Re: [asterisk-users] Echo Cancellation on VoIP networks
Does anyone have any experience with PBXMate and the quality of the software? Does it cancel echo properly? Can someone also give me an price indication of the software? I can’t find it on their website. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valer Nur Sent: Wednesday, August 27, 2014 6:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo Cancellation on VoIP networks If the clients are not doing the echo cancellation properly, you can always use a centralized echo cancellation software for VoIP networks. On Wednesday, August 27, 2014 5:25 PM, Dennis Guse dennis.g...@alumni.tu-berlin.demailto:dennis.g...@alumni.tu-berlin.de wrote: On VoIP echo cancellation is basically: hope that the client is doing AND is doing it well. In the best case each client uses a knowledge about his hardware (microphone, speaker, distance etc.). --- Dennis Guse On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.commailto:emilianovazq...@gmail.com wrote: El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.commailto:emilianovazq...@gmail.com Web: http://www.pccentro.com.arhttp://www.pccentro.com.ar/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancellation on VoIP networks
On VoIP echo cancellation is basically: hope that the client is doing AND is doing it well. In the best case each client uses a knowledge about his hardware (microphone, speaker, distance etc.). --- Dennis Guse On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.com wrote: El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancellation on VoIP networks
If the clients are not doing the echo cancellation properly, you can always use a centralized echo cancellation software for VoIP networks. On Wednesday, August 27, 2014 5:25 PM, Dennis Guse dennis.g...@alumni.tu-berlin.de wrote: On VoIP echo cancellation is basically: hope that the client is doing AND is doing it well.In the best case each client uses a knowledge about his hardware (microphone, speaker, distance etc.). --- Dennis Guse On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.com wrote: El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancellation on VoIP networks
Hello, I'm new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? I did some research on the internet about EC on VoIP networks, but I can't really put a grasp on it. We currently have some Echo Cancellation chips on our Digium cards, but are planning to move to a full VoIP network based on Asterisk. So no more ISDN in the voice path. Does Asterisk have any pre-installed Echo Cancellation software? Which hardware cards are recommended for doing echo cancellation on VoIP networks? I hope someone could point me into the right direction. Regards, Grant -- _ -- 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 Cancellation on VoIP networks
El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -- _ -- 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 Cancellation when calling from softphone to mobile.
Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 Cancellation when calling from softphone to mobile.
Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.
Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani mi...@enterux.in wrote: Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 Cancellation when calling from softphone to mobile.
There are two common types of echo. Accoustic Echo: This is caused by microphone picking up audio from the speaker. This echo cannot generally be removed by echo cancelers. The solution to accoustic echo is to prevent the microphone from picking up audio from the speaker (or handset or earpiece). Line Echo: This is caused by your outgoing audio “reflecting” off the far end of an analog line. This happens in all calls with a 2-wire analog portion, however in analog and digital (aka PRI) the delay in echo is so small we can’t perceive it.VoIP has far larger latencies so we can hear the echo. This type of echo MUST be canceled out before the audio is converted to VoIP. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anurag Rana Sent: Wednesday, June 25, 2014 2:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile. Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani mi...@enterux.inmailto:mi...@enterux.in wrote: Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.commailto:anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -- _ -- 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 Cancellation
Shaun Ruffell sruffell at digium.com writes: On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal Also, just FYI, those cards do support adding a hardware echocancelation module. But I would recommend trying the software solutions first. I am new to asterisk and pbxiaf. I have setup a VM and intend to use it for a number of android phones (using sipdroid) all on local wifi for a conference call. There is a lot of echo. I also added a GoogleVoice account, even on a standard phone call the remote party gets a late but large echo. As I understand, DAHDI works with the cards, seeing as my complete system is software based, how would I get rid of the echo. Thanks in advance. Ghanshyam -- _ -- 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 Cancellation
On Tue, Aug 20, 2013 at 3:01 PM, Ghanshyam btcs.em...@gmail.com wrote: Shaun Ruffell sruffell at digium.com writes: On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal Also, just FYI, those cards do support adding a hardware echocancelation module. But I would recommend trying the software solutions first. I am new to asterisk and pbxiaf. I have setup a VM and intend to use it for a number of android phones (using sipdroid) all on local wifi for a conference call. There is a lot of echo. I also added a GoogleVoice account, even on a standard phone call the remote party gets a late but large echo. As I understand, DAHDI works with the cards, seeing as my complete system is software based, how would I get rid of the echo. Thanks in advance. Ghanshyam -- _ -- 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 Unlike our friend Ghanshyam, i'm sorry for the hijack however does OSLEC only work with telephony cards or, will it also work on a purely Asterisk SIP environment? Thanks in Advance, Nick. -- _ -- 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 Cancellation
Echo must be canceled where the low latency audio (PSTN) meets high latency audio (VoIP).You can't echo cancel VoIP. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Tuesday, August 20, 2013 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Echo Cancellation Unlike our friend Ghanshyam, i'm sorry for the hijack however does OSLEC only work with telephony cards or, will it also work on a purely Asterisk SIP environment? -- _ -- 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 Cancellation
Thanks Eric, I breezed through the documentation and got the impression that this was the case. Good luck on getting rid of that echo Bilal! N. -- _ -- 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 Cancellation
25.07.2013 13:51, bilal ghayyad пишет: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? oslec, imho. -- _ -- 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 Cancellation
On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo canceller software module. Google is your friend. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo Cancellation
Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? 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 Cancellation
On 7/25/2013 5:57 AM, Patrick Lists wrote: On 07/25/2013 11:51 AM, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Use the free OSLEC echo canceller software module or Digium's commercial HPEC echo canceller software module. Google is your friend. Regards, Patrick +1 for OSLEC JohnM -- _ -- 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 Cancellation
On Thu, Jul 25, 2013 at 02:51:02AM -0700, bilal ghayyad wrote: Hello; If our Digium Telephony Card does not support echo cancellation like (1TDM410PLF or 1AEX410PLF), what is the best and simple way to overcome the echo? Regards Bilal Also, just FYI, those cards do support adding a hardware echocancelation module. But I would recommend trying the software solutions first. -- 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 from channel bank
Valer, Thank you for the advice - I have support tickets open with Adtran and Digium and we are tracking down the issue. Hopefully it doesn't come down to adding more hardware, but I'll keep that in mind. -Justin Killen From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Valer Nur Sent: Monday, January 07, 2013 11:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] echo from channel bank It sounds to me like you should first discuss it with adtran. The standard echo cancellation for Asterisk have a hard time cancelling echo generated at the far end, especially if the echo tail/delay is not minimal. If adtran can not solve the problem at their end, you can use a server-side echo cancellation that can handle long echo tail. One option is PBXMate. From: Justin Killen jkil...@allamericanasphalt.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 8, 2013 12:17 AM Subject: [asterisk-users] echo from channel bank I have several adtran 624 with 24 FXS ports hooked up to analog phones. The adtran is connected to asterisk via a channelized T1 into a digium TE820. I have hardware echo canceling enabled on all channels/spans, but there is still echo on the lines for both calls out of the trunk, as well as station-to-station calls. I've checked the output of 'dahdi show channel x' and see the echo turned on: dozer2*CLI dahdi show channel 149 Channel: 149 Description: File Descriptor: 158 Span: 7 Extension: 98300326 Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID subaddress: Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: DAHDI/149-1 Real: DAHDI/149-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps currently ON Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook I have echocanceller set to HWEC in dahdi/system.conf for all spans, and I have echocancel=yes in chan_dahdi.cfg I've tried reading some echo cancelation articles, but they seem to be focused on trunks and not on stations. Also, I'm not 100% sure if this is an issue that I should focus on with asterisk, or if it's something I should first take up with adtran? I'm using dahdi 2.6.1, asterisk 10.10.0. Thanks in advance, -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] echo from channel bank
I have several adtran 624 with 24 FXS ports hooked up to analog phones. The adtran is connected to asterisk via a channelized T1 into a digium TE820. I have hardware echo canceling enabled on all channels/spans, but there is still echo on the lines for both calls out of the trunk, as well as station-to-station calls. I've checked the output of 'dahdi show channel x' and see the echo turned on: dozer2*CLI dahdi show channel 149 Channel: 149 Description: File Descriptor: 158 Span: 7 Extension: 98300326 Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID subaddress: Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: DAHDI/149-1 Real: DAHDI/149-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Gains (RX/TX): 0.00/0.00 Dynamic Range Compression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps currently ON Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook I have echocanceller set to HWEC in dahdi/system.conf for all spans, and I have echocancel=yes in chan_dahdi.cfg I've tried reading some echo cancelation articles, but they seem to be focused on trunks and not on stations. Also, I'm not 100% sure if this is an issue that I should focus on with asterisk, or if it's something I should first take up with adtran? I'm using dahdi 2.6.1, asterisk 10.10.0. Thanks in advance, -Justin -- _ -- 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 from channel bank
It sounds to me like you should first discuss it with adtran. The standard echo cancellation for Asterisk have a hard time cancellingecho generated at the far end, especially if the echo tail/delay is notminimal. If adtran can not solve the problem at their end, you can use a server-side echo cancellation that can handle long echo tail. One option is PBXMate. From: Justin Killen jkil...@allamericanasphalt.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, January 8, 2013 12:17 AM Subject: [asterisk-users] echo from channel bank I have several adtran 624 with 24 FXS ports hooked up to analog phones. The adtran is connected to asterisk via a channelized T1 into a digium TE820. I have hardware echo canceling enabled on all channels/spans, but there is still echo on the lines for both calls out of the trunk, as well as station-to-station calls. I’ve checked the output of ‘dahdi show channel x’ and see the echo turned on: dozer2*CLI dahdi show channel 149 Channel: 149 Description: File Descriptor: 158 Span: 7 Extension: 98300326 Dialing: no Context: from-zaptel Caller ID: Calling TON: 0 Caller ID subaddress: Caller ID name: Mailbox: none Destroy: 0 InAlarm: 0 Signalling Type: FXS Kewlstart Radio: 0 Owner: DAHDI/149-1 Real: DAHDI/149-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Busy Detection: no TDD: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Gains (RX/TX): 0.00/0.00 DynamicRangeCompression (RX/TX): 0.00/0.00 DND: no Echo Cancellation: 128 taps currently ON Wait for dialtone: 0ms Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook I have echocanceller set to HWEC in dahdi/system.conf for all spans, and I have echocancel=yes in chan_dahdi.cfg I’ve tried reading some echo cancelation articles, but they seem to be focused on trunks and not on stations. Also, I’m not 100% sure if this is an issue that I should focus on with asterisk, or if it’s something I should first take up with adtran? I’m using dahdi 2.6.1, asterisk 10.10.0. Thanks in advance, -Justin -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] echo canceler query
are you able to received Fax? is sending fax the only problem? what do you have on chan_dahdi.conf? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina Berretta Sent: Tuesday, July 24, 2012 1:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] echo canceler query Hello I have a question regarding Asterisk echo cancellation. I´m using a fax connected to an FXS interface and an E1 ISDN line to PSTN. The issue I´m having is with sending fax through the E1 and I´m suspecting is due to echo troubles as I have some echo issues in my E1 line. How can I know if the echo canceller is being cancelled when fax tone is detected? Thanks!!! -- _ -- 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 audio delay in SIP VOIP
Hello sir, There is an echo problem in sip voip call. I think it is because of delay in audio. Let me try to explain you my system setup. I have test asterisk on two different system. System : 1 OS :Ubuntu(10.04)Lucid System Type :x64-based pc Processor :Intel(R) Core i5 CPU M520 @ 2.40GHz RAM :4.00 GB System : 2 OS :CentOS (5.7) System Type :x86-based pc Processor :Intel(R) Pentium D CPU 3.00GHz RAM :1.00 GB I use beetel magiq android tablet as video sip dialer.( http://beetel.quasar.in/magiqII ), which is rebranded version of huawei ideos S7 tablet (http://www.huaweidevice.com/resource/mini/201008174756/ideos/products_s7.ht ml) I use D-link DIR-615 WIRELESS N 300 ROUTER (http://www.dlink.co.in/products/?pid=349) as wireless access point. I have try two android video softphone on my tablet. 1 Voipswitch : http://voipswitch.com/en/products/softphones/mobile-softphones/softphone-for -android/ 2 Linphone (1.2.2) : http://www.linphone.org/eng/download/packages/android.html * i also try both codec( G729 , G711) * android tablets and asterisk server are connected in Local Area Network(LAN). -- _ -- 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 05:38:37PM -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 when calling to the pstn
I'm assuming you haven't googled for solution, please go through http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and extra links in that article. If that article were not helpful, please provide more information of you setup, such as what analog card are you using, are you using software or hardware echo canceller, how does your chan_dahdi.conf look like, etc. On Tue, Feb 8, 2011 at 3:11 PM, Vitor Carlos Flausino vitor.carlos.flaus...@gmail.com wrote: Hello all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. Can you help me with this issue? Best regards, Vitor Flausino -- _ -- 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 -- Ye Liu (AKA @jaux) http://jaux.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] echo when calling to the pstn
Hello all. I have a asterisk implementation (asterisknow 1.7.1), with a card with 2 FXO interfaces. When I call (or receive a call) from the pstn, I ear echo. This happens if I use a softphone or IP phone, and does not happens if the call is internal. Can you help me with this issue? Best regards, Vitor Flausino -- _ -- 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 Cancellation Problem - Invalid Argument?!?
Well, I downgraded this box to Asterisk 1.4.38 and all is well again. Echo cancellation works properly, no problems, no errors. I have to assume this is a bug in Asterisk 1.8.x or Wanpipe 3.5.18. --Tim - Original Message - Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) -- _ -- 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 Cancellation Problem - Invalid Argument?!?
Trying again... I think this got lost in the mailing list interruptions during the last day or two... - Forwarded Message - From: Tim Nelson tnel...@fudnet.net To: asterisk-users@lists.digium.com Sent: Wednesday, December 15, 2010 5:07:20 PM Subject: [asterisk-users] Echo Cancellation Problem - Invalid Argument?!? Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) --Tim -- _ -- 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 Cancellation Problem - Invalid Argument?!?
