[on-asterisk] list archives online
The list archives are now online and searchable via mail-archive.com. See links from: http://taug.ca/discuss - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
RE: [on-asterisk] list archives online
Cool! Thanks for setting this up Simon, this will be very useful to us all. Alex __ Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] www.gearytech.com Strategic management of technology for business. -Original Message- From: Simon P. Ditner [mailto:[EMAIL PROTECTED] Sent: Friday, July 13, 2007 11:48 AM To: asterisk@uc.org; Subject: [on-asterisk] list archives online The list archives are now online and searchable via mail-archive.com. See links from: http://taug.ca/discuss - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] -- ExchangeDefender Message Security: Click below to verify authenticity http://www.exchangedefender.com/verify.asp?id=l6DFoq6Q008060[EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Long-length FXS adapters?
Hello again, I am interested in setting up one of these Carrier Access Access Banks with a T1 card. Is there anywhere that I could go as a reference to purchasing and setting this up? I do see that there are some of these devices listed on eBay at this time. A solution for surge protection is something that would also be helpful. Does anyone have some information on this? Thanks, Steven On 7/11/07, Steven McCann [EMAIL PROTECTED] wrote: 6000ft ethernet extenders sound interesting. One solution we have been trying out was HPNA adapters which have range up to 2000ft on a pair of phone lines. I'd like to put internet access to the connection 1500ft away as well, so this is something I will consider. Having some solid telco equipment could also put my mind at ease for some of these connections. Under $500 sounds great for that many extensions. Hopefully I can learn how to config it properly though :-) Funny that you mention the grounding and protection also! The system was first put up in May for about 3 weeks, then a lightning storm came and short circuited the digium FXO card in the * server. It made the PSTN lines permanently busy and no outbound calls could be made as well. It wasn't very nice! I would be interested to know how some good grounding could be put in place. Right now I have just purchased some power bars with telephone inputs, but they aren't as convenient as they only provide protection for one line... I have also attached a rough diagram of the setup for your reference (shows the location of the HPNA adapters). FYI each location of the HPNA adapters also has a phone extension. The building with the HPNA adapter and the blue cat-5 cable coming into it is the spot where the HPNA adapters need to be able to reach for the current setup to work properly (has the internet access and link to to * server). Thanks for the great info! Steven On 7/10/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Saturday 07 July 2007 4:58 pm, Steven McCann wrote: Does anyone know the length you can run a FXS extension on copper cable pair (24 AWG)? I would personally recommend proper telco-grade hardware for anything like this. For FXS, a TE110P and a Carrier Access Access Bank I would be perfect, and reasonably priced on ebay. It's highly unlikely that you'll need any kind of echo cancellation on a length that short, and the ABI can drive long lines. You don't need to worry about disconnect supervision or anything on FXS ports, either, which is why you can get away with the older ABI and ABII instead of moving up to the Adit600. Adtran and Rhino make FXS channel banks as well, and Xorcom makes a USB channel bank as well, but I have not used any of these products. If you're running between buildings you may also need to worry about grounding and protection. I'd need to know more information to be able to tell you anything concrete, but yeah, for under $500 you can have yourself 24 ports of telco-grade FXS to play with. (channel banks and fax machines also get along great, if that is a concern.) -A. - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Polycom Echo Problems
Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you.
