Re: [Asterisk-Users] Optimum online-upload throttling confirmed.
On Thu, 2005-08-18 at 21:08, [EMAIL PROTECTED] wrote: Hello All, I was recently fighting with an optimum online connection in NY. I finally got in touch with someone that confirmed they are throttling my upload connection. I know they watch for people doing peer to peer file sharing and throttle those connections quite severely, but I wasn't aware that they do general throttling. I just wanted to make everyone aware of it, so if you have problems if your ping times jump erratically, this could be the cause. Their suggestions were, although you can upload a lot, do not do it constantly. They do not want any constant outgoing connections. Even on business class, they do throttle. All business class primarily does is allow port 25 to pass. Now I am going to look and see if I can get a decent upload speed dsl or something to correct this problem. I have a friend who uses Vonage (on a ComCast cable modem, not Cablevision) and many times when I talk to him, the voice quality is bad. The reason is the way that _all_ cable companies deploy their data services (it's a CableLabs DOCSIS standard that they all use). Remember that cable modem networks are shared media. The downstream to the cable modem is a broadcast and each cable modem listens for traffic to it. No latency problems here. However, on the upstream, each cable modem requests permission to send and then the cable modem termination system (CMTS) grants it a token to send. Very significant latency and jitter problems here for VoIP. For their own service offerings, the cable companies solve the problem by identifying the VoIP call flows during the call setup and scheduling the media packet stream (RTP packets) grants in advance. Therefore, when I use my Cablevision optimum voice line, my RTP packets are given special priority and the latency and jitter problem is solved. But if you are using Vonage or your own Asterisk box, your RTP packets are treated the same as any other data packet. Not good for VoIP quality. And the problem becomes very bad when the network gets busy. The bottom line is that I wouldn't try to use any cable modem service (ComCast, Cox, Time Warner, Cablevision, doesn't matter) for a VoIP service where voice quality really matters a lot. Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitoring RTP protocol
don't know if Asterisk can do it, but ethereal can. Ethereal is an open source protocol analyzer. Download it from www.etheral.com On Fri, 2005-08-19 at 02:55, Bohuslav Coufal wrote: Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Thanks for answer, Bob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Distribution for Asterisk server use
I needed to learn Linux for a project about 18 months ago, so I went down to a retail computer store and bought RedHat Linux. Installed with no problems, and I was up and playing around with it in about an hour. As I got more into it (and started breaking things) I needed some help. The help I got from RedHat tech support was not very helpful. It was the e-mail help you get with the retail consumer version, so I figured you get what you pay for and let it go. If you had a business support contract, it might be better, but maybe not. The best help I got was from joining my local Linux Users Group (LILUG in my case). They were great. And when I went to a few meetings, and got to ask questions in person it got really good! If you have a local Linux User's group that's not too inconvenient to attend I highly recommend it. Anyway, at the user's group meetings their existed a friendly rivalry between the RedHat crowd and the Debian crowd, so I decided to try Debian. Couldn't get it to install. I really tried, even got some help from the Debian guys on the list, but I just couldn't do it. Now, I'm not a Linux guru, but I can follow instructions, but I just couldn't get it to go. Then I downloaded and installed Fedora Core 1 (RedHat open source / development version). No problem. So my newbie experience is that RedHat is quite a bit easier to install. Used to be that one of the big advantages of Debian was its package management system (apt). RedHat has a good package manager now too (yum). So IMHO, go with RedHat for the following reasons: 1. Sounds like price isn't your big issue, so if you purchase an enterprise edition of Linux, you'll have access to RedHat tech support, and you'll have a certain amount of CYA built in. 2. Some might argue that the community support for Debian or Mandrake is better, but the mailing list / IRC support you'll get with RedHat is probably good enough. 3. There are more books available for RedHat than for other distributions. 4. In my experience, it installs easier. 5. Getting security patches and OS upgrades from RedHat is very simple (probably is with the other distros too). For what you're going to do with Asterisk, I don't think there are huge technical differences between the distributions, so the main consideration ought to be which one can I install and learn the fastest and not which one will support the most clients, or have the most uptime. Having said that, there is one caveat - I would stay away from Fedora Core 3 or Debian unstable or whatever newest release of any version. Also keep in mind that Asterisk runs just fine on Linux kernel 2.4.x. You don't need 2.6.x. Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk documentation
Look for a book coming out from O'Reilly in a few months or so. It's being offered on Amazon (pre-release). I can't remember the names of the authors, but I'm pretty sure they are some of the Asterisk developers. Jeff On Wed, 2005-06-29 at 19:37, Robert Goodyear wrote: On Jun 29, 2005, at 3:40 PM, harry gaillac wrote: Hello, If asterisk.org can't provide you documentations have a look here : http://www.digium.com/index.php? menu=product_detailcategory=softwareproduct=ABE I do hope some people understand my posts. Regards Harry Yeah, loud and clear. By the way, ever heard of a company called RedHat? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS HEAD won't compile for me
I checked out CVS HEAD today and tried to compile it with no luck, so then I checked out the stable version and compiled it successfully. I'm 99% sure that I'm not missing anything and that I'm following the instructions correctly (I'm no guru, but I've compiled lots of programs successfully). My question is this: is it fairly common that the CVS HEAD version won't compile? Before I start digging deeper and troubleshooting, I want to make sure that it wasn't just something broken in the program and it didn't compile because it shouldn't compile. TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does Asterisk Realtime require the use of CVS HEAD ???
