dribble condition detected 689110 packets output, 145860377 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 babbles, 0 late collision, 0 deferred 0 lost carrier, 0 no carrier, 0 PAUSE output 0 output buffer failures, 0 output buffers swapped outRich Adamson [EMAIL PROTECTED] wrote: Based only on what I see below (from previous posts), it sounds like you have two separate issues going on: 1) echo, and, 2) choppy sound. Those should be analyzed as two problems (not one).You will find plenty of posts in the archives relative to both. In general terms, the choppy audio m ost often is caused by shared IRQ's when using a x100p or TDM400 card, and sometimes from a misconfigured ethernet nic on the asterisk machine. For the nic card, ensure you are running full duplex on the nic "and" whatever the nic is plugged in to.Both need to be the same (half duplex will work in a low usage environment, but full duplex is preferred.)For the IRQ issue (and we are all assuming you are using a TDM04b card since you really didn't say), do a 'cat /proc/interrupts' and make sure your TDM card is on its own IRQ. If it is shared with other devices, it is likely the cause for choppy audio. You'll see the TDM driver wctdm on that list. If it is shared, then move the TDM card from one pci slot to another to get it on its own IRQ.The echo problem is going to be almost aways related to "too high" of gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as others have already noted. Try reducing those to 0 and restart asterisk.Then increase the values (if needed) by increments of 2 until you find a balance between low volume and echo. I'd suggest doing that "after" resolving the IRQ/choppy audio issues. I have done this but I still get choppy sound and echo on some calls thanks */Giovanni Miano /* wrote: Of course, Echo is 2 types: electric and ambiental. If u gain rx o tx more than you need, its return in recive and gen echo Try to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth : Can you explain why? */Giovanni Miano /* wrote: t; rxgain=10.0 txgain=10.0 Maybe this is a problem 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] : I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help? We are all using: Polycom 501 -- asterisk -- PSTN zapata.conf: [channels] group = 1 language=en context=incoming& gt; signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes
Re: [Asterisk-Users] echo problem + choppy sound
Well... the next step (for me anyway) would be to use Ethereal on the asterisk nic interface to ensure the sip/rtp pkts are reasonable (eg, no dropouts). If those pkts flow consistently in both directions, then there must be something impacting the wctdm interface. Do sip to sip calls sound reasonable? Is there anything else running on your asterisk box? sdgesa gaeharth wrote: thanks for the info. it is not sharing an irq: 0: 59840409 59803082IO-APIC-edge timer 8: 1 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 11: 0 0 IO-APIC-level ohci_hcd:usb1 14:21418512143209IO-APIC-edge ide0 177: 111558 111273 IO-APIC-level aic7xxx 185: 15 0 IO-APIC-level aic7xxx 193: 736328 748953 IO-APIC-level eth0 201: 239290099 239259220 IO-APIC-level wctdm NMI: 0 0 LOC: 119645889 119645888 ERR: 0 MIS: 0 I checked the switch. The net connection is running at full duplex: FastEthernet0/15 is up, line protocol is up (connected) Hardware is Fast Ethernet, address is 0013.80b7.e24f (bia 0013.80b7.e24f) MTU 1500 bytes, BW 10 Kbit, DLY 100 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulatio n ARPA, loopback not set Keepalive set (10 sec) Full-duplex, 100Mb/s, media type is 100BaseTX input flow-control is unsupported output flow-control is unsupported ARP type: ARPA, ARP Timeout 04:00:00 Last input never, output 00:00:03, output hang never Last clearing of show interface counters never Input queue: 0/75/0/0 (size/max/drops/flushes); Total output drops: 0 Queueing strategy: fifo Output queue: 0/40 (size/max) 5 minute input rate 31000 bits/sec, 15 packets/sec 5 minute output rate 32000 bits/sec, 15 packets/sec 679924 packets input, 225898296 bytes, 0 no buffer Received 3803 broadcasts (0 multicast) 0 runts, 0 giants, 0 throttles 0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored 0 watchdog, 5 multicast, 0 pause input 0 input packets with dribble condition detected 689110 packets output, 145860377 bytes, 0 underruns 0 output errors, 0 collisions, 2 interface resets 0 babbles, 0 late collision, 0 deferred 0 lost carrier, 0 no carrier, 0 PAUSE output 0 output buffer failures, 0 output buffers swapped out */Rich Adamson [EMAIL PROTECTED]/* wrote: Based only on what I see below (from previous posts), it sounds like you have two separate issues going on: 1) echo, and, 2) choppy sound. Those should be analyzed as two problems (not one). You will find plenty of posts in the archives relative to both. In general terms, the choppy audio m ost often is caused by shared IRQ's when using a x100p or TDM400 card, and sometimes from a misconfigured ethernet nic on the asterisk machine. For the nic card, ensure you are running full duplex on the nic and whatever the nic is plugged in to. Both need to be the same (half duplex will work in a low usage environment, but full duplex is preferred.) For the IRQ issue (and we are all assuming you are using a TDM04b card since you really didn't say), do a 'cat /proc/interrupts' and make sure your TDM card is on its own IRQ. If it is shared with other devices, it is likely the cause for choppy audio. You'll see the TDM driver wctdm on that list. If it is shared, then move the TDM card from one pci slot to another to get it on its own IRQ. The echo problem is going to be almost aways related to too high of gains in zapata.conf. Your rxgain=10 and txgain=10 are way too high as others have already noted. Try reducing those to 0 and restart asterisk. Then increase the values (if needed) by increments of 2 until you find a balance between low volume and echo. I'd suggest doing that after resolving the IRQ/choppy audio issues. I have done this but I still get choppy sound and echo on some calls thanks */Giovanni Miano /* wrote: Of course, Echo is 2 types: electric and ambiental. If u gain rx o tx more than you need, its return in recive and gen echo Try to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth : Can you explain why? */Giovanni Miano /* wrote: g t; rxgain=10.0 txgain=10.0 Maybe this is a problem 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] : I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help? We are all using: Polycom 501 -- asterisk -- PSTN
RE: [Asterisk-Users] echo problem + choppy sound
The best way to set gains would be to use ztmonitor (located in /usr/src/zaptel). Make a call and note your channel number. Run /usr/src/zaptel/ztmonitor channel number -v from a telnet session. check to see if your levels are too high or too low and adjust your zapata.conf accordingly. I ended up setting my TX to -4.5 to cut out the choppiness. Regards, Mark. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 14, 2006 12:11 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] echo problem + choppy sound I have done this but I still get choppy sound and echo on some calls thanks Giovanni Miano [EMAIL PROTECTED] wrote: Of course, Echo is 2 types: electric and ambiental. If u gain rx o tx more than you need, its return in recive and gen echo Try to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] : Can you explain why? Giovanni Miano [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: rxgain=10.0 txgain=10.0 Maybe this is a problem 2006/3/13, sdgesa gaeharth mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] : I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help? We are all using: Polycom 501 -- asterisk -- PSTN zapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 _ Yahoo! Mail Bring photos to life! New PhotoMail http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=39174/*http://photomail.mai l.yahoo.com makes sharing a breeze. ___ --Bandwidth and Colocation provided by Easynews.c om http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users _ Yahoo! Mail Bring photos to life! New PhotoMail http://pa.yahoo.com/*http://us.rd.yahoo.com/evt=39174/*http://photomail.mai l.yahoo.com makes sharing a breeze. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --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 _ Yahoo! Travel Find http://us.lrd.yahoo.com/_ylc=X3oDMTFscDlocTFiBF9TAzMyOTc1MDIEX3MDMjcxOTQ4MQ Rwb3MDMgRzZWMDbWFpbC1mb290ZXIEc2xrA3l0LXR0/SIG=12hqieud9/**http%3a//leisure. travelocity.com/Promotions/0,,YHOE%7c1381%7cvacs_main,00.html great deals to the top 10 hottest destinations! _ This message and any attachments are intended only for the use of the addressee and may contain information that is privileged and confidential. If the reader of the message is not the intended recipient or an authorized representative of the intended recipient, you are hereby notified that any dissemination of this communication is strictly prohibited. If you have received this communication in error, please notify us immediately by e-mail and delete the message and any attachments from your system. application/ms-tnef___ --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] echo problem + choppy sound
Of course,Echo is 2 types: electric and ambiental.If u gain rx o tx more than you need, its return in recive and gen echoTry to decrase value, try to set 0 or .. in samecase -1 -2... 2006/3/13, sdgesa gaeharth [EMAIL PROTECTED]: Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote:rxgain=10.0 txgain=10.0 Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ksswitchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yescancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.c om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --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
[Asterisk-Users] echo problem + choppy sound
I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___ --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] echo problem + choppy sound
rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ks switchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4 Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --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] echo problem + choppy sound
Can you explain why?Giovanni Miano [EMAIL PROTECTED] wrote:rxgain=10.0 txgain=10.0Maybe this is a problem2006/3/13, sdgesa gaeharth [EMAIL PROTECTED] :I still hear a slight echo of my voice when I talk with somone out the PSTN. The voice on the other end sounds very choppy and a little distorted. When I talk to other people within our office, the sound is perfect.Can some help?We are all using:Polycom 501 -- asterisk -- PSTNzapata.conf: [channels] group = 1 language=en context=incoming signalling=fxs_ksswitchtype=national usecallerid=yes hidecallerid=no callwaiting=yes musiconhold=default usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yescancallforward=yes callreturn=yes echocancel=yes echotraining=yes echocancelwhenbridged=yes rxgain=10.0 txgain=10.0 channel = 1-4Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___--Bandwidth and Colocation provided by Easynews.c om --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze. ___ --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
[Asterisk-Users] echo problem
I have installed Asterisk and when I hangup the zap channel Asterisk show this message: Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 4: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 4Feb 13 17:45:49 WARNING[1748]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 5: Red AlarmFeb 13 17:45:49 WARNING[1748]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 5Feb 13 17:45:49 NOTICE[1745]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 2Feb 13 17:45:49 NOTICE[1745]: chan_zap.c:8458 pri_dchannel: pri_shutdownFeb 13 17:45:49 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 4Feb 13 17:45:49 NOTICE[1748]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 5Feb 1 3 17:45:49 NOTICE[1745]: chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 2 What I can doing?Thanks Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___ --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
[Asterisk-Users] Echo problem not reflected with ZapBarge application
Hello everyone, How come it is possible that when we make a call (sip to pstn) using a digium tdm04b we have echo, but if we listen that conversation on an other sip phone with the asterisk zapBarge application the conversation is really clear and with no echo ? With the Digium (paid support) we tried everything that is possible, including changing the server and the digium card but without finding the source of the echo. Does the fact that the ZapBarge don't reflect the echo could indicate us that we have a problem with the local network or the phones ? Any suggestions for a good open source voip network analysis tool? Any help or comments are welcome Ken ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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 problem -- network related?
You were right and I was wrong. New sound card fixed all problems. Still can not beleive that problem was caused by audio hardware, but there we are. Thanks to all who replied. Rudolf - Original Message - From: Rob Lith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 6:14 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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 ___ 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 problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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 problem -- network related?
Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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] Echo problem -- network related?
You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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] Echo problem -- network related?
Thanks for reply. You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. Yes, this would be the logical conclusion, although it is hard to beleive given what I hear. It sound like I am talking to myself at a pretty good quality. Actually echo quality is much better than other party. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Sound card used is a built into the main board -- Gigabyte 8PIE1000 board with Realtek AC97. Not a cheap crapy board. I have tried new drivers too. I am going to try few things -- try his computer on my LAN to rule out any network related issues Try USB handset and/or difefrent sound card I wil let you all knwo when I find something out. Thanks again, RUdolf Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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 ___ 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 problem -- network related?
Thanks - always intetested in cures to the dreaded four letter word 'echo' !! Regards Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Thanks for reply. You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. Yes, this would be the logical conclusion, although it is hard to beleive given what I hear. It sound like I am talking to myself at a pretty good quality. Actually echo quality is much better than other party. What you could be getting is feedback or sidetone - so check for things like mic boost and turn that off and it may even be worth trying another sound card - we've had instances where the onboard sound of a motherboard was really crap (with 'echo' like problems) and it was resolved by disabling and putting in the Creative card... Sound card used is a built into the main board -- Gigabyte 8PIE1000 board with Realtek AC97. Not a cheap crapy board. I have tried new drivers too. I am going to try few things -- try his computer on my LAN to rule out any network related issues Try USB handset and/or difefrent sound card I wil let you all knwo when I find something out. Thanks again, RUdolf Rob On 8/14/05, Rudolf Ladyzhenskii [EMAIL PROTECTED] wrote: Hi, I am using SIP phone (Polycom 300). Echo is present even if other party has sound hardware disconnected. It is definetely network and/or PC setup issue, but is not related to audio setup. I will check stereo mix, however. Rudolf - Original Message - From: Peter Svensson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 5:05 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: The problem is not sound setup related. It present even if microphone is disconnected. To repeat the question from Matt Riddell: Does he have Stereo Mix selected as a recording source? We have found the most common cause of a strong echo to be that the sound card is set to record the outgoing earphone signal. If you post inline it is much easier to see what your answers were to different questions or if you have missed one. Peter ___ 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 ___ 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] Echo problem -- network related?
On Sun, 14 Aug 2005, Rudolf Ladyzhenskii wrote: You don't get 'echo' on the network, you'd only get true echo connecting to analogue PSTN lines so as Matt pointed out it will sound set-up/card related. Yes, this would be the logical conclusion, although it is hard to beleive given what I hear. It sound like I am talking to myself at a pretty good quality. Actually echo quality is much better than other party. This sounds exactly like you are recording the outgoing audio. The windows drivers for some sound cards does that by default. Go to the mixer, select the recording options and enable all controls so they are not hidden. Check which sources are used for recording. E.g. all the Dell desktops we purchased this year have audio drivers that by default record the outgoing audio. Peter ___ 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 problem -- network related?
Rudolf Ladyzhenskii wrote: Hi, all I am running asterisk and my friends are running FireFly IAX phone. All is fine except one of them. When anyone tries to talk to him, tehre is a real bad echo. It is nothing to do with sound setup. Is he using a headset or speakers and microphone? Does he have Stereo Mix selected as a recording source? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 problem -- network related?
Hi, The problem is not sound setup related. It present even if microphone is disconnected. Rudolf - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, August 14, 2005 12:12 PM Subject: Re: [Asterisk-Users] Echo problem -- network related? Rudolf Ladyzhenskii wrote: Hi, all I am running asterisk and my friends are running FireFly IAX phone. All is fine except one of them. When anyone tries to talk to him, tehre is a real bad echo. It is nothing to do with sound setup. Is he using a headset or speakers and microphone? Does he have Stereo Mix selected as a recording source? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Echo problem
Hi , I am trying to use a telephone Atcom AT323 Both in SIP mode and in IAX mode, I have a lot of echo on a large number of number called (NOT ALL, it depends on the network I reach) I see that using in /etc/asterisk/capi.conf echosquelch=1 ;echocancel=1 echotail=64 Everithing is really good (not perfect but infinitely better) The question is: where can I find the meaning of these parameters ? I searched a lot, but I didn't find anything. In other words: why 64 ? I tried 128 with no effect ; And the value of echocancel ? should it be 1 or yes ? If anybody is able to give me a link to the meaning/values of these parfameters I will be very happy ! I am using 3 FritzCard bri and chancapi thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ 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 problem
Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ 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 problem
Hi Martin, There was an great post last week about echo. It stated that the order of the lines matters. It does. The channels must be listed last for the echo cancel and most other things to work. Rx and TX gain is one of the things also affected. Now I'm using TE110 card in my system. I hope this helps because I'm not sure about Analog lines. Martin Roy wrote: Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ 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] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote: rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 Unless you are making measurements and actually analyzing the results you're only stabbing in the dark playing with these things. by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) Bell's usually pretty good (I'm a Bell customer too) so unless you've got seriously screwey lines (unbalanced, reversed tip/ring, grounding issues) you should not be having this kind of problem. Take a read here. I reference this document continuously: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html Yes, it's work and yes, you may have some trouble doing it/locating the numbers for milliwatt and quiet term but you know what, this is engineering and this is how to do it correctly. Everything else is just pissing around hoping for a solution rather than making educated guesses and anlyzing the results. I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) You're just stabbing in the dark here. I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Interesting. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Follow the instructions on the link provided. Find the milliwatt and quiet term numbers for your local CO. Corner a Bell tech (most of them are really really good guys) and explain that you're trying to interface to a telephone line with your computer and you need the quiet and milliwatt numbers in order to ensure your gains are set correctly. It's hidden info but not secret info. Make sure your tip and ring aren't reversed. Make sure one's not grounded or that there's not something else squirrely with your lines. There is a (simple) FIR filter available on the TDM400P FXO modules. Use the fxotune util to properly adjust it. Echo is able to be eliminated, it's just sometimes a real tricky bugger to track down the cause. Regards, Andrew ___ 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 problem
I use Digium TDM400 cards as well. Asterisk's software echo cancellation sucks. From what I've heard on the IRC channel, you'll never completely eliminate echo with it. And unfortunately, hardware echo cancellation starts out at a full T1. They don't seem to have any solution for someone with 4 pots lines like myself. I haven't been able to completely eliminate echo, but I've come close by using the following: echocancel=64 echocancelwhenbridged=no echotraining=800 rxgain=4.5 txgain=0.0 echocancel=64 was significantly better than echocancel=128 (supposedly the same setting you get when you use echocancel=yes) echotraining at 400 was too short, but 800 seems to almost completely eliminate any initial echo. Occasionally there is still a little echo to start with, but it isn't very bad and it goes away quickly. What sort of echo are you getting? Loud, quiet, fades in and out, starts halfway through the call, starts loud and gets quit? Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Roy Sent: Wednesday, June 08, 2005 10:37 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Echo problem Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ 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] Echo problem
I'm sorry all, lines means config lines of code. Michael D Schelin wrote: Hi Martin, There was an great post last week about echo. It stated that the order of the lines matters. It does. The channels must be listed last for the echo cancel and most other things to work. Rx and TX gain is one of the things also affected. Now I'm using TE110 card in my system. I hope this helps because I'm not sure about Analog lines. Martin Roy wrote: Ok I tried Digium TDM400 cards, I tried X100p cards, I tried Clipcomm CG-410 4 FXOs device. Now I just ordered a few Sipura 3000. With the Digium TDM04B cards (4 FXOs) and X100p cards I tried the following : echocancel=yes echocancelwhenbridged=yes echotraining=yes (I tried 800 with TDM04B cards but it didn't made any difference) rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Thanks Martin ___ 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] Echo problem
I use Digium TDM400 cards as well. Asterisk's software echo cancellation sucks. From what I've heard on the IRC channel, you'll never completely eliminate echo with it. And unfortunately, hardware echo cancellation starts out at a full T1. They don't seem to have any solution for someone with 4 pots lines like myself. I haven't been able to completely eliminate echo, but I've come close by using the following: echocancel=64 echocancelwhenbridged=no echotraining=800 rxgain=4.5 txgain=0.0 echocancel=64 was significantly better than echocancel=128 (supposedly the same setting you get when you use echocancel=yes) I'm not the OP, but thought I'd add what my TDM04b is using: echotraining=800 echocancel=yes echocancelwhenbridged=yes rxgain=5.0 txgain=0.0 Things are working pretty good. (No other changes at all when compiling cvs-head.) ___ 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 problem
I was having problems and your tip helped, my handset showed a polarity reversal... Now we'll see how well it works... Thanks, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Wednesday, June 08, 2005 2:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Echo problem On Wednesday 08 June 2005 13:37, Martin Roy wrote: rxgain= I tried from -8.0 to 10.0 txgain = I tried from -8.0 to 10.0 Unless you are making measurements and actually analyzing the results you're only stabbing in the dark playing with these things. by the way I live in Canada and the provider is Bell Canada for all lines (I have over 10 lines at one place and 3 lines at another places) Bell's usually pretty good (I'm a Bell customer too) so unless you've got seriously screwey lines (unbalanced, reversed tip/ring, grounding issues) you should not be having this kind of problem. Take a read here. I reference this document continuously: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.ht ml Yes, it's work and yes, you may have some trouble doing it/locating the numbers for milliwatt and quiet term but you know what, this is engineering and this is how to do it correctly. Everything else is just pissing around hoping for a solution rather than making educated guesses and anlyzing the results. I tried on a bunch of different computers. I tried on a P4, a dual Xeon, a dual AMD Opteron, a bunch of Macs too (for X100p cards) a PowerMac 8500, 9600, 9650, G3 Desktop, G3 BW and G4. I have echo problem on all of them. I even tried on different OS. Fedora Core 1, 2 and 3 for the PCs and Yellow Dog Linux 3.01 and 4.0 for the Macs. I even tried the Clipcomm CG-410 on OS X (10.2.8, 10.3.x and even 10.4) You're just stabbing in the dark here. I'm using Cisco IP Phone 7960 with SIP firmware 7.3 and a few Granstream GXP-2000. The echo is a lot worst on Cisco phones. Interesting. Now I just ordered 5 Sipura 3000 to see if that will remove the echo. I can't understand why it wouldn't work with the Digium cards... If someone has a clue to help me figure out how to remove this echo well let me know as right now I'm considering that all Digium cards sucks... For Clipcomm well the echo was there and I can't get Caller ID to work so it's useless... Follow the instructions on the link provided. Find the milliwatt and quiet term numbers for your local CO. Corner a Bell tech (most of them are really really good guys) and explain that you're trying to interface to a telephone line with your computer and you need the quiet and milliwatt numbers in order to ensure your gains are set correctly. It's hidden info but not secret info. Make sure your tip and ring aren't reversed. Make sure one's not grounded or that there's not something else squirrely with your lines. There is a (simple) FIR filter available on the TDM400P FXO modules. Use the fxotune util to properly adjust it. Echo is able to be eliminated, it's just sometimes a real tricky bugger to track down the cause. Regards, Andrew ___ 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] Echo problem
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the
results you're only stabbing in the dark playing with these things.
{clip}
Take a read here. I reference this document continuously:
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
Yes, it's work and yes, you may have some trouble doing it/locating the
numbers for milliwatt and quiet term but you know what, this is engineering
and this is how to do it correctly. Everything else is just
pissing around hoping for a solution rather than making educated guesses and
anlyzing the results.
You may also wish to review
http://lists.digium.com/pipermail/asterisk-users/2005-March/096426.html which
attempts to explain the causal relationship between gain and echo - ie. Nework
Loss Planning. A neutral configuration may not be the optimal soloution in all
cases.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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RE: [Asterisk-Users] Echo problem
I haven't gone all the way back to the original poster, but I noticed
mention of a TDM400 in a couple of places.
If you are not in North America, you need to pass an option to the wctdm
driver when it loads to set it in the right mode. Default is FCC mode.
This leaves the card with an impedance mismatch in Australia for
example. When loaded like that, no amount of tweaking gain and echo
cancel would get it right. Getting it to load AUSTRALIA mode fixed our
problem nicely.
I added the bit at the bottom of the Wiki a while ago
http://www.voip-info.org/wiki-TDM400P
On Wed, 2005-06-08 at 17:51 -0700, Kris Boutilier wrote:
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 08, 2005 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Echo problem
On Wednesday 08 June 2005 13:37, Martin Roy wrote:
rxgain= I tried from -8.0 to 10.0
txgain = I tried from -8.0 to 10.0
Unless you are making measurements and actually analyzing the
results you're only stabbing in the dark playing with these things.
{clip}
Take a read here. I reference this document continuously:
http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html
Yes, it's work and yes, you may have some trouble doing it/locating the
numbers for milliwatt and quiet term but you know what, this is engineering
and this is how to do it correctly. Everything else is just
pissing around hoping for a solution rather than making educated guesses
and
anlyzing the results.
