Re: [asterisk-users] Bad Echo between SIP calls
First of all, Alex, sorry for not seeing your reply. Nearly two weeks ago now :( Honestly, with canreinvite=yes, I'm not sure what is meant by the signalling still travels through asterisk... I would ASSUME that includes out-of-band dtmf as well. Sorry! Moj Alex Crow wrote: Moj, Does this mean that even out-of-band DTMF still gets sent SIP-phone--SIP-phone without Asterisk hearing them? (eg RFC DTMF, can't remember the number right now) Forgive me for butting into this thread but this is interesting... Cheers Alex On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan Company, LLC wrote: theoretically, with canreinvite=yes, it's phone - phone. with canreinvite=no, it's phone - asterisk - phone. BUT there are a few reasons which canreinvite=yes will not be this way. If for example you have a T or a t in the Dial string, asterisk will _remain_ in the media path so it can still detect the DTMF requests for transfer. Moj Deepak Naidu wrote: Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak */Matthew Fredrickson [EMAIL PROTECTED]/* wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] Bad Echo between SIP calls
Darryl Dunkin wrote: What are the end devices? That seems to have been lost here. The real issue is the handsets as those are the devices introducing the echo (the only analog players here). Most likely a volume or gain issue on those handsets, what SIP devices are the echo issues between? If both people hear echo, both devices are at fault, if one person hears it, it is the other end at fault. Thats true. Echo usually appear if your end devices uses additional voice amplifier. Try try to set -2 db on audio input if your gateways have internal gain control. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Deepak Naidu Sent: Tuesday, June 12, 2007 19:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad Echo between SIP calls I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. -- Deepak Darryl Dunkin [EMAIL PROTECTED] wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --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 What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On Jun 11, 2007, at 7:26 PM, Zeeshan Zakaria wrote: Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. Those issues should now be fixed. We have been working very hard to make sure that we have gotten rid of them. If you are still seeing any problems with Digium cards related to interrupts, sharing, conflicts, motherboards, etc, then for sure let us know so that we can fix them. To be perfectly frank, with the changes in the last 3-6 months that we have made in our drivers our cards should be running very well. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
theoretically, with canreinvite=yes, it's phone - phone. with canreinvite=no, it's phone - asterisk - phone. BUT there are a few reasons which canreinvite=yes will not be this way. If for example you have a T or a t in the Dial string, asterisk will _remain_ in the media path so it can still detect the DTMF requests for transfer. Moj Deepak Naidu wrote: Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak */Matthew Fredrickson [EMAIL PROTECTED]/* wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Moj, Does this mean that even out-of-band DTMF still gets sent SIP-phone--SIP-phone without Asterisk hearing them? (eg RFC DTMF, can't remember the number right now) Forgive me for butting into this thread but this is interesting... Cheers Alex On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan Company, LLC wrote: theoretically, with canreinvite=yes, it's phone - phone. with canreinvite=no, it's phone - asterisk - phone. BUT there are a few reasons which canreinvite=yes will not be this way. If for example you have a T or a t in the Dial string, asterisk will _remain_ in the media path so it can still detect the DTMF requests for transfer. Moj Deepak Naidu wrote: Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak */Matthew Fredrickson [EMAIL PROTECTED]/* wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. Transact is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 1200 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Remember to restart asterisk and zaptel when you make this change. On 6/12/07, Matt [EMAIL PROTECTED] wrote: I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On Tue, Jun 12, 2007 at 07:44:02PM -0400, Matt wrote: On 6/12/07, Matt [EMAIL PROTECTED] wrote: I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes Remember to restart asterisk and zaptel when you make this change. Actually: just reload. or even: reload chan_zap.so (or module reload chan_zap.so) Thi is true for most zapata.conf settings. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. -- Deepak Darryl Dunkin [EMAIL PROTECTED] wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. - From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --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 - What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