I read on the Wiki that Asterisk Realtime requires CVS HEAD, but I've also discovered that not everything on the Wiki is 100% accurate (that's not a knock, but with a program that is changing as fast as Asterisk, it's impossible for the documentation to keep up). Is it true that Realitme requires CVS HEAD? TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot create a personalized unavailable message
When I try to create a personalized unavailable message the VoicemailMain application says please record your message beeps and then goes straight to if you want to accept... I checked the output of /var/log/asterisk/messages and see the following: WARNING: Unable to open file /var/lib/asterisk/sounds/voicemail/default/4035/unavail.WAV: No such file or directory. and sure enough when I check the directory tree, the directory /var/lib/asterisk/sounds/voicemail doesn't exist Is there something I forgot to configure? I would think that these directories would be created automatically. I'm running Asterisk version CVS-v1-0-02/17/05-17:34:40 TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Passwords
yep, I think you're right that the voicemail.conf file is being
dynamically rebuilt. The reason that was not being reflected before is
that I had the voicemail.conf file open and therefore asterisk could not
write to it. However, I noticed that when I closed it and re-opened it,
that the changes to the password were reflected just as you surmised.
So that solves my question below.
Thanks!
Jeff Heath
On Wed, 2005-05-11 at 19:33, BJ Weschke wrote:
Looking at app_voicemail.c with the copy I have here, it looks like
vm_change_password is trying to dynamically rebuild the voicemail.conf
file. It writes a voicemail.conf.new file, and then replaces one with
the other once it's done.
What version of asterisk are you running?
Do you get an WARNINGs or any other kind of logging info when you
reset the password? Looking at the code, it's supposed to issue
warnings if it cannot open the old file for read and/or open the new
file for write.
On 5/11/05, BJ Weschke [EMAIL PROTECTED] wrote:
I see what you're saying. Unless someone else responds with the
issue, I'll look at the code a little later this evening. It sounds
like the changed password via IVR is going into the ast-db, and then
that new value is ignoring what's in voicemail.conf.
On 5/11/05, Jeff Heath [EMAIL PROTECTED] wrote:
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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Re: [Asterisk-Users] Voicemail Passwords
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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Re: [Asterisk-Users] Voicemail Passwords
On Tue, 2005-05-10 at 21:25, BJ Weschke wrote:
voicemail.conf
edit that file and issue a reload to change them.
I tried this, but I still can't get access to voicemail from one of the
phones.
This is a test system that I setup about a month ago. Got busy and am
just now getting back to it. I have 2 SIP phones and the Asterisk
server. The default voicemail password is 1234 for both extensions. I
changed the password for one of them and (doh!) forgot/lost it.
Since this is a test system, I tried an experiment. I went into the
phone where I can get access to voicemail, and I manually changed the
password from 1234 to 4567. Then I issued a reload (the default
passwords in voicemail.conf are 1234). Then I accessed voicemail again,
and the password is 4567 not 1234.
This makes sense to me. Otherwise, every time asterisk was restarted or
reloaded all the user's personal voicemail passwords would be reset.
Surely, I'm not the first dope that's changed a password and forgot it
:-)
I can't believe there's not a file somewhere that the administrator can
directly edit to change user voicemail passwords, but I've been
searching the Wiki and googling on lists.digium.com and searched all the
Asterisk documentation I can find and I can't find it.
So, how does the administrator reset a user's password?
fyi, here are my extensions.conf and voicemail.conf
extensions.conf
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail(u4035)
exten = 4035,102,Voicemail(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail(u4009)
exten = 4009,102,Voicemail(b4009)
exten = 4009,103,Hangup
; This defines the number to access VM.
; The caller's extension number is passed as a variable, so
; all the user needs to do is type in the password.
exten = 4040,1,VoicemailMain(${CALLERIDNUM})
[local]
include = from-sip
voicemail.conf
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[default]
4009 = 1234,Jeff
4035 = 1234,Pam
On 5/10/05, Jeff Heath [EMAIL PROTECTED] wrote:
Where are user's voicemail passwords stored and how does the asterisk
administrator change them?
TIA,
Jeff Heath
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[Asterisk-Users] Voicemail Passwords
Where are user's voicemail passwords stored and how does the asterisk administrator change them? TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Personal Communications Assistant
Is there a GUI for Asterisk that has similar capabilities to the Cisco Personal Communications Assistant. I looked at the user interfaces on the Wiki and they all seem to fall a little short. Is anyone on the list using something they really like? TIA, Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any useful results?
On Tue, 2005-05-03 at 14:15, Christopher Jacob wrote: Josiah Bryan, Any useful results from your number of installed systems survey? If so, could you email them to me off-list? actually, could you e-mail them on-list (might be more appropriate for Asterisk-Biz though). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Citel Handset Gateways
Does anyone have any direct experience with these? What do they cost per port? Do they support most of the features of the original phone (i.e. if I have a Meridian phone, do all the buttons like conference, flash, hold, etc. work the same) ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Business Case - Who is using it!?
you might want to also post this to asterisk-biz. On Tue, 2005-04-19 at 14:01, Denis Galvão - iSolve wrote: Hi all. Im participating of a project(a huge one) that will study Asterisk as its PABX base system. They ask me: Who is using Asterisk as its base PABX!? Now I ask you: Anyone know about some important and big company that have been implemented Asterisk!? Im not talking about VoIP providers... Maybe this question will be the point of a decision to this project. Thanks a lot! Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Echo
On Tue, 2005-04-12 at 15:28, Noah Silverman wrote: Hi, I tried, and still get an echo. I don't think the problem is with the zap interface. It must be on the asterisk or phone side. -N Echo requires 2 phenomena: 1) reflected energy 2) enough delay that it is discernable. That you are hearing echo means that something at the far end is reflecting the electrical or accoustical energy of your voice. Echo cancellation should be done as close to the source of unwanted reflected energy as possible. The fact that you're hearing echo means that the echo cancelers at the far end either a) don't exist or b) didn't work. It will be very difficult to cancel reflected energy coming back at you from the other side of the network. Tell me more about the phone call where you experienced the echo and I _might_ be able to help. Specifically, - was the phone at the other end a speaker phone and if so, was it an expensive Polycom phone that's designed to be a speaker phone or a cheap Walmart phone that happens to have speaker capability? - was it a local call or a long distance call - what codecs are in use? - what's your best guess at the round trip delay (i.e. what networks had to be traversed and what is the jitter buffer set for?) Rich Adamson wrote: I have a strange echo problem. When speaking on the phone with someone, I hear MY OWN voice with a sever echo. The other party sounds perfect, and they can hear me perfectly. It is as if only the sidetone has an echo. I'm running * on a dedicated box, small LAN, and am using 4 FXO cards to connect the box to PTSN lines. My phones are Polycom IP500 SIP phones. The only echo cancellation stuff that I can find relates to cancelling echo between my system and the PTSN lines. Since the call is perfect, I don't see how this would apply. Any suggestions?? Try these parameters for each zap channel: echotraining=800 echocancel=yes echocancelwhenbridged=yes Don't forget you have to stop and restart asterisk. a reload will not work. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How do I reduce echo on asterisk
On Tue, 2005-04-12 at 15:54, Joel Jn-Francois wrote: On April 12, 2005 11:59 am, Joel Jn-Francois wrote: I get an echo only from the caller end when I am making calls. I only get it for some VOIP providers. I am using asterisk Asterisk CVS-v1-0-03/26/05-16:54:47 and Grandstream HandyTone 486 and 488. My default codec is ulaw. Is there any way I can reduce the echo without comprising quality? Your terminology is confusing. When you place a call through your handytones, do you hear echo, or does the other side hear echo? Sorry... When I make a call I hear an echo, but the person I am speaking to on the other side does not. Joel There's probably not much you can do about it. Echo cancelation should occur as close to the source of reflected energy as possible. Here's one scenario that would explain your experience. Cheapskate VoIP Inc. connects your call at the far end to the local PSTN. Relected energy is created at the hybrid in the local PSTN between the 4 wire T-1 interface and their 2 wire local subscriber loop. The local telephone companies do not cancel this echo because when they connect local calls, the reflected energy is not delayed in time enough for humans to perceive it as echo. Good inter-exchange carriers install echo cancelers near this interface and cancel it because their customers probably will experience the reflected energy as echo. Back to our scenario... Cheapskate VoIP Inc. doesn't want to pony up for echo cancelers so you get echo. Now comes the hard part (and someone please jump in here because I don't know the details of how echo cancellation has been implemented in Asterisk). In addition, Cheapskate VoIP Inc. compresses the hell out of their traffic. So now you've started out as G.711 then Cheapskate compressed it to G.729 and uncompressed it at the far end then compressed the returned signal (which contained your reflected energy). Now the echo canceler in Asterisk must compare the original signal that was encoded G.711 with a returned signal that has been encoded twice in G.729 (once on the outbound side and then again on the inbound side). That alone would severely diminish the performance of an echo canceler, but just to make it a little harder we add in a bunch of delay and then just for good measure we adjust the jitter buffer every once in a while to add some variability to the delay. The echo canceler is doomed. If you can figure out how to make an echo canceler work under these conditions e-mail me off list and we'll put together a business plan to make a lot of money (seriously, if you can figure this out e-mail me at [EMAIL PROTECTED]). Realistically, I doubt there is much you can do except try to get a different VoIP carrier. - Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Local Echo
Here's what's happening. First some background. Anytime there's a 4 wire (T-1) to 2 wire (local subscriber loop) conversion (this is called a hybrid) there's a good chance that some electrical energy will be reflected. This is because there is usually an impedance mismatch between the 4 wire and 2 wire circuits. This happens all the time in the local telco. You come in to switch A and are destined for switch Z. The telco transports the traffic between A and Z over T-1 (which is muxed up to T-3 or SONET). When the T-1 gets to switch Z it eventually gets attached to a 2 wire local loop (POTS) to get to the far end. Energy from A is reflected back towards A by the hybrid at the Z side. But reflected energy is only one of two necessary conditions for echo. The other condition is sufficient delay for a human being to perceive it as echo. In order for us to perceive it as echo, the reflected energy must be delayed by about 25 msec. Anything less than that and we perceive it as sidetone (sidetone is actually a good thing). The local telephone company doesn't have echo cancelers in their network because they don't need them. Why? because in the local POTS network you'll never have a call that is delayed by more than 25 msec. Long distance carriers (IXCs) install echo cancelers because their customers will experience delays longer than 25 msec, but not local telcos. Now introduce VoIP. VoIP turns every call (even the simple setup you outlined) into a long distance call. If you have your jitter buffer set to 3 you've introduced 60 msec of delay. I forget the rule of thumb for distance vs electrical delay, but I think you can go from NY to SanDiego in about 85 msec. That explains why the echo is there. What I can't help you with (I've got lots of telecom experience, but little Asterisk experience) is changing the settings in Asterisk to cancel it. The good news, though, is that this is a straight-forward echo cancellation problem, and once you find someone who knows what the right settings are, you should be able to get rid of it. -- Jeff Heath On Tue, 2005-04-12 at 17:28, Noah Silverman wrote: Jeff, Thanks for the help. Your explanation of an echo makes perfect sense. Here are some notes on our system that might help: 1) The echo occurs on EVERY call either inbound or outbound, local or ld. 2) We don't use any VOIP services, just PTSN lines from the phone company 3) Our system is like this: SIP phone - Asterisk box - TDM400 card with FXO - Telco Pots line 4) I hear my own voice echo. The other party sounds fine to me, and I sound fine to them. 5) The phones are on a very small LAN in our office with almost no traffic. 6) Our phones are Polycom IP500 7) I have the codec set to ulaw Thanks!!! -N Jeff Heath wrote: On Tue, 2005-04-12 at 15:28, Noah Silverman wrote: Hi, I tried, and still get an echo. I don't think the problem is with the zap interface. It must be on the asterisk or phone side. -N Echo requires 2 phenomena: 1) reflected energy 2) enough delay that it is discernable. That you are hearing echo means that something at the far end is reflecting the electrical or accoustical energy of your voice. Echo cancellation should be done as close to the source of unwanted reflected energy as possible. The fact that you're hearing echo means that the echo cancelers at the far end either a) don't exist or b) didn't work. It will be very difficult to cancel reflected energy coming back at you from the other side of the network. Tell me more about the phone call where you experienced the echo and I _might_ be able to help. Specifically, - was the phone at the other end a speaker phone and if so, was it an expensive Polycom phone that's designed to be a speaker phone or a cheap Walmart phone that happens to have speaker capability? - was it a local call or a long distance call - what codecs are in use? - what's your best guess at the round trip delay (i.e. what networks had to be traversed and what is the jitter buffer set for?) Rich Adamson wrote: I have a strange echo problem. When speaking on the phone with someone, I hear MY OWN voice with a sever echo. The other party sounds perfect, and they can hear me perfectly. It is as if only the sidetone has an echo. I'm running * on a dedicated box, small LAN, and am using 4 FXO cards to connect the box to PTSN lines. My phones are Polycom IP500 SIP phones. The only echo cancellation stuff that I can find relates to cancelling echo between my system and the PTSN lines. Since the call is perfect, I don't see how this would apply. Any suggestions?? Try these parameters for each zap channel: echotraining=800 echocancel=yes echocancelwhenbridged=yes Don't forget you have to stop and restart asterisk. a reload will not work. ___ Asterisk
Re: [Asterisk-Users] Local Echo
Sidetone is, by definition, echo. But it's good echo. What do I mean by that? It means that sidetone gives you some feedback - you hear yourself talk. bad echo is when the delay gets too long. If the echo is annoying you, it's not sidetone. The listener at the other end of the conversation doesn't hear an echo because (I'm assuming they're on a POTS line) the delay in his echo path is only about 10 msec. Also keep in mind that if the far end heard an echo, he would hear an echo of his own voice, not an echo of your voice. Also, the other responder is correct to check your gain settings. If the reflected signal is coming in too hot (most echo cancelers set the threshohld at -6dB) the echo canceler assumes it's the other end talking and disables itself. I re-read your post and the only thing that doesn't add up is the fact that you're getting echo on all your long distance calls too. Getting echo on all local calls fits my previous explanation, but not ld. The reason is that the ld companies do a good job of canceling echo at the source (otherwise we would hear it much of the time on the PSTN). Also keep in mind that you do use VoIP services - between the Asterisk box and the SIP phones. That's where the delay is coming from. You're not going to have significant jitter or delay problems on your local network, so adjust your jitter buffer down to 1. It will make your calls better from a latency point of view and it might help with the echo too. -- Jeff Heath On Tue, 2005-04-12 at 19:19, Noah Silverman wrote: Hi, I think that you guys are missing the problem. The echo is only from the sidetone. I don't hear the other party with an echo and they don't hear me with an echo. That leads me to believe that it hs nothing to do with the zapata stuff. It is somewhere between my SIP phone as Asterisk. -N Rod Bacon wrote: In addition to making sure that echo cancellation is enabled on the interface(s) in question, you will also need to play with the gain settings. Specifically, try turning down the rxgain. I dropped mine to -10.0, and the echo disappeared altogether. The problem was then that incoming voice was too quiet. After a lot of messing around, I eventually settled on -3.0 This figure gives me good incoming volume and only a faint echo... not enough to bother me or my users. I also found that the order of settings in the zapata.conf makes a difference. If I had the gain settings too far down in the config file, they had no effect. Make sure you stop and restart * after changing any of these settings. A simple reload won't suffice (I even unloaded and reloaded the kernel modules, just to be sure). - Original Message - From: Jeff Heath [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 13, 2005 7:54 AM Subject: Re: [Asterisk-Users] Local Echo Here's what's happening. First some background. Anytime there's a 4 wire (T-1) to 2 wire (local subscriber loop) conversion (this is called a hybrid) there's a good chance that some electrical energy will be reflected. This is because there is usually an impedance mismatch between the 4 wire and 2 wire circuits. This happens all the time in the local telco. You come in to switch A and are destined for switch Z. The telco transports the traffic between A and Z over T-1 (which is muxed up to T-3 or SONET). When the T-1 gets to switch Z it eventually gets attached to a 2 wire local loop (POTS) to get to the far end. Energy from A is reflected back towards A by the hybrid at the Z side. But reflected energy is only one of two necessary conditions for echo. The other condition is sufficient delay for a human being to perceive it as echo. In order for us to perceive it as echo, the reflected energy must be delayed by about 25 msec. Anything less than that and we perceive it as sidetone (sidetone is actually a good thing). The local telephone company doesn't have echo cancelers in their network because they don't need them. Why? because in the local POTS network you'll never have a call that is delayed by more than 25 msec. Long distance carriers (IXCs) install echo cancelers because their customers will experience delays longer than 25 msec, but not local telcos. Now introduce VoIP. VoIP turns every call (even the simple setup you outlined) into a long distance call. If you have your jitter buffer set to 3 you've introduced 60 msec of delay. I forget the rule of thumb for distance vs electrical delay, but I think you can go from NY to SanDiego in about 85 msec. That explains why the echo is there. What I can't help you with (I've got lots of telecom experience, but little Asterisk experience) is changing the settings in Asterisk to cancel it. The good news, though, is that this is a straight-forward echo
[Asterisk-Users] Cannot access voicemail
I'm having trouble checking voicemail. When I make a call and the
recipient doesn't answer, the call goes to voicemail, and it's being
recorded (I checked the files in
/var/spool/asterisk/voicemail/from-sip/4035/INBOX).
My problem is that I can't get access to the recorded message. I dial
the extension I setup to go to voicemail (4040) and then the voicemail
system asks for a password. I press 1234 (which is what I *think* I
setup in voicemail.conf), but I get a message that the password is
incorrect.
I've included my config files below.
this is the output on the Asterisk CLI:
*CLI -- Executing VoiceMailMain2(SIP/4035-256b, 4035) in new
stack
-- Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '4035' (context = any)
-- Playing 'vm-incorrect'Untitled 1 (language 'en')
-- Playing 'vm-password' (language 'en')
Apr 8 12:12:24 WARNING[22786]: app_voicemail.c:3389 vm_execmain: Unable
to read password
== Spawn extension (from-sip, 4040, 1) exited non-zero on
'SIP/4035-256b'
--- extensions.conf ---
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail2(u4035)
exten = 4035,102,Voicemail2(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail2(u4009)
exten = 4009,102,Voicemail2(b4009)
exten = 4009,103,Hangup
exten = 4040,1,VoicemailMain2(${CALLERIDNUM})
[local]
include = from-sip
--- voicemail.conf ---
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[from-sip]
4009 = 1234,Jeff
4035 = 1234,Pam
sip.conf ---
[general]
port = 5060
[4035]
type = friend
username = 4035
secret = pamela
context = from-sip
callerid = Pam 4035
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4035
defaultip = 192.168.1.104
[4009]
type = friend
username = 4009
secret = jeff
context = from-sip
callerid = Jeff 4009
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4009
defaultip = 192.168.1.105
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Re: [Asterisk-Users] Cannot access voicemail
I think maybe it has something to do with tones from my phone not being
recognized. I just called an IAX number to make a test call and when
the IVR asked me to press a number it didn't take. That would be
consistent the CLI msg that says Unable to read password. It's also
consistent with my experience that it takes about 5 seconds for Asterisk
to come back to me with the incorrect password response.
So is there a setting I need to make on the phone or in one of the
config files to get asterisk to recognize DTMF digits or something?
Thanks,
Jeff
On Fri, 2005-04-08 at 12:48, Jeff Heath wrote:
I'm having trouble checking voicemail. When I make a call and the
recipient doesn't answer, the call goes to voicemail, and it's being
recorded (I checked the files in
/var/spool/asterisk/voicemail/from-sip/4035/INBOX).
My problem is that I can't get access to the recorded message. I dial
the extension I setup to go to voicemail (4040) and then the voicemail
system asks for a password. I press 1234 (which is what I *think* I
setup in voicemail.conf), but I get a message that the password is
incorrect.
I've included my config files below.
this is the output on the Asterisk CLI:
*CLI -- Executing VoiceMailMain2(SIP/4035-256b, 4035) in new
stack
-- Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '4035' (context = any)
-- Playing 'vm-incorrect'Untitled 1 (language 'en')
-- Playing 'vm-password' (language 'en')
Apr 8 12:12:24 WARNING[22786]: app_voicemail.c:3389 vm_execmain: Unable
to read password
== Spawn extension (from-sip, 4040, 1) exited non-zero on
'SIP/4035-256b'
--- extensions.conf ---
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail2(u4035)
exten = 4035,102,Voicemail2(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail2(u4009)
exten = 4009,102,Voicemail2(b4009)
exten = 4009,103,Hangup
exten = 4040,1,VoicemailMain2(${CALLERIDNUM})
[local]
include = from-sip
--- voicemail.conf ---
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[from-sip]
4009 = 1234,Jeff
4035 = 1234,Pam
sip.conf ---
[general]
port = 5060
[4035]
type = friend
username = 4035
secret = pamela
context = from-sip
callerid = Pam 4035
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4035
defaultip = 192.168.1.104
[4009]
type = friend
username = 4009
secret = jeff
context = from-sip
callerid = Jeff 4009
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4009
defaultip = 192.168.1.105
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Re: [Asterisk-Users] Cannot access voicemail -- RESOLVED
FYI --
I changed the phone's DTMF mode from in-audio to via RTP (RFC2833)
and that fixed the problem.
Jeff
On Fri, 2005-04-08 at 13:15, Jeff Heath wrote:
I think maybe it has something to do with tones from my phone not being
recognized. I just called an IAX number to make a test call and when
the IVR asked me to press a number it didn't take. That would be
consistent the CLI msg that says Unable to read password. It's also
consistent with my experience that it takes about 5 seconds for Asterisk
to come back to me with the incorrect password response.
So is there a setting I need to make on the phone or in one of the
config files to get asterisk to recognize DTMF digits or something?
Thanks,
Jeff
On Fri, 2005-04-08 at 12:48, Jeff Heath wrote:
I'm having trouble checking voicemail. When I make a call and the
recipient doesn't answer, the call goes to voicemail, and it's being
recorded (I checked the files in
/var/spool/asterisk/voicemail/from-sip/4035/INBOX).
My problem is that I can't get access to the recorded message. I dial
the extension I setup to go to voicemail (4040) and then the voicemail
system asks for a password. I press 1234 (which is what I *think* I
setup in voicemail.conf), but I get a message that the password is
incorrect.
I've included my config files below.
this is the output on the Asterisk CLI:
*CLI -- Executing VoiceMailMain2(SIP/4035-256b, 4035) in new
stack
-- Playing 'vm-password' (language 'en')
-- Incorrect password '' for user '4035' (context = any)
-- Playing 'vm-incorrect'Untitled 1 (language 'en')
-- Playing 'vm-password' (language 'en')
Apr 8 12:12:24 WARNING[22786]: app_voicemail.c:3389 vm_execmain: Unable
to read password
== Spawn extension (from-sip, 4040, 1) exited non-zero on
'SIP/4035-256b'
--- extensions.conf ---
[general]
static = yes
writeprotect = yes
[from-sip]
exten = 4035,1,Dial(SIP/4035,20)
exten = 4035,2,Voicemail2(u4035)
exten = 4035,102,Voicemail2(b4035)
exten = 4035,103,Hangup
exten = 4009,1,Dial(SIP/4009,20)
exten = 4009,2,Voicemail2(u4009)
exten = 4009,102,Voicemail2(b4009)
exten = 4009,103,Hangup
exten = 4040,1,VoicemailMain2(${CALLERIDNUM})
[local]
include = from-sip
--- voicemail.conf ---
[general]
format = wav49|gsm|wav
serveremail = asterisk
attach = yes
maxmessage = 180
maxgreet = 60
skipms = 3000
maxsilence = 10
silencethreshold = 128
maxlogins = 3
[from-sip]
4009 = 1234,Jeff
4035 = 1234,Pam
sip.conf ---
[general]
port = 5060
[4035]
type = friend
username = 4035
secret = pamela
context = from-sip
callerid = Pam 4035
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4035
defaultip = 192.168.1.104
[4009]
type = friend
username = 4009
secret = jeff
context = from-sip
callerid = Jeff 4009
qualify = 1000
host = dynamic
canreinvite = no
mailbox = 4009
defaultip = 192.168.1.105
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Re: [Asterisk-Users] Convertnig from Norstar to * to save money