You may also wish to review
http://lists.digium.com/pipermail/asterisk-users/2005-March/096426.html which
attempts to explain the causal relationship between gain and echo - ie.
Nework Loss Planning. A neutral configuration may not be the optimal
soloution in all cases.
Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District
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[Asterisk-Users] echo problem
I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside from the voip-info page on echo cancelation, can anyone suggest directives for sip.conf or similar to play with for echo cancelation. I have looked into sjphone as the cause, after speaking with developers there it was suggested to see if my asterisk box is the cause. Any suggestions? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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 problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Wednesday, 25 May 2005 2:39 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] echo problem I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside from the voip-info page on echo cancelation, can anyone suggest directives for sip.conf or similar to play with for echo cancelation. Have you tried Xten's softphone for the ipaq? I am using it on a 5550 and a 6325 and the quality is as good as a regular phone. It just worked! I had echo problems with sjphone on the 5550, and I never even tried it on the 6325 because of that problem. I have looked into sjphone as the cause, after speaking with developers there it was suggested to see if my asterisk box is the cause. Do you have echo problems with any of the other phones on your asterisk server? Any suggestions? -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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 problem
On Wed, 2005-05-25 at 14:50 +1000, Terry H. Gilsenan wrote: Have you tried Xten's softphone for the ipaq? I am using it on a 5550 and a 6325 and the quality is as good as a regular phone. It just worked! I had echo problems with sjphone on the 5550, and I never even tried it on the 6325 because of that problem. I started with that and quickly abandoned it becuase it seemed to take a lot more cpu. Maybe that extra usage is useful :) Do you have echo problems with any of the other phones on your asterisk server? Hard to say they are laptops with no headset. As such the echo problem can be minimized to the point that be ear (yes a bad way to test) it sounds like its all feedback... -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ 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 problem
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Wednesday, 25 May 2005 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] echo problem On Wed, 2005-05-25 at 14:50 +1000, Terry H. Gilsenan wrote: Have you tried Xten's softphone for the ipaq? I am using it on a 5550 and a 6325 and the quality is as good as a regular phone. It just worked! I had echo problems with sjphone on the 5550, and I never even tried it on the 6325 because of that problem. I started with that and quickly abandoned it becuase it seemed to take a lot more cpu. Maybe that extra usage is useful :) I am using the Pro version of the software and it coexists fine with all my other apps, including the GMS phone application on the 6325 The SIP calls seem clearer than the GSM calls. Do you have echo problems with any of the other phones on your asterisk server? Hard to say they are laptops with no headset. As such the echo problem can be minimized to the point that be ear (yes a bad way to test) it sounds like its all feedback... I would have thought that using the ear was the _best_ way to test for echo/feedback. I use iaxphone with a usb handset on my laptop and that works fine too. I did find that there were no adequate software solutions for the laptop using the speakers and built-in microphones, but as soon as the headset or USB handset were added, most of them work fine. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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 problem on SPA-841
I'm running the latest firmware on the SPA-841 and have a problem with echo. The echo occurs on all calls (PRI ISDN on a E110p or SIP) and is not present when I use the SNOM190 phones so I can def. isolate it down to the SPA-841s. The codec used is g711u and the phones are on their own dedicated 100mbit switch with no other traffic. The server is a 3Ghz PIV sitting at 99.9% idle all the time. Specifically, i believe I am getting feedback from the headset speaker to the microphone, although the problem is also apparent on the headset (do not have a selection of headsets to try so cannot draw any conclusions from this) The echo can be cured (or at least mitigated) by turning down the volume on the headset/handset speaker, though at this point the callee struggles to hear the caller. If i turn down (or even leave the gain at 0 from +6) the microphone on the headset/handset then the caller cannot hear the callee (callers complain that the callee sounds far away - 'have you moved to bangalore' is the usual comment (!) -, though the echo is better attenuated. Using the Echo() application I can examine the feedback pretty closely by using DTMF tones. In order to stop the echo, I simply push mute and it vanishes instantly. It is otherwise damped in a time that appears to be a function of the amplitude of the noise made - predictably enough... Short of treating the symptoms, does anyone know of a better way of solving this problem? Is there an echo cancellation algorithm for SIP traffic, for example? Failing that, does anyone want to buy 50 boxed hardly used spa-841s - one careful owner...none thrown against the wall in anger...yet ;-) ___ 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 Problem
Brian M. Arlinghaus wrote: I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. Just wanted to second this. I have about 20 7940's, 2 7960, and a 4 port FXS for fax machines going into a Bellsouth T1 (pri) and we get echo on some calls. I turned on echotraining (not for the faxes of course -- echocancelwhenbridged=no) and it will train out, but I thought that voip - pri could not have echo problems. Anyway, please keep me updated if you figure out a (real) solution to this. Jeb -- Jeb Campbell [EMAIL PROTECTED] ___ 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 Problem
I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. Just wanted to second this. I have about 20 7940's, 2 7960, and a 4 port FXS for fax machines going into a Bellsouth T1 (pri) and we get echo on some calls. I turned on echotraining (not for the faxes of course -- echocancelwhenbridged=no) and it will train out, but I thought that voip - pri could not have echo problems. Anyway, please keep me updated if you figure out a (real) solution to this. echotraining=800 did fix the OP's problem. But, there can still be far-end echo even with PRI's. Those cases involve hybrid issues at some distant end that are difficult at best to address at your end. As has been stated many times before, the echo canceller within * is not as good/reliable as commercial can's and won't handle some of the far-end echo problems. ___ 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 Problem