Re: [on-asterisk] Polycom Echo Problems
Intermittently, both have the echo. They have other other sets in the office not experiencing the echo problem. Their PSTN connection has been properly tuned for echo (via Milliwatt, etc). I'm just looking for a good config for AEC and AES on the Polys (and perhaps gains). By default they are turned off in the stock sip.cfg - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Bill Sandiford' [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, July 13, 2007 12:30 PM Subject: RE: [on-asterisk] Polycom Echo Problems Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
RE: [on-asterisk] Polycom Echo Problems
Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
[on-asterisk] Greetings
Greetings, I'm new to the Toronto Asterix User Group just wanted to take the opportunity to introduce myself and test my ability to receive mailing list e-mails. Sammy
RE: [on-asterisk] Greetings
Hi Sammy, Looks like you're working fine. Welcome to the group. Cheers, AR __ Alex Robar, Technical Support, GearyTech Inc. 3075 Fourteenth Avenue, Unit 3, Markham, Ontario L3R 0G9 Markham: 905-513-8000 x 223 Fax: 905-513-8040 Toronto: 416-226-3614 Toll Free: 888-890-3499 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.gearytech.com http://www.gearytech.com Strategic management of technology for business. From: Sacha Ally [mailto:[EMAIL PROTECTED] Sent: Friday, July 13, 2007 1:47 PM To: asterisk@uc.org Subject: [on-asterisk] Greetings Greetings, I'm new to the Toronto Asterix User Group just wanted to take the opportunity to introduce myself and test my ability to receive mailing list e-mails. Sammy -- ExchangeDefender Message Security: Click below to verify authenticity http://www.exchangedefender.com/verify.asp?id=l6DJv75f004382[EMAIL PROTECTED]
Re: [on-asterisk] Polycom Echo Problems
Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you that isn't a practical solution. The best you can do is put an echo canceler as close as you can to the endpoint and in this case it's on the Asterisk box. Unfortunately Asterisk's standard built in echo cancelers are crap. They don't even come close to reaching the level of the ITU G.164 standard for echo cancel. That is why you buy cards with add-on hardware echo cancelers that meet the G.164 standard (Sangoma, Digium). Recently you can also buy add-on software echo cancellation from both Sangoma Digium which meet the G.164 standard but beware it exacts a heavy toll on your CPU. But depending on call volume and hardware it might work just fine for you. All of this is a long winded way of saying; you can tune your phone settings until your blue in the face but you won't get rid of the echo. Sorry. So to prove my theory conduct the following tests: Polycom -- Polycom (no echo) Polycom -- Cell phone (no echo) (cell phones do extensive echo cancel) Polycom -- Longdistance (no echo) (telcos do echo cancel on LD) Polycom -- wireline residential (echo!!) That is why your customer reports intermittent echo problems. Hope the above helps you out. John On Fri, 2007-07-13 at 12:51 -0400, Bill Sandiford wrote: Intermittently, both have the echo. They have other other sets in the office not experiencing the echo problem. Their PSTN connection has been properly tuned for echo (via Milliwatt, etc). I'm just looking for a good config for AEC and AES on the Polys (and perhaps gains). By default they are turned off in the stock sip.cfg - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Bill Sandiford' [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, July 13, 2007 12:30 PM Subject: RE: [on-asterisk] Polycom Echo Problems Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED] - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional
[on-asterisk] Linksys spa 3106 to enable free local calling via mobile phone
Hello All, I have heard from various people that you could use the linksys spa 3106 in order to get free local calling via your cell phone. My understanding of how this is done is the following: 1-You subscribe to a cellular package such as the Fab 5 package or its carrier's equivalent. You add your home phone to your FAB 5 package which would allow both incoming and outgoing calls between your mobile phone and your home phone to be completely free. 2-You subscribe to a home phone carrier such as bell, [EMAIL PROTECTED], etc which you would plug into the linksys spa 3106's FXO port (line) 3-You would then subscribe to a VOIP provider for a monthly fee, and connect that to the router as well. 4-You then enable a DISA entry on your PBX ([EMAIL PROTECTED]/Trixbox). This would then allow you to dial in via the PSTN line from your mobile phone (which would be free due to the FAB 5 plan) and then the DISA entry could provide you with a dial tone via the VOIP service you subscribed to, that you could dial out from. Sorry if this is very vague but I'm just trying to get the main idea right in order to assess whether this is the right option for me. Im sure configuration is a much more involved process. But is my understanding of this correct? The only concern I have, if this is correct, is that I used to have a Vonage line and I found it sucked royally! Dropped calls, echos, delayed audio, anything you could imagine went wrong with it. Based on this experience I have hesitations about using a VOIP line to route my cell phone calls through. Also it doesn't seem like your really saving that much when you think about it. I currently don't have a home phone line (PSTN) so I would have to purchase that and I also don't currently utilize a VOIP service so I would have to purchase that as well. The two of those combined I would image would be atleast approx. $50/month on top of the regular FAB 5 cell phone package. Is this really that economical for a home? Is it based on the premise that home users are already paying for a LAN line? I could see it being very cost effective if you could route from the PSTN back through the PSTN with a single home phone line, but I would guess that this is not a feasible option. Is it? Perhaps I'm misunderstanding this,.. any thoughts?
Re: [on-asterisk] Polycom Echo Problems
John: Thanks for the description, it is very good. Here is the scenario in which this customer is experiencing echo. Polycom --- Asterisk --- Internet --- Polycom also Polycom --- Asterisk --- IX Private Line (delay 10ms) --- Polycom also Polycom --- Asterisk SIP Trunk to carrier --- Carrier's CLASS 5 --- PSTN This particular customer is seeing echo in all three scenarios. Hence the reason I'm looking into the AEC and AES features of the Polycom. Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Cc: [EMAIL PROTECTED] Sent: Friday, July 13, 2007 4:38 PM Subject: Re: [on-asterisk] Polycom Echo Problems Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you that isn't a practical solution. The best you can do is put an echo canceler as close as you can to the endpoint and in this case it's on the Asterisk box. Unfortunately Asterisk's standard built in echo cancelers are crap. They don't even come close to reaching the level of the ITU G.164 standard for echo cancel. That is why you buy cards with add-on hardware echo cancelers that meet the G.164 standard (Sangoma, Digium). Recently you can also buy add-on software echo cancellation from both Sangoma Digium which meet the G.164 standard but beware it exacts a heavy toll on your CPU. But depending on call volume and hardware it might work just fine for you. All of this is a long winded way of saying; you can tune your phone settings until your blue in the face but you won't get rid of the echo. Sorry. So to prove my theory conduct the following tests: Polycom -- Polycom (no echo) Polycom -- Cell phone (no echo) (cell phones do extensive echo cancel) Polycom -- Longdistance (no echo) (telcos do echo cancel on LD) Polycom -- wireline residential (echo!!) That is why your customer reports intermittent echo problems. Hope the above helps you out. John On Fri, 2007-07-13 at 12:51 -0400, Bill Sandiford wrote: Intermittently, both have the echo. They have other other sets in the office not experiencing the echo problem. Their PSTN connection has been properly tuned for echo (via Milliwatt, etc). I'm just looking for a good config for AEC and AES on the Polys (and perhaps gains). By default they are turned off in the stock sip.cfg - Original Message - From: Jim Van Meggelen [EMAIL PROTECTED] To: 'Bill Sandiford' [EMAIL PROTECTED]; asterisk@uc.org Sent: Friday, July 13, 2007 12:30 PM Subject: RE: [on-asterisk] Polycom Echo Problems Couple of things that need to be known: Who has the echo? Your users? or the people who are calling them? How does the system connect to the outside world? (PSTN) Jim -Original Message- From: Bill Sandiford [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 12:23 PM To: asterisk@uc.org Subject: [on-asterisk] Polycom Echo Problems Hi All: I'm having a problem with a customer that has a bunch of Polcom 501 and 601 sets. They are complaining about echo. Does anyone have some suggestions for some good settings for AEC and AES in sip.cfg for the Polys? Any other suggested settings or changes to the stock sip.cfg? Thanks, Bill Sandiford Telnet Communications 905-674-2000 x100 [EMAIL PROTECTED] IMPORTANT NOTICE: This message is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and exempt from disclosure under applicable law. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this communication in error, please notify the sender immediately by email and delete the message. Thank you. No virus found in this incoming
Re: [on-asterisk] Polycom Echo Problems
In the first two scenarios you describe, you are essentially doing pure SIP to SIP using the Polycoms and that should not cause echo unless echo cancel is disabled on the far end handset. I just now re-read your original posting and indeed you have AEC turned off so that is definitely your problem. One thing to try is to ask the far end to put the phone on mute and see if the echo goes away. If it does, then your echo is being caused by acoustic echo, not an impedance miss-match or other network problem. If you narrow it down to acoustic echo (which is actually the only possibility) then the responsibility of eliminating that echo is squarely with the handset (Polycom) and you'll have to try tuning the related settings. We don't use Polycoms but their conference phones have a reputation for very good echo cancel so I'd be surprised if their handsets weren't equally as