On Fri, 2005-04-08 at 17:16, Mike Robinson wrote: Snacktime, it sounds like you are trying to build a business case for the migration from Norstar to *. If so, there is a solution that makes the business case a no-brainer. There is a gateway that enables you to re-use all your Norstar phones and wiring but replace the PBX itself with *. You get rid of the Norstar and move to the full IP PBX capabilities of * (and eliminate the PBX maintenance $), but you don't have to spend a bunch of money on new IP phones and the LAN upgrade to power them. Since the cost and hassle of the migration is reduced, the business case is much easier to make. Also, any nervous execs feel more comfortable with the switch because the phone on their desk doesn't change and you can always go back quickly if things don't work out. See the wiki link below for more info. http://www.voip-info.org/tiki-index.php?page=Digital%20Telephone%20Adapt ers Mike, I checked out the link. Seems like a good idea. Do you have direct experience with this? Jeff -Original Message- From: Jeff Glassman [mailto:[EMAIL PROTECTED] Sent: Thursday, April 07, 2005 7:10 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 9, Issue 67 Message: 6 Date: Thu, 7 Apr 2005 16:24:18 -0700 From: snacktime [EMAIL PROTECTED] Subject: [Asterisk-Users] Getting a good deal on a PRI To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 We have 10 incoming POTS lines to our offices, and a nortel norstar pbx. I've been looking at replacing it with * at some point in the future, and the point that looks most cost effective is when we move to PRI. Problem is, I'm not really sure how to go about getting a good deal, or what questions to ask. 90% of calls will be inbound. I called up Qwest and they quoted me $800 month. I haven't called up any CLEC's yet to see what they can do. Any suggestions? We are in Seattle, Washington. Chris In Columbus Ohio we pay about $600.00 per month for a PRI from Time Warner. Unlimited incoming/outgoing. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting with Asterisk-SIP
This may not work for you on a student budget unless you're willing to cut into your beer budget ... :-) I recently got the book and cd from Signate and got a system up pretty quickly that can make calls back and forth between a couple of SIP phones directly attached to the server. take a look at www.signate.com Jeff Heath On Sat, 2005-04-02 at 11:24, ruben cuevas rumin wrote: Hi all, I'm a Telecomunication Engeenering student. I have to develop a VoIP apliccation using SIP protocol. I have to develop the SIP Server, and the SIP clients. I think I can use Asterisk for this issue. I have installed it and I have run it, but I don't know how I have to configure it. I have read the documentation, but It's so much big and I don't know what I have to do. Someone could tell me what configuration files have I to use, and what have I to put in this files?. If is it posible, I would like someone send me some simple examples of this files. It would be wonderful if someone could help me. Thanks in advance. Best Regards, Rubén. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec audio quality comparisons?
The quality measurement you're looking for is Mean Opinion Score (MOS). It's an ITU standard for measuring call clarity. Google codec MOS and one of the first links to come up will be a Cisco technical note. It's not a comprehensive list, but it's pretty good. MOS is a scale from 1 to 5 with 5 being the best. Really, the scale is 1 to 4.5 (but that's a long explanation). Here are some codec MOS scores: codec MOS delay (msec) G.711 4.1 0.75 G.729 3.9210 the difference between 4.1 and 3.92 is significant and so is the 10 msec delay. Jeff Heath On Thu, 2005-03-31 at 02:04, Scott Bussinger wrote: I've seen lots of comparisons between the various codecs with respect to bandwidth requirements, but are there any comparisons with respect to quality? I'm currently using ulaw internally and gsm to connect to my ITSP, but should I be using different codecs to get better sound? Could someone who's played with all of them rank them in order of quality or point me to somewhere that's got a comprehensive comparison? For example, is it worth paying money for g.729 licenses for my users? Should I be hounding my ITSP to support SPEEX? Is g.711 absolutely the best possible sound given the bandwidth it uses? Just to be clear, I'm calling highest quality the least noise and hiss, fewest sound anomolies, least echo effects, and least delay. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo on internal SIP
The echo is almost certainly acoustic echo from the phones at the other side of the conversation (other side being the side opposite where the echo is heard). Acoustic echo in phones comes from coupling within the phone (i.e. sound waves being transmitted inside the phone case or through the body of the phone - this is a problem in cell phones). The other place it comes from is sound waves bouncing off the walls inside a room (speaker phones only). Good quality phones are engineered to reduce the acoustic coupling inside the phone and have acoustic echo cancelers that cancel any echo that does occur. Turning down the volume on the phones is a good first step. Unfortunately, I don't know enough about Asterisk echo cancellation to render any useful advice about Asterisk settings, but I do know this much... Most (almost all) echo cancelers in telephone network equipment are designed to cancel the electrical echo that is created by a hybrid that converts signals from 4 wire to 2 wire. Canceling acoustic echo is more difficult than canceling electrical echo. Many times an echo canceler that works on electrical echo won't work nearly as well on acoustic echo. Hopefully, someone who knows more about Asterisk's echo cancellation capabities will post a follow up about whether or not there are settings in Asterisk that might help. Jeff Heath On Thu, 2005-03-31 at 10:36, Philip Siegrist wrote: Hi All, On my * server I am getting echo on internal SIP calls. I.E. Sip 2 Sip. Calls going over the T1 via the T100p are fine. I have used ulaw and gsm, gsm has less echo but it is still noticable. All phones are snom 190s. Any ideas on what i can do to cancel this. Thanks, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco's description of echo