What are the end devices? That seems to have been lost here. The real issue is the handsets as those are the devices introducing the echo (the only analog players here). Most likely a volume or gain issue on those handsets, what SIP devices are the echo issues between? If both people hear echo, both devices are at fault, if one person hears it, it is the other end at fault. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Tuesday, June 12, 2007 19:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bad Echo between SIP calls I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. -- Deepak Darryl Dunkin [EMAIL PROTECTED] wrote: This should only be for TDM to TDM calls, SIP to SIP calls don't use the zaptel driver. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Tuesday, June 12, 2007 16:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls I don't see this listed anywhere here in the replies so. In your zapata.conf file try changing: echocancelwhenbridged=no to: echocancelwhenbridged=yes ___ --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 What kind of emailer are you? Find out today - get a free analysis of your email personality. Take the quiz at the Yahoo! Mail Championship http://uk.rd.yahoo.com/mail/uk/taglines/default/championships/quiz/*htt p://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/ . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Deepak Naidu wrote: I like the way people replied to this message of mine. It seems this thread is going back to the hybrid echo issue(no this is not the problem). As said by many ZAP is not in picture for SIP--SIP ie Ext-Ext internal calls. To put my inputs I did tons of QA on this issue to ground on whats the source. Its not just the phone or only the network but may be both. I am not sure how Asterisk would contribute to this. At time for a given 2 internal extension there was no echo but suddenly turned up. People dialing on my phone have echo but not on other at the same time I have few phones which I dial no echo. So ya dont know whats wrong. Thanks all for your inputs sharing ur experience. I suggest you try a different machine/motherboard. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
The echo cancellation card is for SIP-Zap calls only, no echo cancellation is done in Asterisk for SIP only calls. SIP to SIP, media is just passed through the server untouched (using media flow through, which is the option in sip.conf of canreinvite=no) if you are not handling any translation, even when handling translation between SIP calls there shouldn't be any echo cancellation done in Asterisk for SIP only calls. The place to look at would be the remote SIP devices which is typically what is adding the echo, this is usually a gain issue of some sort depending on which handsets you are using. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Monday, June 11, 2007 16:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Sounds crazy right? even was I, more over support guy logged in unloaded the zap modules to test them, still an echo. Ya, I was clear saying that we have SIP--- SIP issue ie internal extension echo problem. It seems the echo with SIP--SIP has many factors. I am just curios to eliminate any possibility of Asterisk failing to cancel the echo. OK, one question here howz the call flow when a SIP---SIP call is established ie. is the connection between 2 phones when an Internal call is made or does the SIP call goes via Asterisk once the SIP--SIP call is establised. -- Deepak Matthew Fredrickson [EMAIL PROTECTED] wrote: On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for.. Are you sure that they understood that you were having this problem between 2 SIP endpoints? That advice only makes sense to test if one side is Zap and the other side is SIP. --- Matthew Fredrickson Software Engineer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do something with the kernel and USB modules and something needed to be fixed in BIOS and zaptel settings somewhere (not in zapata or zaptel confs) so that it can have a properly working timing source. I don't remember the details now but I remember I managed to fix it by building a different kernel version on that server after installaing some other version of zaptel, disabling USB modules on the motherboard, fixing something in zaptel Makefile, disabling unused modules in /etc/sysconfig/zaptel. I don't remember what else I did. but echo and other problems disappeared after whatever I did. It was about 2 years ago and I remember how frustrating it was. Anyways, I guess once you upgraded your hardware, something changed in zaptel settings somewhere which is now effecting the SIP-SIP calls and resulting in echo. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Hey thanx for sharing your troubleshooting. Ya over days I kind of did some QA. There are SIP--SIP echo's between random phones. We have 75 phones of Polycom 501. I think might be the network or combination of network polycom creating this. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. This is an entie new setup by me, the old one was using 1.4 build I am using 1.2 build both are different server. -- Deepak Zeeshan Zakaria [EMAIL PROTECTED] wrote: Once upon a time I used to have a lot of SIP-SIP calls issues, which not always but sometimes included echo problems. There were no zap devices on the server. Googling and struggling to fix it, I found out that it was because of timing issues and ztdummy was not working properly. It had to do something with the kernel and USB modules and something needed to be fixed in BIOS and zaptel settings somewhere (not in zapata or zaptel confs) so that it can have a properly working timing source. I don't remember the details now but I remember I managed to fix it by building a different kernel version on that server after installaing some other version of zaptel, disabling USB modules on the motherboard, fixing something in zaptel Makefile, disabling unused modules in /etc/sysconfig/zaptel. I don't remember what else I did. but echo and other problems disappeared after whatever I did. It was about 2 years ago and I remember how frustrating it was. Anyways, I guess once you upgraded your hardware, something changed in zaptel settings somewhere which is now effecting the SIP-SIP calls and resulting in echo. Do you have the backup of old setup without this card, which you can install and check what exactly the settings were before. Also I recommend going with Sangoma. I hear a lot of bad stories about digium cards imcompatibility with certain motherboards and conflicts with USB modules on the motherboard, and conflicts with IRQs. Thats why When I went for PRI, I used Sangoma. I've used their A101c and A101d cards, and there have never been any issues. ___ --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 - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