Greetings folks- I'm experiencing issues with a freshly installed box. When a call comes in via PRI (Sangoma AFT-A104), I see this in my logs: [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 12 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 8 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 10 (Invalid argument) [Dec 15 14:26:10] WARNING[23546] chan_dahdi.c: Unable to enable echo cancellation on channel 9 (Invalid argument) Relevant components: Asterisk: Asterisk SVN-trunk-r290509 built by root @ prigw01 on a i686 running Linux on 2010-11-30 22:12:05 UTC DAHDI: dahdi-linux-complete-2.4.0+2.4.0 LibPRI: libpri-1.4.11.5 Wanpipe: wanpipe-3.5.18 Kernel: Linux prigw01 2.6.32-24-generic #39-Ubuntu SMP Wed Jul 28 06:07:29 UTC 2010 i686 GNU/Linux The card does not have a hardware echo canceler. It should use MG2 as specified in DAHDI's system.conf: #autogenerated by /usr/sbin/wancfg_dahdi do not hand edit #autogenrated on 2010-12-08 #Dahdi Channels Configurations #For detailed Dahdi options, view /etc/dahdi/system.conf.bak loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 span=1,1,0,esf,b8zs bchan=1-23 #dchan=24 echocanceller=mg2,1-23 hardhdlc=24 And, from chan_dahdi.conf: ;Sangoma A104 port 1 [slot:2 bus:2 span:1] wanpipe1 switchtype=national context=ldrouted group=1 echocancel=yes signalling=pri_net channel =1-23 Any thoughts, pointers, suggestions? The echo is horrible, please help me make it stop. :-) --Tim -- _ -- 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 calls
Hi, Complete Asterisk noob here, I just have a few general questions I need to ask before I start installing it: If I want to setup Asterisk simply to respond to echo calls (SIP calls from a softphone only), do I need to install AsteriskNow as well, or can I simply configure dialplan to do that? Thanks, Ali -- _ -- 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 calls
On Fri, 26 Nov 2010, Ali Khalfan wrote: If I want to setup Asterisk simply to respond to echo calls (SIP calls from a softphone only), do I need to install AsteriskNow as well, or can I simply configure dialplan to do that? 'echo calls' doesn't mean anything to me. What does it mean to you? AsteriskNow = Linux + Asterisk + [FreePBX | AsteriskGUI]. Whether you choose a GUI interface or not, Asterisk still executes a dialplan which you can see by entering 'dialplan show' at the CLI prompt. -- 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 calls
I think you refer to the echo test that comes with the default extensions.conf file. Within the regular installation of asterisk on the extensions.conf file you will have an echo test extension. I don't think you need asterisk now for that. On Fri, Nov 26, 2010 at 1:24 PM, Steve Edwards asterisk@sedwards.comwrote: On Fri, 26 Nov 2010, Ali Khalfan wrote: If I want to setup Asterisk simply to respond to echo calls (SIP calls from a softphone only), do I need to install AsteriskNow as well, or can I simply configure dialplan to do that? 'echo calls' doesn't mean anything to me. What does it mean to you? AsteriskNow = Linux + Asterisk + [FreePBX | AsteriskGUI]. Whether you choose a GUI interface or not, Asterisk still executes a dialplan which you can see by entering 'dialplan show' at the CLI prompt. -- 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 -- Robert -- _ -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
The suggestions did fix the problem. Thank you Shaun and Paul for the help. Regards, Jared On Fri, Oct 15, 2010 at 4:48 PM, Jared Geiger ja...@compuwizz.net wrote: I haven't heard if this fixed it yet. However I was seeing the echo cancelers loaded before so I never realized I'd have to do this. Its a FreePBX install also so I checked all the include files and didn't see a reference to these values anywhere. Thanks everyone for the input, I should know soon if it is the fix. ~Jared On Fri, Oct 15, 2010 at 3:56 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] echo on TE122
I have setup a asterisk with freepbx, a TE122 and i have an ISDN. My problem now is that callers are experiencing echo. checked on dmesg i saw this: # dmesg -c dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2 i searched google but found no soolution and i have no idea what that error means. below are other details that might be of value. # dahdi_hardware pci::04:00.0 wcte12xp+d161:8001 Wildcard TE122 # dahdi_scan [1] active=yes alarms=OK description=Wildcard TE122 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE122 location=PCI Bus 04 Slot 01 basechan=1 totchans=31 irq=20 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI,HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS chan_dahdi.conf contains: echocancel=yes echocancelwhenbridged=yes echotraining=800 -- _ -- 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 on TE122
Thank you Shaun, will try that. will that help on the echo issues users are encountering during calls? On 10/20/10 10:28 PM, Shaun Ruffell wrote: On 10/20/2010 03:20 AM, Ron wrote: I have setup a asterisk with freepbx, a TE122 and i have an ISDN. My problem now is that callers are experiencing echo. checked on dmesg i saw this: # dmesg -c dahdi: Disabled echo canceller NLP because of CED rx detected on channel 2 That's saying that the echo canceller activated for channel 2 is disabled because it believes that the far side is a fax machine or modem. i searched google but found no soolution and i have no idea what that error means. below are other details that might be of value. If you're not actively trying to run faxes through your system, you could try enabling CONFIG_DAHDI_NO_ECHOCAN_DISABLE in include/dahdi/dahdi_config.h. Cheers, Shaun -- _ -- 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 on TE122
On 10/20/10 11:04 AM, Ron wrote: Thank you Shaun, will try that. will that help on the echo issues users are encountering during calls? If all the echo problems are due to erroneous tones, then I believe it should. If your users are still reporting echo you might want to contact Digium technical support for help with troubleshooting. Cheers, Shaun -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
On 10/15/2010 08:55 AM, Jared Geiger wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I've tried the default settings for echo in the system.conf file as well as I've compiled OSLEC to try and see if thats any better. I'm not sure what to try next. Does anyone have any suggestions? What are the outputs of the following commands when your system is up and running? #] cat /etc/dahdi/system.conf #] grep -E ^echo /etc/asterisk/chan_dahdi.conf #] dahdi_scan #] lsmod | grep dahdi -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
[r...@voice ~]# cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Fri Sep 24 21:44:03 2010 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Span 1: WCT1/0 Wildcard TE120P Card 0 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=OSLEC,1-23 # Global data loadzone= us defaultzone = us This might be my problem?*** [r...@voice ~]# grep -E ^echo /etc/asterisk/chan_dahdi.conf * *So I added this under [channels]: echocancel=yes echocancelwhenbridged=no echotraining=800* [r...@voice ~]# dahdi_scan [1] active=yes alarms=OK description=Wildcard TE120P Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE120P location=PCI Bus 03 Slot 12 basechan=1 totchans=24 irq=217 type=digital-T1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding=B8ZS framing=ESF [r...@voice ~]# lsmod | grep dahdi dahdi_echocan_oslec 6912 27 echo9600 1 dahdi_echocan_oslec dahdi_transcode12164 1 wctc4xxp dahdi_voicebus 45760 2 wctdm24xxp,wcte12xp dahdi 196552 78 dahdi_echocan_oslec,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp crc_ccitt 6337 2 wctdm24xxp,dahdi On Fri, Oct 15, 2010 at 11:03 AM, Shaun Ruffell sruff...@digium.com wrote: On 10/15/2010 08:55 AM, Jared Geiger wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I've tried the default settings for echo in the system.conf file as well as I've compiled OSLEC to try and see if thats any better. I'm not sure what to try next. Does anyone have any suggestions? What are the outputs of the following commands when your system is up and running? #] cat /etc/dahdi/system.conf #] grep -E ^echo /etc/asterisk/chan_dahdi.conf #] dahdi_scan #] lsmod | grep dahdi -- 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 -- _ -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
On 10/15/2010 10:33 AM, Jared Geiger wrote: This might be my problem?*** [r...@voice ~]# grep -E ^echo /etc/asterisk/chan_dahdi.conf * *So I added this under [channels]: echocancel=yes echocancelwhenbridged=no echotraining=800* Most likely (unless you were including another file that included that definition). Did this resolve your problem? -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- 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 on PRI after upgrading to Asterisk 1.6.2 from 1.2
I haven't heard if this fixed it yet. However I was seeing the echo cancelers loaded before so I never realized I'd have to do this. Its a FreePBX install also so I checked all the include files and didn't see a reference to these values anywhere. Thanks everyone for the input, I should know soon if it is the fix. ~Jared On Fri, Oct 15, 2010 at 3:56 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Fri, Oct 15, 2010 at 9:55 AM, Jared Geiger compuw...@gmail.com wrote: I've recently upgraded an Asterisk system from 1.2 to 1.6.2 (did a full reformat and recompile) and I started getting echo over the PRI. I did an update on a server last year, had the same problem. I needed to explicitly set echocancel=yes in my configs, before 1.6 it was enabled by default. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on Sangoma A400 and background noise
On Wed, Sep 15, 2010 at 6:10 PM, Al lists asteris...@gmail.com wrote: I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: Hello, I am sorry you've had this bad experience and I apologize on behalf of Sangoma if we gave you the impression of not caring about your technical issue. To rephrase what tech support meant. Our HWEC is known to provide better results, but in no way that means that we will not look at your issue with the card that does not have HWEC. A senior tech support engineer will be contacting you soon today to follow up on your issue appropriately. Regards, Moises Silva Senior Software Engineer Sangoma Technologies Inc. | NEW 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. m...@sangoma.com -- _ -- 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 on Sangoma A400 and background noise
I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: As you mentioned you have tried Oslec algorithms for echo cancellation.Which is a good way to solve echo cancellation issues. If that is not woking for you you may want to upgrade to hardware echo cancellation..with cards which have echo cancellers. Hope this helps. -Sri -- _ -- 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 on Sangoma A400 and background noise
- Al lists asteris...@gmail.com wrote: I'm a long time user of Digium carts and stupid me i wanted to give Sangoma a try. We got Sangoma A400 with 6 FXO ports. Asterisk version: 1.4.35 Zaptel version: 1.4.11 Wanpipe version: 3.5.11 we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. was wondering if you have any experiense with that because Sangoma tech support is not helpfull, just look at their response: As you mentioned you have tried Oslec algorithms for echo cancellation.Which is a good way to solve echo cancellation issues. If that is not woking for you you may want to upgrade to hardware echo cancellation..with cards which have echo cancellers. I've had the exact opposite experience you've had. Digium tech support was less than stellar whereas Sangoma has always been top-notch. Also, their cards with HWEC 'just work' with absolutely no tuning required. --Tim -- _ -- 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 on Sangoma A400 and background noise
On Sep 15, 2010, at 6:10 PM, Al lists wrote: we tried to use fxtune but looks like it wont work with Sangoma card, ( please correct me if i'm wrong) Echo is really bad and also we have background noise on all lines. We tried both mg2 and oslec echo canceler. I've only used Sangoma with hardware echo cancelation, but I did find I had to manually tune my cards. In my case using a test tone number was of no help, rather I just listened in on calls in progress and used ztmonitor to see where the levels were and adjusted things until I had a good level and the sound quality was acceptable. Until I did the manual tuning I had serious issues of static and background noise on several lines, but not all. I don't blame this on the Sangoma card, but rather on the line quality as I run Sangomas in other locations and have never had to do any tuning there, they have just worked. -chris www.mythtech.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 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 cancellation on DAHDI
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Hi, Carlos I checked dmesg on my server and i found following message what is meaning for this ? i cant understand VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) regards Dhaval That means you have a VPM450 echo cancelling module attached to your digital board. All you need to do in order to activate echo cancelling is setting echocancel=yes on your chan_dahdi.conf. After that you can check if the echo canceller is really enabled by triggering a dahdi show channel X on the Asterisk CLI. X should be a channel that's currently on a call. Here's an example: stara*CLI dahdi show channel 64 Channel: 64 File Descriptor: 77 Span: 3LI Extension: Dialing: no Context: pabx Caller ID: Calling TON: 0 Caller ID name: Destroy: 0 InAlarm: 0 Signalling Type: ISDN PRI Radio: 0I Owner: DAHDI/64-1 Real: DAHDI/64-1 Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: yes Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently ON PRI Flags: Call PRI Logical Span: Implicit Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Hookstate (FXS only): Onhook Of course, the interesting line to you is the Echo Cancellation one. -- _ -- 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 cancellation on DAHDI
On Wed, 2010-03-03 at 10:14 +0530, DHAVAL INDRODIYA wrote: Hi, Carlos I checked dmesg on my server and i found following message what is meaning for this ? i cant understand VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) Well, that means that your card does have the echo cancellation module installed and it is active. Please post your DAHDI configuration to make sure your channels are properly configured. You should not have echo on any channel but remember that the E1 is not the only source of echo for calls. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 cancellation on DAHDI
Dear All, How can we know the On board supports echo cancellation I have *Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02)*board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval -- _ -- 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 cancellation on DAHDI
- DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval Do you have an echo cancelling module attached to that board? If so, all you need is to set echocancel=yes and echocancelwhenbridged=no on your chan_dahdi.conf. If you don't... well you should! Anyway, you can turn on the echocancelling via software with echocancel=256. I strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in order to use it, but it's totally worth it. -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote: - DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval Do you have an echo cancelling module attached to that board? If so, all you need is to set echocancel=yes and echocancelwhenbridged=no on your chan_dahdi.conf. If you don't... well you should! Anyway, you can turn on the echocancelling via software with echocancel=256. I strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in order to use it, but it's totally worth it. On the subject of DAHDI -v- OSLEC. I never had any luck getting it to work with DAHDI 2.2.1 despite following: http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi All I ever go was a bad case of the blues :-( make[3]: *** No rule to make target `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/linux/drivers/dahdi/echo.o'. I guess I missed something somewhere??? -- _ -- 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 cancellation on DAHDI
- Brian brel.astersik100...@copperproductions.co.uk escreveu: On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote: - DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval Do you have an echo cancelling module attached to that board? If so, all you need is to set echocancel=yes and echocancelwhenbridged=no on your chan_dahdi.conf. If you don't... well you should! Anyway, you can turn on the echocancelling via software with echocancel=256. I strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in order to use it, but it's totally worth it. On the subject of DAHDI -v- OSLEC. I never had any luck getting it to work with DAHDI 2.2.1 despite following: http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi All I ever go was a bad case of the blues :-( make[3]: *** No rule to make target `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/linux/drivers/dahdi/echo.o'. I guess I missed something somewhere??? Get the most recent version of Linux 2.6 kernel. Inside you'll find a directory named staging/echo. Copy that entire directory to the drivers/linux directory of the DAHDI sources. In the end you gotta have a directory named linux/drivers/staging/echo inside your DAHDI sources. -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote: - Brian brel.astersik100...@copperproductions.co.uk escreveu: On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote: - DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval Do you have an echo cancelling module attached to that board? If so, all you need is to set echocancel=yes and echocancelwhenbridged=no on your chan_dahdi.conf. If you don't... well you should! Anyway, you can turn on the echocancelling via software with echocancel=256. I strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in order to use it, but it's totally worth it. On the subject of DAHDI -v- OSLEC. I never had any luck getting it to work with DAHDI 2.2.1 despite following: http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi All I ever go was a bad case of the blues :-( make[3]: *** No rule to make target `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/linux/drivers/dahdi/echo.o'. I guess I missed something somewhere??? Get the most recent version of Linux 2.6 kernel. Inside you'll find a directory named staging/echo. Copy that entire directory to the drivers/linux directory of the DAHDI sources. In the end you gotta have a directory named linux/drivers/staging/echo inside your DAHDI sources. I already have those :-( ls -alh /usr/src/dahdi/dahdi/linux/drivers/staging/echo drwxr-xr-x 2 root root 4.0K 2010-03-02 14:04 . drwxr-xr-x 3 root root 4.0K 2010-03-02 14:04 .. -rw-r--r-- 1 root root 5.7K 2010-03-02 14:04 bit_operations.h -rw-r--r-- 1 root root 20K 2010-03-02 14:04 echo.c -rw-r--r-- 1 root root 7.2K 2010-03-02 14:04 echo.h -rw-r--r-- 1 root root 7.4K 2010-03-02 14:04 fir.h -rw-r--r-- 1 root root 251 2010-03-02 14:04 Kconfig -rw-r--r-- 1 root root 29 2010-03-02 14:04 Makefile -rw-r--r-- 1 root root 14K 2010-03-02 14:04 mmx.h -rw-r--r-- 1 root root 2.8K 2010-03-02 14:04 oslec.h -rw-r--r-- 1 root root 367 2010-03-02 14:04 TODO To be sure I copied them again... cp -rf /usr/src/dahdi/linux-2.6.28/drivers/staging/echo/* /usr/src/dahdi/dahdi/linux/drivers/staging/echo (the /dahdi/dahdi is not a typo...) But still no dice :-( /usr/src/dahdi/dahdi# make make -C linux all make[1]: Entering directory `/usr/src/dahdi/dahdi/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware' make -C /lib/modules/2.6.27-7-server/build SUBDIRS=/usr/src/dahdi/dahdi/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi/dahdi/linux/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[2]: Entering directory `/usr/src/linux-headers-2.6.27-7-server' make[3]: *** No rule to make target `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'. Stop. make[2]: *** [_module_/usr/src/dahdi/dahdi/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/linux-headers-2.6.27-7-server' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi/dahdi/linux' make: *** [all] Error 2 It would be nice to resolve this - but it's probably beyond my understanding and ability. -- _ -- 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 cancellation on DAHDI
On Tue, 2 Mar 2010, Brian wrote: It would be nice to resolve this - but it's probably beyond my understanding and ability. Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi directory? Gordon -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 15:59 +, Brian wrote: On Tue, 2010-03-02 at 12:45 -0300, Vinícius Fontes wrote: - Brian brel.astersik100...@copperproductions.co.uk escreveu: On Tue, 2010-03-02 at 09:22 -0300, Vinícius Fontes wrote: - DHAVAL INDRODIYA dhaval.it01...@gmail.com escreveu: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . regards Dhaval Do you have an echo cancelling module attached to that board? If so, all you need is to set echocancel=yes and echocancelwhenbridged=no on your chan_dahdi.conf. If you don't... well you should! Anyway, you can turn on the echocancelling via software with echocancel=256. I strongly recommend using OSLEC in that case. You'll need to patch your DAHDI in order to use it, but it's totally worth it. On the subject of DAHDI -v- OSLEC. I never had any luck getting it to work with DAHDI 2.2.1 despite following: http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi All I ever go was a bad case of the blues :-( make[3]: *** No rule to make target `/usr/src/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/linux/drivers/dahdi/echo.o'. I guess I missed something somewhere??? Get the most recent version of Linux 2.6 kernel. Inside you'll find a directory named staging/echo. Copy that entire directory to the drivers/linux directory of the DAHDI sources. In the end you gotta have a directory named linux/drivers/staging/echo inside your DAHDI sources. I already have those :-( ls -alh /usr/src/dahdi/dahdi/linux/drivers/staging/echo drwxr-xr-x 2 root root 4.0K 2010-03-02 14:04 . drwxr-xr-x 3 root root 4.0K 2010-03-02 14:04 .. -rw-r--r-- 1 root root 5.7K 2010-03-02 14:04 bit_operations.h -rw-r--r-- 1 root root 20K 2010-03-02 14:04 echo.c -rw-r--r-- 1 root root 7.2K 2010-03-02 14:04 echo.h -rw-r--r-- 1 root root 7.4K 2010-03-02 14:04 fir.h -rw-r--r-- 1 root root 251 2010-03-02 14:04 Kconfig -rw-r--r-- 1 root root 29 2010-03-02 14:04 Makefile -rw-r--r-- 1 root root 14K 2010-03-02 14:04 mmx.h -rw-r--r-- 1 root root 2.8K 2010-03-02 14:04 oslec.h -rw-r--r-- 1 root root 367 2010-03-02 14:04 TODO To be sure I copied them again... cp -rf /usr/src/dahdi/linux-2.6.28/drivers/staging/echo/* /usr/src/dahdi/dahdi/linux/drivers/staging/echo (the /dahdi/dahdi is not a typo...) But still no dice :-( /usr/src/dahdi/dahdi# make make -C linux all make[1]: Entering directory `/usr/src/dahdi/dahdi/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/dahdi/dahdi/linux/drivers/dahdi/firmware' make -C /lib/modules/2.6.27-7-server/build SUBDIRS=/usr/src/dahdi/dahdi/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi/dahdi/linux/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[2]: Entering directory `/usr/src/linux-headers-2.6.27-7-server' make[3]: *** No rule to make target `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'. Stop. make[2]: *** [_module_/usr/src/dahdi/dahdi/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/linux-headers-2.6.27-7-server' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi/dahdi/linux' make: *** [all] Error 2 It would be nice to resolve this - but it's probably beyond my understanding and ability. Actually - looking at that make[3]: *** No rule to make target `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.c', needed by `/usr/src/dahdi/dahdi/linux/drivers/dahdi/echo.o'. Stop. There is no echo.c in /usr/src/dahdi/dahdi/linux/drivers/dahdi/ - That file is in /usr/src/dahdi/dahdi/linux/drivers/staging/echo/ I've followed this with care: http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi So I'm stumped... -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote: On Tue, 2 Mar 2010, Brian wrote: It would be nice to resolve this - but it's probably beyond my understanding and ability. Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi directory? Gordon Ah Gordon! Thank God you are here! No my friend, I did not. I was blindly following this http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi But looking at Kbuild all it has in it is... FILE CONTENTS obj-m += echo.o /FILE CONTENTS So I don't have two lines to uncomment ??? Methinks something ain't right here. Colombo would be proud of me. It appears there are issues with make -v- the path of echo.c + echo.o from my limited comprehension of such matters. Perhaps I'll whirl it again, this time unpacking to /usr/src/dahdi rather than /usr/src/dahdi/dahdi. I had adjusted the paths but just in case.. .. .. .. -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . Check dmesg on your system for messages like: VPM400: Support Enabled/Disabled VPM450: Support Enabled/Disabled That should tell you if the hardware echo cancellation is working or not. The TE410P does not have hardware echo cancellation the model was TE411P. If you can open the server you should be able to see if the card has a daughter board installed which is the echo module. regards Dhaval -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part -- _ -- 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 cancellation on DAHDI
On Tue, 2010-03-02 at 16:49 +, Brian wrote: On Tue, 2010-03-02 at 16:27 +, Gordon Henderson wrote: On Tue, 2 Mar 2010, Brian wrote: It would be nice to resolve this - but it's probably beyond my understanding and ability. Did you un-comment the 2 lines in Kbuild in the ...linux/drivers/dahdi directory? Gordon Ah Gordon! Thank God you are here! No my friend, I did not. I was blindly following this http://www.rowetel.com/ucasterisk/oslec.html#install_dahdi But looking at Kbuild all it has in it is... FILE CONTENTS obj-m += echo.o /FILE CONTENTS So I don't have two lines to uncomment ??? Methinks something ain't right here. Colombo would be proud of me. It appears there are issues with make -v- the path of echo.c + echo.o from my limited comprehension of such matters. Perhaps I'll whirl it again, this time unpacking to /usr/src/dahdi rather than /usr/src/dahdi/dahdi. I had adjusted the paths but just in case.. .. .. .. My issue was nothing more complex than having downloaded the full dahdi package. The result is it unpacks to: /usr/src/dahdi/linux/drivers/staging - not /usr/src/dahdi/drivers/staging. Fix was nothing more simple than moving the contents of /usr/src/dahdi/linux/ to /usr/src/dahdi/ and the 'howto' worked pretty much like a charm :-) -- _ -- 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 cancellation on DAHDI
Hi, Carlos I checked dmesg on my server and i found following message what is meaning for this ? i cant understand VPM400: Not Present VPM450: echo cancellation for 128 channels VPM450: hardware DTMF disabled. VPM450: Present and operational servicing 4 span(s) regards Dhaval On Tue, Mar 2, 2010 at 10:25 PM, Carlos Chavez cur...@telecomabmex.comwrote: On Tue, 2010-03-02 at 15:44 +0530, DHAVAL INDRODIYA wrote: Dear All, How can we know the On board supports echo cancellation I have Digium, Inc. Wildcard TE410P quad-span T1/E1/J1 card 3.3V (rev 02) board all working fine but sometimes i got echo when user are calling a PRI. is there any way to know on board echo cancellation . Check dmesg on your system for messages like: VPM400: Support Enabled/Disabled VPM450: Support Enabled/Disabled That should tell you if the hardware echo cancellation is working or not. The TE410P does not have hardware echo cancellation the model was TE411P. If you can open the server you should be able to see if the card has a daughter board installed which is the echo module. regards Dhaval -- _ -- 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 -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo cancellation in a sip channel
Hello all, I have a Linksys spa3102 with one FXS and one FXO port. The problem is that I have a lot of echo when using the fxo port, the sound is of very low quality. So, since I am passing from a FXO port to a SIP channel I ask: is there any Sip echo canceler software for asterisk?? Thanks in advance. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo cancellation in a sip channel
Problem solved. :) I was adding to the pstn line a gain of 6 DB for both sides. It has to be less than zero. After that the echo almost disappeared. 2010/1/21 Alexandre Rodrigues alex...@gmail.com Hello all, I have a Linksys spa3102 with one FXS and one FXO port. The problem is that I have a lot of echo when using the fxo port, the sound is of very low quality. So, since I am passing from a FXO port to a SIP channel I ask: is there any Sip echo canceler software for asterisk?? Thanks in advance. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo on Polycom phones
We are using Polycom 550 and 650 phones and OSLEC echo cancellation software with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is always from our voice echoing back to us. How can I fix this echo? I have tried installing the VPMADT032 module on our TE121 card, but that made it worse. Thank you! This is my chan_dahdi.conf: [channels] context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=no echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 This is part of my sip.cfg file: volume voice.volume.persist.handset=1 voice.volume.persist.headset=1 voice.volume.persist.handsfree=1/ gains voice.gain.rx.analog.handset=0 voice.gain.rx.analog.headset=0 voice.gain.rx.analog.chassis=0 voice.gain.rx.analog.chassis.IP_300=-6 voice.gain.rx.analog.chassis.IP_4000=3 voice.gain.rx.analog.chassis.IP_430=0 voice.gain.rx.analog.chassis.IP_650=0 voice.gain.rx.analog.chassis.IP_601=6 voice.gain.rx.analog.ringer=0 voice.gain.rx.analog.ringer.IP_300=-6 voice.gain.rx.analog.ringer.IP_4000=3 voice.gain.rx.analog.ringer.IP_430=0 voice.gain.rx.analog.ringer.IP_650=0 voice.gain.rx.analog.ringer.IP_601=6 voice.gain.rx.digital.handset=-15 voice.gain.rx.digital.headset=-21 voice.gain.rx.digital.chassis=0 voice.gain.rx.digital.chassis.IP_4000=0 voice.gain.rx.digital.chassis.IP_430=0 voice.gain.rx.digital.chassis.IP_650=6 voice.gain.rx.digital.chassis.IP_601=0 voice.gain.rx.digital.ringer=-21 voice.gain.rx.digital.ringer.IP_4000=-21 voice.gain.rx.digital.ringer.IP_430=-21 voice.gain.rx.digital.ringer.IP_650=-12 voice.gain.rx.digital.ringer.IP_601=-21 voice.gain.rx.analog.handset.sidetone=-14 voice.gain.rx.analog.headset.sidetone=-24 voice.gain.tx.analog.handset=12 voice.gain.tx.analog.headset=3 voice.gain.tx.analog.chassis=3 voice.gain.tx.analog.chassis.IP_300=0 voice.gain.tx.analog.chassis.IP_4000=3 voice.gain.tx.analog.chassis.IP_430=42 voice.gain.tx.analog.chassis.IP_650=36 voice.gain.tx.analog.chassis.IP_601=0 voice.gain.tx.digital.handset=0 voice.gain.tx.digital.headset=0 voice.gain.tx.digital.chassis=3 voice.gain.tx.digital.chassis.IP_4000=0 voice.gain.tx.digital.chassis.IP_430=-3 voice.gain.tx.digital.chassis.IP_650=0 voice.gain.tx.digital.chassis.IP_601=6 voice.gain.tx.analog.preamp.handset=14 voice.gain.tx.analog.preamp.headset=23 voice.gain.tx.analog.preamp.chassis=32 voice.gain.tx.analog.preamp.chassis.IP_430=32 voice.gain.tx.analog.preamp.chassis.IP_601=32/ AEC voice.aec.hs.enable=0 voice.aec.hs.lowFreqCutOff=100 voice.aec.hs.highFreqCutOff=7000 voice.aec.hs.erlTab_0_300=-24 voice.aec.hs.erlTab_300_600=-24 voice.aec.hs.erlTab_600_1500=-24 voice.aec.hs.erlTab_1500_3500=-24 voice.aec.hs.erlTab_3500_7000=-24 voice.aec.hd.enable=0 voice.aec.hd.lowFreqCutOff=100 voice.aec.hd.highFreqCutOff=7000 voice.aec.hd.erlTab_0_300=-24 voice.aec.hd.erlTab_300_600=-24 voice.aec.hd.erlTab_600_1500=-24 voice.aec.hd.erlTab_1500_3500=-24 voice.aec.hd.erlTab_3500_7000=-24 voice.aec.hf.enable=1 voice.aec.hf.lowFreqCutOff=100 voice.aec.hf.highFreqCutOff=7000 voice.aec.hf.erlTab_0_300=-6 voice.aec.hf.erlTab_300_600=-6 voice.aec.hf.erlTab_600_1500=-6 voice.aec.hf.erlTab_1500_3500=-6 voice.aec.hf.erlTab_3500_7000=-6/ AES voice.aes.hs.enable=0 voice.aes.hs.duplexBalance=7 voice.aes.hd.enable=0 voice.aes.hd.duplexBalance=0 voice.aes.hf.enable=1 voice.aes.hf.duplexBalance.0=9 voice.aes.hf.duplexBalance.1=8 voice.aes.hf.duplexBalance.2=7 voice.aes.hf.duplexBalance.3=6 voice.aes.hf.duplexBalance.4=5 voice.aes.hf.duplexBalance.5=4 voice.aes.hf.duplexBalance.6=3 voice.aes.hf.duplexBalance.7=2 voice.aes.hf.duplexBalance.8=1 voice.aes.hf.duplexBalance.IP_4000.0=10 voice.aes.hf.duplexBalance.IP_4000.1=9 voice.aes.hf.duplexBalance.IP_4000.2=8 voice.aes.hf.duplexBalance.IP_4000.3=7 voice.aes.hf.duplexBalance.IP_4000.4=6 voice.aes.hf.duplexBalance.IP_4000.5=5 voice.aes.hf.duplexBalance.IP_4000.6=4 voice.aes.hf.duplexBalance.IP_4000.7=3 voice.aes.hf.duplexBalance.IP_4000.8=2/ NS voice.ns.hs.enable=0 voice.ns.hs.signalAttn=-6 voice.ns.hs.silenceAttn=-9 voice.ns.hd.enable=0 voice.ns.hd.signalAttn=0 voice.ns.hd.silenceAttn=0 voice.ns.hf.enable=1 voice.ns.hf.signalAttn=-6 voice.ns.hf.silenceAttn=-9 voice.ns.hf.IP_4000.enable=1 voice.ns.hf.IP_4000.signalAttn=-6 voice.ns.hf.IP_4000.silenceAttn=-9/ AGC voice.agc.hs.enable=0 voice.agc.hd.enable=0 voice.agc.hf.enable=0/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo on Polycom phones
hin lee wrote: We are using Polycom 550 and 650 phones and OSLEC echo cancellation software with Asterisk. Occasionally, we get echo on our PRI phone calls. The echo is always from our voice echoing back to us. How can I fix this echo? I have tried installing the VPMADT032 module on our TE121 card, but that made it worse. rxgain=0.0 txgain=0.0 Try reducing your transmit gains (txgain=-4.0 Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
I think you need to remove the line echocanceller in system.conf You could also try to use fxotune, it'a really improving things. You also need to put echocancel=yes in chan_dahdi.conf This is a PRI, so fxotune is not the thing to use in this case. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
If I installed a Digium echo cancellation module on my TE121 card, do I need to remove the echocanceller line under the system.conf? How should I have it? This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Thank you! Hin From: hin lee hi...@yahoo.com To: noahisaacmil...@gmail.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, December 11, 2009 8:56:13 AM Subject: Re: [asterisk-users] Echo issue The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module? Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
Hi, I think you need to remove the line echocanceller in system.conf You could also try to use fxotune, it'a really improving things. You also need to put echocancel=yes in chan_dahdi.conf Matthieu NICAISE Responsable technique GSM : 06 72 19 09 55 techni...@thinkrosystem.com Thinkro System http://www.thinkrosystem.com/ Le 15 déc. 09 à 23:15, hin lee a écrit : If I installed a Digium echo cancellation module on my TE121 card, do I need to remove the echocanceller line under the system.conf? How should I have it? This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Thank you! Hin From: hin lee hi...@yahoo.com To: noahisaacmil...@gmail.com Cc: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Fri, December 11, 2009 8:56:13 AM Subject: Re: [asterisk-users] Echo issue The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module? Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? What kind of settings do you recommend for the txgain and rxgain? Ideally, you will need to measure to find out what settings you want. See this page on the wiki (see the note on values for PRI circuits): http://www.voip-info.org/wiki/view/Asterisk+zapata+gain+adjustment (use dahdi_monitor instead of ztmonitor) You can also just experiment with different values. Change just one setting at a time, and then reload Dahdi. Try this to start: txgain = 0.0 rxgain = 1.0 and then on the asterisk cli, enter: module reload chan_dahdi.so If that doesn't help, try increasing to rxgain=2.0. Keep going until it sounds better. You may want to try negative values for txgain. Do I make the gain changes in chan_dahdi.conf? Yes. Make sure to put them before your channel numbers. You can specify values on a per-channel basis. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. Are you routing internal calls from SIP - DAHDI - SIP? The digium echo module will not have any effect on pure SIP - SIP calls. Do you have acoustic echo cancellation active on the Polycom phones? Internal calls should be SIP to SIP. Yes we do have the acoustic echo cancellation active on the Polycom phones. This is my system.conf: bchan=1-23 dchan=24 echocanceller=mg2,1-23 Did you use these same settings when you were using the hardware echo module? Yes, I believe so. I asked an Asterisk expert to make sure everything is working correctly when installing the hardware module. If the setting don't look correct, what should be there when we use the hardware module? Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
The echo between our extensions (using Polycom 550 handsets) disappears once I removed the Digium echo module. We are still experiencing some echo on land line calls, using dahdi to connect to our PRI circuit. What kind of settings do you recommend for the txgain and rxgain? Do I make the gain changes in chan_dahdi.conf? Thank you! This is my system.conf: === # Span 1: WCT1/0 Wildcard TE121 Card 0 (MASTER) span=1,1,0,esf,b8zs # termtype: te bchan=1-23 dchan=24 echocanceller=mg2,1-23 # Span 2: WCTDM/0 Wildcard AEX410 Board 1 fxoks=25 echocanceller=mg2,25 fxoks=26 echocanceller=mg2,26 fxoks=27 echocanceller=mg2,27 # channel 28, WCTDM/0/3, no module. # Global data loadzone= us defaultzone= us This is my chan_dahdi.conf [trunkgroups] [channels] context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no faxdetect=incoming echotraining=800 rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 ;Uncomment these lines if you have problems with the disconection of your analog lines ;busydetect=yes ;busycount=3 immediate=no #include dahdi-channels.conf #include chan_dahdi_additional.conf From: Noah Miller noahisaacmil...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, December 8, 2009 7:37:28 AM Subject: Re: [asterisk-users] Echo issue Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo issue
On Tue, Dec 8, 2009 at 7:37 AM, Noah Miller noahisaacmil...@gmail.com wrote: Hi - I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. You can likely eliminate most echo on a PRI by setting txgain and rxgain. Are you using dahdi or zaptel? If Dahdi, what do your system.conf and chan_dahdi.conf look like? If zaptel, what do your zaptel.conf and zapata.conf look like? When you say you have echo on calls that are internal extension to internal extension, are the endpoints using dahdi/zaptel or some voip technology (sip, iax, mgcp, skinny, etc)? If voip, any echo is acoustically generated by the endpoints themselves. On voip calls I've often had this happen when the endpoints are using headsets, or have gain levels set very high. - Noah We found ourselves in a similar situation during our rollout and solved it with a quad-span Ditech echo-cancellation appliance (http://www.ditechnetworks.com/products/quad-2_echo-canceller.html). It's a couple of grand, but after months of playing with software EC, the hardware modules and every zaptel setting we could find, this appliance removed echo like flipping a switch. The metrics we later obtained from it clearly showed that we simply had tail on a long loop to an old CO switch that exceeded the maximum 128ms that either software EC or the hardware module could handle. The side benefits are that we get all sorts of metrics from the appliance, and we also get adaptive gain, which solved another problem we had with trying to find gain settings that suited both softly-spoken and strident users. The support from Ditech was excellent and we haven't looked back. CP ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo issue
I am having echo issues on our Asterisk box using a PRI circuit. I was using the software echo cancellation and that helped a bit but didn't solve it completely. So I went and bought a Digium echo cancellation module for the TE121 card. That made it even worst, getting more echo on external calls and between internal extension to extension. The echo doesn't happen all the time, but enough to get complaints from our users. Completely fed up with the issue, I removed the module from the card. Can someone guide me on how to fix/tune/address the echo issues. Thank you! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo cancellation on DAHDI
hey , all i have one issue on incoming DAHDI PRI it works fine many times but sometimes it creates bad audio and also having echo in line also recording going to be disturbed by this i cannot understand this properly can any one have solutions and how to improve this also how to monitor lines All dahdi lines and any causes for telco i am fro INDIA . regards Dhaval ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo and Playtones not working on SIP after upgrade
Hello list I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Over IAX2, both Echo and Playtones works fine on this same extension and system! I googled and tried several things, but nothing seems to work. Basically the log shows it is working, there are no errors or warnings, but there is no sound at all. No beeps, no Echo. Calls, voicemail, moh, and everything else we are using works just fine. We are using Asterisk 1.4.21.2~dfsg-3 (on debian stable), SIP channels with both grandstream and soft phones. Everything on the same network segment. Codec does not seem to affect this behavior (tried them all) Any clues? Thanks! -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and Playtones not working on SIP after upgrade
On Monday 07 September 2009 09:36:53 Ivan Stepaniuk wrote: I had the following echo-test extension on my Asterisk 1.2 setup. exten = 1003,1,Wait(1) exten = 1003,n,Playtones(!1050/1000) exten = 1003,n,Wait(1) exten = 1003,n,StopPlaytones exten = 1003,n,Echo exten = 1003,n,Hangup After migrating my testing server to Asterisk 1.4, and a minor extensions.conf update, everything works just fine. Except for the Playtones and the Echo applications on SIP channels. Try adding an Answer() in there, before the first Playtones. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo
Echo Cancellation: 128 taps unless TDM bridged, currently ON The "currently ON" is telling you that the echo canceller is active. You could try changing echotraining to no in chan_dahdi.conf as well. What were you running before you upgraded? So, Asterisk doesn't start echo canceling a line until it is in use? I thought that might be the case. I was running Zaptel before this, not sure what version. I upgrade to Dahdi. The echo was present in Zaptel, but not as bad. Does anyone have any experience with hardware echo cancel modules? Are they better/worse than software? What would be the best solution to remove echo? Jason Baker IT Coordinator Glastender, Inc. 5400 North Michigan Road Saginaw, Michigan 48604 USA Phone: 989.752.4275 ext. 228 Fax: 989.752.4276 www.glastender.com Dave Fullerton wrote: Jason Baker wrote: I recently upgraded my Asterisk system to Dahdi and now I have an echo problem. I am running Asterisk 1.4.25 with Dahdi Complete 2.2.0, on a Digium TE121B PCI express card with a HARDWARE echo cancellation module. All this is housed on a CentOS 5.5 box, 2.6.18 Kernel. My incoming phone service is an ATT PRI (24 channel T1). My configs: chan_dahdi.conf* [channels] ; configuration for T1 card as PRI language = en group = 1 echocancel = yes echotraining = yes signalling = pri_cpe switchtype = 4ess usecallerid = yes context = incoming channel = 1-23 ***/etc/dahdi/system.conf* loadzone=us defaultzone=us span=1,0,0,esf,b8zs bchan=1-23 dchan=24 When I run dahdi_cfg -vvv I get the following: DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0.1 Echo Canceller(s): MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Echo Canceler: none) (Slaves: 01) snip Channel 23: Clear channel (Default) (Echo Canceler: none) (Slaves: 23) Channel 24: D-channel (Default) (Echo Canceler: none) (Slaves: 24) 24 channels to configure. Setting echocan for channel 1 to none snip Setting echocan for channel 24 to none It is showing MG2 as the echo canceller, even though I don't have an echo canceller specified. Is that the harwdare module? Do I even need to specify an echo canceller in the configs if I have a hardware echo module? MG2 is a software canceller. I don't think that line means that MG2 is being used on all your channels. If you look at the Channel map it says "(Echo Canceler: none)". If it had been set to MG2 you would see MG2 instead of none. You do not need to specify an echo canceller in system.conf when you have a hardware canceller. One thing I would check is to make sure asterisk is activating the echo canceller when a call is in progress. To do this execute "core show channels" at the asterisk command line (make sure someone on the system has placed a call on the PRI). Look for a DAHDI/#-x line. Then execute "dahdi show channel #" where # is the channel number. You'll get a screen full of output. Look for a line that looks like this (it will be near the end): Echo Cancellation: 128 taps unless TDM bridged, currently ON The "currently ON" is telling you that the echo canceller is active. You could try changing echotraining to no in chan_dahdi.conf as well. What were you running before you upgraded? -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