On Sat, 2005-03-26 at 09:16, Michael George wrote: We are having trouble with an installation that is getting a lot of echo on some calls. The installation is all SIP phones and they have a VoIP provider. When we call through the voip provider and into another of their customers (voip throughout) there is no echo problem. If we call in their landline, through the TDM400's FXO to one of the SIP phones, there is no echo problem. Sometimes when we dial from SIP -- Voip provider -- PSTN -- destination it is okay, but other times the echo is horrible. In trying to figure this out, I found this article at Cisco's site: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml#1041385 It claims that echo always comes from the far end of the connection. So if I hear echo, then the origin of the echo is in the equipment on the end of the line near the person to whom I'm talking. The description seems to make sense, but the zapata.conf setting for echo cancellation seem to also help echo on the near end of the connection. I have read about echo on the wiki and in the mailing list, but it almost always discusses it with respect to the digium cards, not SIP alone. Is the Cisco article accurate? Thanks! The Cisco article is accurate, but incomplete. Echo is a difficult problem to troubleshoot. You described an intermittent echo problem. Here's the most likely reason for it Telephone companies install echo cancelers in racks and the echo cancelers are a shared resource. That means that the same echo canceler is not always attached to the same T-1. If one of the echo cancelers is wired backwards (not often, but it happens) it will still pass signal, but it won't cancel the echo from the 4-wire to 2-wire hybrid. This scenario is one reason why you might have intermittent echo. Another is if the far end is using a low quality speaker phone and there is acoustic echo (i.e. sound waves bouncing around the room, just like echo in a cave). Acoustic echo is harder to cancel than electrical echo. If the echo cancelers use old technology and don't take frequency distortions into account (many don't) then the acoustic echo won't be canceled very well. I'm new to Asterisk, but from what (admittedly little) I've learned so far about Asterisk echo cancellation, I don't think the tail length is long enough. This link http://www.voip-info.org/wiki-Asterisk+echo+cancellation says that the Asterisk echo cancellation algorithm uses a finite impulse response filter with 128 taps. 128 taps divided by 8 taps per millisecond = 16 milliseconds of echo cancellation. If this is correct, then the algorithm is doing absolutely nothing for you. Echo cancelers that sit very close to the source of the echo (the hybrid) have tail lengths of 16 msec. For echo cancellation, the round trip delay is important because an echo canceler works by comparing a copy of the original signal to the returned signal to determine whether or not there is echo. Tail length is how long echo canceler will look for a signal that is close to the original signal (i.e echo). The Asterisk PBX is a long way (in time) from the source of the echo. Consider this Asterisk --- network --- PSTN --- hybrid (echo) --- end user | | near user --- echo cnx --- jitter buffer --- ntwk --- PSTN -| What's the round trip time? It's at least 80 milliseconds. If you're designing an echo canceler you must allow for it to be at least 128 msec (that's because echo cancelers make use of FFTs and therefore always have tail lengths that are a factor of 2). This means that the Asterisk tail length needs to be 256 milliseconds (256 msec * 8 taps per msec = 2048 taps). As I said before, I'm new to Asterisk, so I don't know if there's a configuration setting to increase the number of taps. I will say this though... echo cancellation requires a lot of processing cycles. In the PSTN, echo cancelers are hardware devices that use DSPs with the FFT algorithms in silicon. Do a 2048 tap echo canceler in software for 100 simultaneous call streams and you'll burn a lot of processor cycles. Echo is a complex problem. Jeff Heath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] small qos switch
On Sun, 2005-03-27 at 12:08, Andrew Latham wrote: I heard a great solution at Linux World Boston. A rather talented young man mentioned using a IPV6 VPN on the IPV4 internet. IPV6 supports QOS by default. Just VPN straight back to the CO and have your POP there so you only need one firewall too. You could also get an old, cheap computer off eBay put it between the switch(es) and the dsl modem, install linux and then use it to do your QoS prioritization. Not very elegant or professional looking, but it would work if you don't care about such niceties. On Fri, 25 Mar 2005 09:13:24 -0800, Bob Knight [EMAIL PROTECTED] wrote: I have multiple locations running * where all the phone are on their own lan and all the data is on a separate lan. The problem is they are sharing the same dsl connection. The locations are IAX2 trunked together, but it only takes one data down/up load to just kill the voice. What I am looking for is a small switch with QoS that I can stick in ahead of the dsl modem. Plug in one connection from the voice lan and one from the data lan. I have found quite a few 24 or 48 port switches that will do this, but I really do not need anything that big. There are already switches in place. Any recommendations please? thanks, bk... -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