good. Mind you I just had a look at their spec sheets and they don't claim G.164 so maybe they don't? If you mute the far end and you still get echo then something else very strange is going on. Like your Asterisk is actually looping the call through the PRI or its traversing an analog circuit or some other thing that shouldn't be happening. In the final scenario (SIP Trunk to carrier), you can't do anything since you don't control the Carrier - PSTN where the echo cancel needs to happen. Your carrier should have their own echo cancel so If they are running asterisk with PRI interface cards that don't have echo cancel they you should consider changing carriers. Also have a look at this which gives a pretty good explanation of what causes echo. http://en.wikipedia.org/wiki/Echo_cancellation But bottom line; turn on AEC on the handsets and the problem will go away. Regards, John On Fri, 2007-07-13 at 17:24 -0400, Bill Sandiford wrote: John: Thanks for the description, it is very good. Here is the scenario in which this customer is experiencing echo. Polycom --- Asterisk --- Internet --- Polycom also Polycom --- Asterisk --- IX Private Line (delay 10ms) --- Polycom also Polycom --- Asterisk SIP Trunk to carrier --- Carrier's CLASS 5 --- PSTN This particular customer is seeing echo in all three scenarios. Hence the reason I'm looking into the AEC and AES features of the Polycom. Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Cc: [EMAIL PROTECTED] Sent: Friday, July 13, 2007 4:38 PM Subject: Re: [on-asterisk] Polycom Echo Problems Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you that isn't a practical solution. The best you can do is put an echo canceler as close as you can to the endpoint and in this case it's on the Asterisk box. Unfortunately Asterisk's standard built in echo cancelers are crap. They don't even come close to reaching the level of the ITU G.164 standard for echo cancel. That is why you buy cards with add-on hardware echo cancelers that meet the G.164 standard (Sangoma, Digium). Recently you can also buy add-on software echo cancellation from both Sangoma Digium which meet the G.164 standard but beware it exacts a heavy toll on your CPU. But depending on call volume and hardware it might work just fine for you. All of this is a long winded way of saying; you can tune your phone settings until your blue in the face but you won't get rid of the echo. Sorry. So to prove my theory conduct the following tests: Polycom -- Polycom (no echo) Polycom -- Cell phone (no echo) (cell phones do extensive echo cancel) Polycom -- Longdistance (no echo) (telcos do echo cancel on LD) Polycom -- wireline residential (echo!!) That is why your customer reports intermittent echo problems.
RE: [on-asterisk] Greetings
Welcome to the herd. Jim -- Jim Van Meggelen [EMAIL PROTECTED] http://www.oreillynet.com/pub/au/2177 A child is the ultimate startup, and I have three. This makes me rich. Guy Kawasaki -- -Original Message- From: Sacha Ally [mailto:[EMAIL PROTECTED] Sent: July 13, 2007 1:47 PM To: asterisk@uc.org Subject: [on-asterisk] Greetings Greetings, I’m new to the Toronto Asterix User Group just wanted to take the opportunity to introduce myself and test my ability to receive mailing list e-mails. Sammy No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.4/898 - Release Date: 12/07/2007 4:08 PM - To unsubscribe, e-mail: [EMAIL PROTECTED] For additional commands, e-mail: [EMAIL PROTECTED]
Re: [on-asterisk] Polycom Echo Problems
John: Once again, thank you. Your response was again very detailed and helpful. Since my last post (~30 minutes ago), I was able to solve the problems with the third scenario (PSTN). Turned out the carrier had the echo cans turned off for this client on their SIP trunking. Carrier turned on the echo cans and the echo was gone. However, there is still echo in the first 2 scenarios which were both all SIP/VoIP (no PSTN). Its essentially this: Polycom --- Asterisk --- Polycom I tried your suggestion of having the far end mute and you are correct. I am on site and when I place a call I have near-end echo (I hear myself). When the far side mutes, the echo is gone. When the far side unmutes the echo is back. I have turned on the AEC and AES in the Polycom sip.cfg but it hasn't had much of an impact. There are some other settings to do with AEC and AES in the file and hence I am still looking for some