I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a ISDN (i4l) card. As I said, I hear an echo which is sometimes worse than other times even throughout a single call, but reports from every other person I speak to says they get a very bad echo of their own voice as well. Any hints? Does the i4l asterisk driver support echo cancellation? I currently have: echocancel=yes in my zapata.conf file. I haven't tried dialling from my ata186 out to the world, but other people say they do still get an echo... Regards, Adam
On Wed, 19 Feb 2003 around 20:42:58 +1100, Adam Goryachev wrote:> I used to get bad echo and also voice breaking up during a conversation, so > I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved > the voice breaking up issues but still seem to have a fairly bad echo on > *both* sides of the conversation. > > Currently I am only using the system for outbound calls, so I am always > calling from the digium USB port phone and out to the telco using a ISDN > (i4l) card. As I said, I hear an echo which is sometimes worse than other > times even throughout a single call, but reports from every other person I > speak to says they get a very bad echo of their own voice as well. > > Any hints? > > Does the i4l asterisk driver support echo cancellation? I currently have: > echocancel=yes > in my zapata.conf file.i4l uses the modem.conf file, not the zapata.conf file. and no, it does not echo cancelation, i'm waiting for Mark to seperate it out from the zap driver, like he did for the dsp functions.> I haven't tried dialling from my ata186 out to the world, but other people > say they do still get an echo...The ata186 does its own echo cancelation. But it you go to the outside world over i4l, than again you introduce echo. Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7 DDA6 0378 5DEA BD3B 9A50 B416 E2D0 C3C2 For more details look at my website http://www.polyware.nl/~middelink -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 2107 bytes Desc: not available URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20030222/ac756634/attachment.bin>
Read the faq, checked the config files... can't find anything about an echo problem like this. Here's what I've got: 4 channel t1 card, span 2 going to channel bank with both fxs and fxo lines Polycom IP600 phones on same LAN with asterisk iax connection to voicepulse (T1 going out on another router) Asterisk on a 2.4GHz machine what I hear: FXO <-> FXS no echo at all FXO or FXS <-> voicepulse no echo at all IP600 <-> voicepulse no echo at all IP600 <-> FXO or FXS echo heard by IP600 caller, no echo heard by remote party IP600 <-> IP600 can't get it to work yet - SIP times out (separate issue I guess) I also did a few more tests: - I made an extension that just does Wait(). Called it from the ip600. No echo. - Used the built in Echo function. I only hear one echo. I measured the latency by recording it with a microphone. It is 100ms, which seems a bit excessive for ethernet. The fact that I hear only one echo when doing the Echo test, and no echo anywhere except in the IP600 <-> POTS path, would lead me to believe that the source of the echo is within asterisk (not in transmission or in the phones), and only when bridging SIP<->POTS. Any ideas?
I'm having bad echo between TDM and SIP. There's no echo between TDM-TDM though. I've seen this post from JTodd: ; Config notes: ; - in /usr/src/zaptel/Makefile, set KFLAGS+=-DECHO_CAN_MARK2 ; - in /usr/src/zaptel/Makefile, set KFLAGS+=-DAGGRESSIVE_SUPPRESSOR ; ; I compile with these two echo cancellation flags as it seems they ; sound better with SIP phones interacting with Zap (analog) devices. I am not into compiling, so I need to verify if I will just edit the Makefile in /usr/src/zaptel folder and execute "makefw"? or just load ztcfg after modifying the Makefile. Thanks for the usual promptness. :) I welcome other ideas to resolve this echo issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040209/27367d72/attachment.htm
Hello! There is a strong echo while calling from one X-Lite to an other via asteirsk. Why it occurs? How it can be eliminated?
Hi, OK I have been struggling with this echo problem for a while now and I need some help. we have to * machines, one is in a remote office running 5 snom 200 phones(sip), and the other is at our corporate HQ also running sip. the two machine communicate via an IAX2 trunk and the machine in our HQ has a single full PRI with digium hardware. I have tried to use the zap utilities to adjust the gain but the vu meter for rx is always pegged, the tx meter is around 50-70 %. any help would be excellent!