Best way to do this is not touch the sip.cfg, ever. Leave it as included in each release and include your overrides in a different file. Then reference your files like this in your MAC.cfg file, your file will override the sip.cfg defaults. CONFIG_FILES=phone_user.cfg,server.cfg,sip.cfg In server.cfg, if you wanted to change the server, for example: ?xml version=1.0 standalone=yes? sip voIpProt local voIpProt.local.port=/ server voIpProt.server.1.address=asterisk.yourdomain.com /voIpProt /sip From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Saturday, June 09, 2007 22:54 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls It doesn't matter if it's supported, they are all, however I have seen some echo problems after firmware upgrades, the only way to fix it was to either copy the differences or overwrite my old config files with the new ones that came with the firmware and then modify as needed for my setup. On 6/10/07, Deepak Naidu [EMAIL PROTECTED] wrote: The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : http://us.i1.yimg.com/us.yimg.com/i/mesg/tsmileys2/03.gif -- Deepak C F [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf . Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships http://uk.rd.yahoo.com/mail/uk/taglines/default/championships/games/*ht tp://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/ . Plus: play games and win prizes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Deepak Naidu wrote: Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. Make SURE you have the handset plugged into the handset port of the phone, not the headset port of the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies [EMAIL PROTECTED] wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stephen Davies Sent: Saturday, June 09, 2007 4:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve Most likely the phones. Is it worse on speakerphone? Are they cheap like the Grandstream 101s? Try with a couple softphones and headsets, any better. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
Do you have reinvites enabled? Are you running this over a linksys four port SoHo router/switch or something? Thanks, Steve Totaro http://www.asteriskhelpdesk.com http://www.asteriskhelpdesk.com/component/option,com_wrapper/Itemid,37/ KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu Sent: Saturday, June 09, 2007 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Bad Echo between SIP calls Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. -- Deepak Stephen Davies [EMAIL PROTECTED] wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Yahoo! Answers - Get better answers from someone who knows. Try it now http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc 2VjA21haWwEc2xrA3RhZ2xpbmU . ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Yeah I have made sure its the correct port. We have 75 polycoms currently. ? the SIP-to-SIP echo is there. -- Deepak Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Deepak Naidu wrote: Steve I understand your theory. We have Poycom 501 phones. Prior upgrading to PRI we were till date using 4 analog lines connected with TDM card from digium no echo for pure SIP to SIP lines. Now I have TE212P which had onboard echo cancellor. I am trying make myself clear before I blame on any network. B'cos for sure we have a spegati of networks no QoS. Also the intresting thing is if I call from one extension to other dialing the main line then extension the call is crystal clear. but when dialing a direct extension its a hell of echo. Make SURE you have the handset plugged into the handset port of the phone, not the headset port of the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your freeaccount today.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bad Echo between SIP calls
reinvite is disabled. Also its a Dell PowerEdge 850 server running asterisk
connected to a Cisco switch. other network in company have Cisco Switch.
Also we have approx 75 Polycoms all over.
canreinvite=no
--
Deepak
Steve Totaro [EMAIL PROTECTED] wrote:
v\:* {behavior:url(#default#VML);} o\:* {behavior:url(#default#VML);}
w\:* {behavior:url(#default#VML);} .shape {behavior:url(#default#VML);}
st1\:*{behavior:url(#default#ieooui) }Do you have reinvites
enabled? Are you running this over a linksys four port SoHo router/switch or
something?
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
KB3OPB
-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Deepak Naidu
Sent: Saturday, June 09, 2007 4:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Bad Echo between SIP calls
Steve I understand your theory. We have Poycom 501 phones. Prior
upgrading to PRI we were till date using 4 analog lines connected with TDM card
from digium no echo for pure SIP to SIP lines.
Now I have TE212P which had onboard echo cancellor.
I am trying make myself clear before I blame on any network. B'cos for
sure we have a spegati of networks no QoS. Also the intresting thing is if I
call from one extension to other dialing the main line then extension the
call is crystal clear. but when dialing a direct extension its a hell of echo.
--
Deepak
Stephen Davies [EMAIL PROTECTED] wrote:
On 09/06/07, Deepak Naidu wrote:
Ya, I have done that, below is zapata.conf. Also we had an TMP card with
analog lines. SIP cals were great on them. now when we switched over.
SIP calls have echo.. which shouldnt be at all.
If you are getting echo on pure SIP to SIP calls, there's no point in
fiddling around with your zapta.conf. That file is for configuring
chan_zap, which is used to talk to Zap/ channels. Your calls are SIP
to SIP so the zap channel and your PRI aren't being used at all.
SIP calls are pure digital 4 wire lines so no electrical (Hybrid)
echo will be present. The phones should not generate echo. If they
are, they are presumably nasty phones (what kind are they?) and you
should get properly made phones.
Steve
___
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Re: [asterisk-users] Bad Echo between SIP calls
Stephen Davies wrote: On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood [EMAIL PROTECTED] wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu [EMAIL PROTECTED] wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : -- Deepak C F [EMAIL PROTECTED] wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championships. Plus: play games and win prizes.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
It doesn't matter if it's supported, they are all, however I have seen some echo problems after firmware upgrades, the only way to fix it was to either copy the differences or overwrite my old config files with the new ones that came with the firmware and then modify as needed for my setup. On 6/10/07, Deepak Naidu [EMAIL PROTECTED] wrote: The sip config firmware are the supported one for the existing firmware. If you have any stable working Polycom 501 SIP without echo between SIP--SIP wouldnt mind to share the sip.cfg, sip.ld bootrom would be great, bcos I have not got concreate resolution for this issue. Hope I can resolve this mess. Feels bad when one does best in aggregating things some louzy device screws up... Oh my frustation is comming on mail : -- Deepak *C F [EMAIL PROTECTED]* wrote: Are the config files you are using with the phones what was meant with that firmware? or did you upgrade the firmware and reused the old config files? On 6/9/07, Steve Underwood wrote: Stephen Davies wrote: On 09/06/07, Deepak Naidu wrote: Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. If you are getting echo on pure SIP to SIP calls, there's no point in fiddling around with your zapta.conf. That file is for configuring chan_zap, which is used to talk to Zap/ channels. Your calls are SIP to SIP so the zap channel and your PRI aren't being used at all. SIP calls are pure digital 4 wire lines so no electrical (Hybrid) echo will be present. The phones should not generate echo. If they are, they are presumably nasty phones (what kind are they?) and you should get properly made phones. By this measure most phones are nasty. The handset should be echo cancelled, to prevent leakage of the earpiece into the mike. It is getting less and less common to do this, now. Polycoms, Sipuras, Snoms, you name it, they do it badly. Many are not too annoying until someone turns the volume up. Call someone a little hard of hearing and you will hear echo. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- New Yahoo! Mail is the ultimate force in competitive emailing. Find out more at the Yahoo! Mail Championshipshttp://uk.rd.yahoo.com/mail/uk/taglines/default/championships/games/*http://uk.rd.yahoo.com/evt=44106/*http://mail.yahoo.net/uk/. Plus: play games and win prizes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bad Echo between SIP calls
Hi, We have a PRI connection when its was on test networks we had echo problems withoutside line. So I bought a TE212P card resolve the echo problem. Which did to an extent. Its using asterisk 1.2.18 RHEL4-Update 4. But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. I am not sure whats the problem. Also there's slight echo when calling Digium support. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Has anyone come through this issue. -- Deepak - Yahoo! Answers - Get better answers from someone who knows. Tryit now.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bad Echo between SIP calls
Ya, I have done that, below is zapata.conf. Also we had an TMP card with analog lines. SIP cals were great on them. now when we switched over. SIP calls have echo.. which shouldnt be at all. [channels] language=en #include zapata_additional.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes callerid=asreceived echocancelwhenbridged=no echotraining=128 ;rxgain=-3.0 ;txgain=-7.0 group=0 channel=1-23 -- Deepak Alex Balashov [EMAIL PROTECTED] wrote: On Sat, 9 Jun 2007, Deepak Naidu wrote: But now when we are live, there is a terrible echo between 2 SIP calls. If I call the same extension from outside the voice is clear. My impression is that the transcoding that takes place between two purely software SIP calls never goes through the TE212P card. There are probably echo cancellation options you can enable that are relevant to software channels. I distantly recall there even being some stuff youc an uncomment in the source. Totally lost Digium says we need to remove the echo module to resolve SIP echo problems. Then ? the heck we pay for... Not sure why Digium would say that. -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --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 - All New Yahoo! Mail Tired of unwanted email come-ons? Let our SpamGuard protect you.___ --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