recommended settings from anyone that has used them. Regards, Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: Bill Sandiford [EMAIL PROTECTED] Cc: asterisk@uc.org Sent: Friday, July 13, 2007 5:55 PM Subject: Re: [on-asterisk] Polycom Echo Problems In the first two scenarios you describe, you are essentially doing pure SIP to SIP using the Polycoms and that should not cause echo unless echo cancel is disabled on the far end handset. I just now re-read your original posting and indeed you have AEC turned off so that is definitely your problem. One thing to try is to ask the far end to put the phone on mute and see if the echo goes away. If it does, then your echo is being caused by acoustic echo, not an impedance miss-match or other network problem. If you narrow it down to acoustic echo (which is actually the only possibility) then the responsibility of eliminating that echo is squarely with the handset (Polycom) and you'll have to try tuning the related settings. We don't use Polycoms but their conference phones have a reputation for very good echo cancel so I'd be surprised if their handsets weren't equally as good. Mind you I just had a look at their spec sheets and they don't claim G.164 so maybe they don't? If you mute the far end and you still get echo then something else very strange is going on. Like your Asterisk is actually looping the call through the PRI or its traversing an analog circuit or some other thing that shouldn't be happening. In the final scenario (SIP Trunk to carrier), you can't do anything since you don't control the Carrier - PSTN where the echo cancel needs to happen. Your carrier should have their own echo cancel so If they are running asterisk with PRI interface cards that don't have echo cancel they you should consider changing carriers. Also have a look at this which gives a pretty good explanation of what causes echo. http://en.wikipedia.org/wiki/Echo_cancellation But bottom line; turn on AEC on the handsets and the problem will go away. Regards, John On Fri, 2007-07-13 at 17:24 -0400, Bill Sandiford wrote: John: Thanks for the description, it is very good. Here is the scenario in which this customer is experiencing echo. Polycom --- Asterisk --- Internet --- Polycom also Polycom --- Asterisk --- IX Private Line (delay 10ms) --- Polycom also Polycom --- Asterisk SIP Trunk to carrier --- Carrier's CLASS 5 --- PSTN This particular customer is seeing echo in all three scenarios. Hence the reason I'm looking into the AEC and AES features of the Polycom. Bill - Original Message - From: John Lange [EMAIL PROTECTED] To: asterisk@uc.org Cc: [EMAIL PROTECTED] Sent: Friday, July 13, 2007 4:38 PM Subject: Re: [on-asterisk] Polycom Echo Problems Echo *always* comes from the far end point. The amount a given person perceives the echo is determined by how loud and how delayed the echo is. Volume and delay are influenced by a number of factors along the call path. Echo is a very complex issue but I'll try and give a brief explanation. In the situation where you have a Polycom phone connected to an Asterisk server which is in turn connected to the PSTN talking to a residential wireline customer, e.g.: Polycom -- Asterisk -- PRI -- Wireline Handset If the Polycom customer hears echo it's coming from the wireline handset (and/or the hybrid but I'm trying to keep this example simple). Most consumer handsets just don't care about generating echo because its never been a problem. So echo is normal on all local wireline calls but you don't perceive (hear) echo because the echo is not delayed. Now when you throw Asterisk in the mix the act of encoding and decoding the voice adds delay. This added delay causes you to perceive echo even though the volume of the echo is roughly the same. Technically, to solve echo you fix the endpoint that's causing the echo. But since you can't replace every wireline phone ever made and the telco certainly isn't going to help you that
Re: [on-asterisk] Greetings
Welcome! Always glad to have another Sacha around :) Sent from my BlackBerry device on the Rogers Wireless Network -Original Message- From: Sacha Ally [EMAIL PROTECTED] Date: Fri, 13 Jul 2007 13:46:48 To:asterisk@uc.org Subject: [on-asterisk] Greetings Greetings, I’m new to the Toronto Asterix User Group just wanted to take the opportunity to introduce myself and test my ability to receive mailing list e-mails. Sammy