I've got mostly Cisco 7960s and a few Analog phones on TDM Ports. On the 7960s, the echo is quite bad. On the TDM ports, it is there, but not as bad. I have tried setting echo cancellation to various numbers, but have had no luck. This began after a HEAD version of * was installed. Since then, I installed what I think is the latest stable version (Asterisk CVS-v1-0-12/14/04-16:49:32) and the echo is still there. A support guy at Digium said it was a SIP problem. What can I do to track down and fix the issue? Thanks in advance, Brian From sip.conf... [89XX-1] type=friend host=dynamic secret=89XX-1 context=local callerid=NAMEXX 859-392-89XX disallow=all allow=ulaw qualify=yes From zapata.conf... [channels] echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 switchtype=national signalling=pri_cpe context=incoming group=1 channel = 1-23 signalling=fxo_ks context=longdistance group=2 channel = 25-28 ___ 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 problem - (sorry if this is an nmf question)
I recently installed * on my firewall and that of a relative some miles away. I route sipphone (kphone and x-lite) calls from deep within the backbone (two layers of firewall) on each end to the other. Works fine between @200Mhz pentium doorstop linux boxes (even w/2.4 kernel).The problem of course is the output of the speaker at the other end is picked up by the microphone (confirmed by switching to headset eliminating echo on opposite end). Before I dig into doc of x-lite/kphone I thought I would try asking here for a solution other than simple using headset at each end. (would like to have the speakers ring to let folk know there is inbound calls so they can hear when they are not at the computer) (and yes, the next step is of course to actually buy some hardware so that all the house phones can make voip calls in addition to pots which eliminates this problem, but that is probably not in this quarter's budget.) ___ 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 Problem with IAX and Zaptel
My configuration are: AlcatelOmni PCX ßà1st Asterisk Server with ZapCard ßàIAX trunk over Internetß 2nd Asterisk Server ßà SIP phone I have problem with echo in this configuration. But when I use sip phone or call trough BRI even if I use IAX trunk I have no problem Can someone help me ??? Ive try compile Zaptel with Aggressive Option but It make a Noise and it Hang-up. _ Erwan Desvergnes - ANDIUM - 82/86 rue Château Gaillard 69100 Villeurbanne Tel. 04 3743 44 45 / Fax 04 37 43 44 44 E-mail: [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] Echo problem update - POSSIBLE SOLUTION
So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? snip No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? Nope. the intent of that post was only to suggest that echo resolution varies by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/insert-your-favorite-box-here you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in one US case does not have echo after the echotraining=800 implementation. Don't read anything more into it then just that. (The echotraining=800 was enough of a change for that exact system implementation to function well. The next one may not.) Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency, motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involve more then one item. (Gary Mart is focusing on this and I'm sure he would appreciate any technical/programming help he can get. Now I wish I wouldn't have let those skills go years ago.) Swapping motherboards can impact echo but doing so does not address the root cause, only the symptoms. It would be nice to know XXX board works and YYY board does not, but the professional approach should focus on the underlying issue(s) and correcting/compensating for those, if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general. It is known that a lot of implementations don't have echo, and apparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines AND those boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not a simple task. In the past eight months we've all listened to suggestions that include killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those were probably addressing the symptom and not the root cause. It's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching, adjust gain parameters to reasonable levels, use of proper interface cards for your country's pstn standards, etc. Rich ___ 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] Echo problem update - POSSIBLE SOLUTION
Thanks for that. Like many I believe * is unusable in production until these echo issues are quoshed are resolved. Lets hope someone takes up the bounty offer.Rich Adamson [EMAIL PROTECTED] wrote: So are saying that T2240 will gurantee no echo issues? Did you get any echo issues with a different PC with the same cards and Pstn lines? No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes?Nope. the intent of that post was only to suggest that echo resolutionvaries by system, and has nothing to do with how fancy/speedy of a Compaq/Dell/HP/IBM/ you might be considering or have available, or how much you spent for it. The T2240 with tdm-x100p cards in "one US case" does not have echo afterthe echotraining=800 implementation. Don't read anything more into itthen just that. (The echotraining=800 was enough of a change for thatexact system implementation to function well. The next one may not.)Some strong arguments have been made off-list the existing echo cancellation function is highly dependent upon interrupt latency,motherboard chipset in use, PCI controller, and/or other system-level items that might even include driver inefficiencies of the NIC card. Its way to early to pin the issue any closer, and might even involvemore then one item. (Gary Mart is focusing on this and I'm sure hewould appreciate any technical/programming help he can get. Now Iwish I wouldn't have let those skills go years ago.)Swapping motherboards can impact echo but doing so does not addressthe root cause, only the symptoms. It would be nice to know XXX boardworks a nd YYY board does not, but the professional approach shouldfocus on the underlying issue(s) and correcting/compensating for those,if possible. It could be something as simple as a linux installation default (eg, assuming 33mhz buss, choice of drivers), or as complex as rewriting how the cancellation algorithm functions in general.It "is" known that a lot of implementations don't have echo, andapparently those boxes are using internal system resources that fall within the tolerances of the existing cancellation routines ANDthose boxes have been correctly interfaced to their pstn. Why others don't needs to be identified, and unfortunately, is not asimple task.In the past eight months we've all listened to suggestions thatinclude killing the system's GUI interface, don't share interrupts, reverse tip ring, etc, etc. However, it now _appears_ those wereprobably addressing the symptom and not the root cause.It 's still most appropriate to ensure the pstn interfacing is implemented correctly including source of T1 sync, impedance matching,adjust gain parameters to reasonable levels, use of proper interfacecards for your country's pstn standards, etc.Rich___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself
Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION
could one at least in the case of the fxo/fxs cards just call out one port and be looped back into the other, record the outgoing and incomming call (one recording / port) then compare the phase difference of the 2 recordings? -Galt On Fri, 16 Jul 2004 13:28:46 -0600, Rich Adamson [EMAIL PROTECTED] wrote: On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: snip Anyone have the knowledge/experience to be able to write something that might provide all of us with a clue in terms of buss latency (or whatever we might want to call this)? I'm not a programmer, but it would seem like this test app would have to run in a manner similar to *, interact with digium cards, and return some value that would represent overall latency. Don't think its all that important whether it returns an accurate number of milliseconds or some integer value, as long as the value can be compared from one motherboard to another (and from one site to another). Sort of a run this and tell me what value is returned kind of thing. Can anyone help? Rich ___ 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 -- They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Echo problem update - POSSIBLE SOLUTION
So are saying that T2240 will gurantee no echo issues?Did you get anyecho issueswith a different PC with the same cards and Pstn lines? Taff.Steve Underwood [EMAIL PROTECTED] wrote: Rich Adamson wrote:On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04bor x100p running any Head cvs after June 23rd (totally stock install).Wouldn't necessarily recommend this box for any commercial productionuse, but...What's common and not so common between these _very_ diverse boxes? My guess would be interrupt and/or PCI latency. Echo is produced bydelays in the audio path so if some motherboards are adding delays it'sgoing to make the echo worse. Fiddling with PCI bus settings both in theBIOS and from Linux (using the pci tools) may help in some cases.The unfortunate part about this is that there are SO many variables thatcan influence latency that you can't really tell if a motherboard isgoing to work or not until you try it. Even two MBs with the same CPUsand the same north/south bridges could produce different results.Probably the best we can hope for right now is to start building awhitelist of known good motherboards for people to reference whenbuilding Asterisk systems. I'm kinda thinking you're right in the ball park of where 'at least some'of the remaining echo issues might be coming from. We have an entirelaundry list of what its _not_, but nothing substantial in terms ofwhat _might_ be causing it on selected systems and no good way toquantify it. Frame slips could explain some. All the reports of pages getting chopped while using the SofFax in spandsp, which I have followed up on, have been due to frame slips. It seems a lot of people have their clocking wrong, and those slips willscrew the training of an echo canceller just as well as they screw up modems.Anyone have the knowledge/experience to be able to write "something"that might provide all of us with a clue in terms of buss latency(or whatever we might want to call this)?I'm not a programmer, but it would seem like this test app would have torun in a manner similar to *, interact with digium cards, and return some value that would represent overall latency. Don't think its allthat important whether it returns an accurate number of millisecondsor some integer value, as long as the value can be compared from onemotherboard to another (and from one site to another). Sort of a"run this and tell me what value is returned" kind of thing.& gt; An app that loops back multiple ports and pumps data around in circles for hours would shake out a lot of flaky systems. I used to use one in the early days of the Tormenta 1 card, but I probably don't have it any more.Regards,Steve___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ALL-NEW Yahoo! Messenger - so many all-new ways to express yourself
Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION
John Galt wrote: could one at least in the case of the fxo/fxs cards just call out one port and be looped back into the other, record the outgoing and incomming call (one recording / port) then compare the phase difference of the 2 recordings? -Galt That is probably the simplest way to achieve the required analysis. Any dropped, or inserted, audio sample represents a large phase jump. Send a single tone at a frequency which doesn't correlate well with the 8000 samples/second rate (i.e. things like 1kHz would be a poor choice). Then just run a sliding medium term correlation (simply the sliding dot product of maybe 20ms of audio is adequate) between the transmitted and received audio, and look for significant jumps, after an initial settling period. That should be a solid, noise tolerant, way of looking for these phase jumps. With digital interfaces (BRI, E1, T1) you should get back precisely what you send. It is better in those cases to loop back, send random numbers and check you get back the same random values. The path length is not very long, so its not a big problem to sync to the delay between the outgoing and incoming stream. Once things initially settle, any wrong value, and especially any change in the delay between transmit and receive is a bad thing. A simple tool like this for people to check out new installations has real long term value, beyond the short term goal of fixing echo problems. The number of people having data slip problems affecting their use of my SoftFax has made me consider writing such a tool recently. If anyone would like to save me the trouble, please do. :-) Regards, Steve ___ 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] Echo problem update - POSSIBLE SOLUTION
After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief RD engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/ hardware scsi raid we had echo issues. We moved everything to a Compaq DL380 dual PIII-833MHz (notice a slower machine) w/ hardware scsi raid. Same T1 cards, same compile method, and NO echo on the DL... We did manage to get the echo to almost un-noticable on the Supermicro, but it was still there. There is NO echo on the DL380 We are currently testing the Compaq 6400R for compatability and will let the list know how it goes. When we install asterisk we DO modify the makefile for the zaptel source to look like this... (you'll add 2 lines, the Mark2 and Aggressive_suppressor) KFLAGS+=-DSTANDALONE_ZAPATA KFLAGS+=-DECHO_CAN_MARK2 KFLAGS+=-DAGGRESSIVE_SUPPRESSOR CFLAGS+=-DSTANDALONE_ZAPATA *** info. about our current setup *** running -- Asterisk CVS-HEAD-07/15/04-01:47:37 currently running on asterisk (pid = 1203) /etc/zaptel.conf # # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1) # 5: -7.5db (CSU) # 6: -15db (CSU) # 7: -22.5db (CSU) # # The framing is one of d4 or esf for T1 or cas or ccs for E1 # # Note: d4 could be referred to as sf or superframe # # The coding is one of ami or b8zs for T1 or ami or hdb3 for E1 # # E1's may have the additional keyword crc4 to enable CRC4 checking # # If the keyword yellow follows, yellow alarm is transmitted when no # channels are open. # #span=1,0,0,esf,b8zs #span=2,1,0,esf,b8zs #span=3,0,0,ccs,hdb3,crc4 # # Next come the dynamic span definitions, in the form: # dynamic=driver,address,numchans,timing # # Where driver is the name of the driver (e.g. eth), address is the # driver specific address (like a MAC for eth), numchans is the number # of channels, and timing is a timing priority, like for a normal span. # use 0 to not use this as a timing source, or prioritize them as # primary, secondard, etc. Note that you MUST have a REAL zaptel device # if you are not using external timing. # # dynamic=eth,eth0/00:02:b3:35:43:9c,24,0 # # Next come the definitions for using the channels. The format is: # device=channel list # # Valid devices are: # # em : Channel(s) are signalled using EM signalling (specific # implementation, such as Immediate, Wink, or Feature Group D # are handled by the userspace library). # fxsls : Channel(s) are signalled using FXS Loopstart protocol. # fxsgs : Channel(s) are signalled using FXS Groundstart protocol. # fxsks : Channel(s) are signalled using FXS Koolstart protocol. # fxols : Channel(s) are signalled using FXO Loopstart protocol. # fxogs : Channel(s) are signalled using FXO Groundstart protocol. # fxoks : Channel(s) are signalled using FXO Koolstart protocol. # sf : Channel(s) are signalled using in-band single freq tone. # Syntax as follows: #channel# = sf:rxfreq,rxbw,rxflag,txfreq,txlevel,txflag # rxfreq is rx tone freq in hz, rxbw is rx notch (and decode) # bandwith in hz (typically 10.0), rxflag is either 'normal' or # 'inverted', txfreq is tx tone freq in hz, txlevel is tx tone # level in dbm, txflag is either 'normal' or 'inverted'. Set # rxfreq or txfreq to 0.0 if that tone is not desired. # unused : No signalling is performed, each channel in the list remains idle # clear : Channel(s) are bundled into a single span. No conversion or # signalling is performed, and raw data is available on the master. # indclear: Like clear except all channels are treated individually and # are not bundled. bchan is an alias for this. # rawhdlc : The zaptel driver performs HDLC encoding and decoding on the # bundle, and the resulting data is communicated via the master # device. # fcshdlc : The zapdel driver performs HDLC encoding and decoding on the # bundle and also performs incoming and outgoing FCS insertion # and verification. dchan is an alias for this. # nethdlc
Re: [Asterisk-Users] Echo problem update - POSSIBLE SOLUTION
After speaking with several people, and even participating in a forum of several other people with echo issues, I thought I'd share what we've done (well actually what our chief RD engineer, Brett Bourn has done...) First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/ hardware scsi raid we had echo issues. We moved everything to a Compaq DL380 dual PIII-833MHz (notice a slower machine) w/ hardware scsi raid. Same T1 cards, same compile method, and NO echo on the DL... We did manage to get the echo to almost un-noticable on the Supermicro, but it was still there. There is NO echo on the DL380 We are currently testing the Compaq 6400R for compatability and will let the list know how it goes. When we install asterisk we DO modify the makefile for the zaptel source to look like this... (you'll add 2 lines, the Mark2 and Aggressive_suppressor) KFLAGS+=-DSTANDALONE_ZAPATA KFLAGS+=-DECHO_CAN_MARK2 KFLAGS+=-DAGGRESSIVE_SUPPRESSOR CFLAGS+=-DSTANDALONE_ZAPATA *** info. about our current setup *** running -- Asterisk CVS-HEAD-07/15/04-01:47:37 currently running on asterisk (pid = 1203) No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? ___ 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] Echo problem update - POSSIBLE SOLUTION
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? My guess would be interrupt and/or PCI latency. Echo is produced by delays in the audio path so if some motherboards are adding delays it's going to make the echo worse. Fiddling with PCI bus settings both in the BIOS and from Linux (using the pci tools) may help in some cases. The unfortunate part about this is that there are SO many variables that can influence latency that you can't really tell if a motherboard is going to work or not until you try it. Even two MBs with the same CPUs and the same north/south bridges could produce different results. Probably the best we can hope for right now is to start building a whitelist of known good motherboards for people to reference when building Asterisk systems. -- Joshua M. Thompson [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] Echo problem update - POSSIBLE SOLUTION
On Jul 16, 2004, at 11:07 AM, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? I find it really, really bizarre that analog echos in digital signals behave differently on different systems. On the other hand, this isn't the first report of this happening. Does anyone have a possible mechanism for this? Scott ___ 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] Echo problem update - POSSIBLE SOLUTION
On Friday 16 July 2004 12:43, W. Kevin Hunt wrote: First let me say that normal cheapy PC hardware couldn't be made to function with out echo. We tried on both the single port Digium T1 card and the 4 port Digium T1 card. Even on a SuperMicro Dual PIII-933 w/ hardware scsi raid we had echo issues. We moved everything to a Compaq DL380 dual PIII-833MHz (notice a slower machine) w/ hardware scsi raid. Same T1 cards, same compile method, and NO echo on the DL... We did manage to get the echo to almost un-noticable on the Supermicro, but it was still there. There is NO echo on the DL380 We are currently testing the Compaq 6400R for compatability and will let the list know how it goes. We have a T100P and a TE405P on Supermicro dual Xeon 2.8, SCSI backplane, *server* systems... we get some echo on some calls out to the Bell Canada PRI, and some on calls through nufone. It's definitely destination related for us. The T100P connects to an Adit600 FXS channel bank which goes to our Meridian MICS -- Eventually I'll be replacing the channel bank and POTS trunk lines on the MICS with a proper PRI card. Regards, Andrew ___ 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] Echo problem update - POSSIBLE SOLUTION
On Fri, 2004-07-16 at 12:07 -0600, Rich Adamson wrote: No echo on eMachine T2240 2.2ghz Celery, 360m RAM, with either tdm04b or x100p running any Head cvs after June 23rd (totally stock install). Wouldn't necessarily recommend this box for any commercial production use, but... What's common and not so common between these _very_ diverse boxes? My guess would be interrupt and/or PCI latency. Echo is produced by delays in the audio path so if some motherboards are adding delays it's going to make the echo worse. Fiddling with PCI bus settings both in the BIOS and from Linux (using the pci tools) may help in some cases. The unfortunate part about this is that there are SO many variables that can influence latency that you can't really tell if a motherboard is going to work or not until you try it. Even two MBs with the same CPUs and the same north/south bridges could produce different results. Probably the best we can hope for right now is to start building a whitelist of known good motherboards for people to reference when building Asterisk systems. I'm kinda thinking you're right in the ball park of where 'at least some' of the remaining echo issues might be coming from. We have an entire laundry list of what its _not_, but nothing substantial in terms of what _might_ be causing it on selected systems and no good way to quantify it. Anyone have the knowledge/experience to be able to write something that might provide all of us with a clue in terms of buss latency (or whatever we might want to call this)? I'm not a programmer, but it would seem like this test app would have to run in a manner similar to *, interact with digium cards, and return some value that would represent overall latency. Don't think its all that important whether it returns an accurate number of milliseconds or some integer value, as long as the value can be compared from one motherboard to another (and from one site to another). Sort of a run this and tell me what value is returned kind of thing. Can anyone help? Rich ___ 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