We get echo on outgoing calls sip -> IAX2 -> asteriskHQ -> PRI the receiving party hears only an excellent quality call. I have included our zapata.conf file it is identical on both machines. [channels] language=en context=incoming busydetect=no callprogress=no echocancel=yes echocancelwhenbridged=yes echotraining=yes transfer=yes threewaycalling=no callforward=yes cancallforward=yes immediate=no usecallerid=yes rxgain=0.0 txgain=-10.0 signalling=pri_cpe switchtype=dms100 group = 1 channel=1-23 On Wed, 2004-04-28 at 18:42, Ed Rubright wrote:> In what scenarios do you have echo? > > HQ sip phones thru PRI? > Remote sip phones thru IAX2 to PRI? > Remote sip phones to HQ sip phones? > > Ed > > > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Justin Carlson > Sent: Wednesday, April 28, 2004 5:59 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] ECHO > > Hi, > > OK I have been struggling with this echo problem for a while now and I need > some help. we have to * machines, one is in a remote office running 5 snom > 200 phones(sip), and the other is at our corporate HQ also running sip. the > two machine communicate via an IAX2 trunk and the machine in our HQ has a > single full PRI with digium hardware. I have tried to use the zap utilities > to adjust the gain but the vu meter for rx is always pegged, the tx meter is > around 50-70 %. > > any help would be excellent! > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
We too experience intermittent echo issues but have greatly reduced them. What build of Asterisk are you running? Do you have echocancel and echotraining set to on? Best regards, Ryan Thrash On Apr 28, 2004, at 7:59 AM, Justin Carlson wrote:> OK I have been struggling with this echo problem for a while now and I > need some help. we have to * machines, one is in a remote office > running 5 snom 200 phones(sip), and the other is at our corporate HQ > also running sip. the two machine communicate via an IAX2 trunk and > the > machine in our HQ has a single full PRI with digium hardware. I have > tried to use the zap utilities to adjust the gain but the vu meter for > rx is always pegged, the tx meter is around 50-70 %. > > any help would be excellent!
In what scenarios do you have echo? HQ sip phones thru PRI? Remote sip phones thru IAX2 to PRI? Remote sip phones to HQ sip phones? Ed -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Justin Carlson Sent: Wednesday, April 28, 2004 5:59 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ECHO Hi, OK I have been struggling with this echo problem for a while now and I need some help. we have to * machines, one is in a remote office running 5 snom 200 phones(sip), and the other is at our corporate HQ also running sip. the two machine communicate via an IAX2 trunk and the machine in our HQ has a single full PRI with digium hardware. I have tried to use the zap utilities to adjust the gain but the vu meter for rx is always pegged, the tx meter is around 50-70 %. any help would be excellent! _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Good day all We are running x-lite as a sof client and using the zaptel cards Each time I make a call out I get a big echo but when I get a call in there is no echo?Why is this Please Help
I've got a customer on an IAXy and another with their own Asterisk box as a PBX with an array of Cisco, GrandStream, ATCOM, and xten hard\soft phones. Same LEC, same Asterisk box on our end, same broadband provider on the client ends With no packet loss, <15 ms pings, 13 hops, the IAXy sometimes has an echo, some times not. The client with the Asterisk box... no problems at all. What could I do to figure out what's going on here? --Mike ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.
On Sun, 2006-01-15 at 22:23 -0500, Steve Totaro wrote:> Just checking....Just checking.... :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20060116/f7f82d59/attachment.pgp
Steve Totaro wrote:>Just checking.... >_______________________________________________ >--Bandwidth and Colocation provided by Easynews.com -- > >Asterisk-Users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >I was asking myself the saim thing... -- Razvan Turtureanu NOC Engineer Mobil: 072.3637714 E-mail: razvan@edata.ro ---------------------------- S.C. Edata S.R.L., Sucursala Bucuresti Str. Toamnei nr. 68, et. III, sector 2 Tel: +4031.401.6828 Fax: +4031.401.6829 E-mail: office@edata.ro Web: www.edata.ro ---------------------------- This e-mail is confidential and may contain legally privileged information. If you are not the intended recipient, you should not copy, distribute, disclose or use the information it contains. Please e-mail the sender immediately and delete this message from your system. E-mails are susceptible to corruption, interception and unauthorised amendment; we do not accept liability for any such changes, or for their consequences. You should be aware, that a company may monitor your emails and their content. Acest mesaj este confidential si poate contine informatii protejate legal. Daca nu sinteti destinatarul intentionat, nu trebuie sa copiati, difuzati, dezvaluiti sau utilizati informatiile pe care acesta le contine. Va rugam sa retransmiteti imediat mesajul expeditorului si sa-l stergeti din sistemul dvs. Mesajele sint pasibile de denaturare, interceptie sau modificare neautorizata; nu ne asumam raspunderea pentru nici o asemenea eventuala schimbare sau pentru consecintele acesteia. Trebuie sa stiti ca o companie va poate monitoriza mesajele si continutul acestora. ----------------------------
ping pong On 1/15/06, Steve Totaro <stotaro@totarotechnologies.com> wrote:> Just checking.... > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >