Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
Indicating rfc2833 is exactly what asterisk does when it receives an invite from a server or device that indicates rfc2833 is available regardless of whether or not dtmfmode=inband. I will get a sip debug and open a bug report when I have a few minutes. Kevin P. Fleming wrote: - Stagg Shelton [EMAIL PROTECTED] wrote: I did ultimately force asterisk to the point where it will not accept or send rfc2833. I did this by modifying chan_sip.c in the function Asterisk should not be sending an SDP with RFC-2833 in it when the dtmfmode=inband in sip.conf. If it is doing that, please capture a 'sip debug' of this happening and opening a bug on bugs.digium.com :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
Mick,
I did ultimately force asterisk to the point where it will not accept or
send rfc2833. I did this by modifying chan_sip.c in the function
add_noncodec_to_sdp I simply put a return statement in the logic before
it got to the point where asterisk would add all the supported codecs to
the sdp. Here is the snippet of the code that I altered.
static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int
sample_rate,
char **m_buf, size_t *m_size, char
**a_buf, size_t *a_size,
int debug)
{
int rtp_code;
if (debug)
ast_verbose(Adding non-codec 0x%x (%s) to SDP\n,
format, ast_rtp_lookup_mime_subtype(0, format));
/* Stagg Forcing Return so as to force INBAND in SDP Nothing
below should execute*/
return;
...
As I put it in my reply to the the mailing list, this is an ugly hack,
but at least for my situation it works 99% of the time. I am currently
using my server as a proxy server between my customers and my upstream
carrier, so I really don't care about rfc2833, and as such I can afford
to implement this ugly hack.
--Stagg
Mick Noordewier wrote:
Dear Stagg,
I saw your posting on Asterisk-Users, and I have a similar problem. I
haven't seen any other solution, so I though I would ask if you had an
update before I posted to the list myself.
I tried a version of your modification, removing the a=fmtp line on
the SIP invite response. However, there is still another line m=audio
10988 RTP/AVP 0 101 response that the carrier detects. This seems
unresponsive to any dtmfmode statement in sip.conf. My carrier then
sends out of band dtmf as it did before, even though I've got
dtmfmode=inband in sip.conf. Before I start hacking further, do you
know a way to suppress the rfc2833 indication in this response?
Many thanks for any help,
Mick Noordewier
[EMAIL PROTECTED]
Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
Stagg Shelton
Sun, 02 Jul 2006 13:38:23 -0700
I answered my own question. My objective was reached with a simple
return statement on line 4384 of chan_sip.c in asterisk 1.2.9.1 ftp
download. The effect that this has is that asterisk will return a 200 OK
that indicates in the SDP that only inband DTMF is supported. My carrier
detects this and their NexTone Session Switch sends out dtmf inband.
It sucks having to force asterisk to operate in this manner, but
hopefully asterisk implementation of rfc2833 will get the bugs worked
out, if they are in fact bugs, and not design desicions.
Stagg Shelton wrote:
I'm trying to figure out a way around a problem that I'm having. My
carrier sends me a SIP INVITE that indicates that the dtmf modes
available are inband (0), and rfc2833 (101). My asterisk server
(1.2.9.1) sends back a 200 OK message and shows in its SDP Media
Description that we accept inband (0) and rfc2833 (101). My carrier
therefore sends all DTMF via rfc2833 which obviously causes problems
since asterisk is configured for inband.
I've tried going pure rfc2833 with the carrier, and am having DTMF
related problems. From the research that I have done with my issue
it seems to be a problem with the way asterisk sends the rfc2833
packets out at nearly the same time. Altering the timing that
asterisk uses to send the rfc2833 packets seems too deeply seated in
asterisk. I therefore have settled on the idea on using inband for
dtmf. My termination tests using inband have been successful.
So here is what i think will solve my particular problem. I just
want to respond with a 200 OK that does not contain anything about
rfc2833 in the SDP. Is this fairly doable. I've been diging through
chan_sip.c and think I could just make a couple of modifications to
make asterisk do what I need it to. I'm hoping to have someone who
is familiar with chan_sip.c enlighten me as to whether or not this
can be done, and what functions I would need to modify in order to
make it happen.
Thank You
Stagg Shelton
www.oneringnetworks.com
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
I'm trying to figure out a way around a problem that I'm having. My carrier sends me a SIP INVITE that indicates that the dtmf modes available are inband (0), and rfc2833 (101). My asterisk server (1.2.9.1) sends back a 200 OK message and shows in its SDP Media Description that we accept inband (0) and rfc2833 (101). My carrier therefore sends all DTMF via rfc2833 which obviously causes problems since asterisk is configured for inband. I've tried going pure rfc2833 with the carrier, and am having DTMF related problems. From the research that I have done with my issue it seems to be a problem with the way asterisk sends the rfc2833 packets out at nearly the same time. Altering the timing that asterisk uses to send the rfc2833 packets seems too deeply seated in asterisk. I therefore have settled on the idea on using inband for dtmf. My termination tests using inband have been successful. So here is what i think will solve my particular problem. I just want to respond with a 200 OK that does not contain anything about rfc2833 in the SDP. Is this fairly doable. I've been diging through chan_sip.c and think I could just make a couple of modifications to make asterisk do what I need it to. I'm hoping to have someone who is familiar with chan_sip.c enlighten me as to whether or not this can be done, and what functions I would need to modify in order to make it happen. Thank You Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dtmfmode=inband but SDP also indicates rfc2833
I answered my own question. My objective was reached with a simple return statement on line 4384 of chan_sip.c in asterisk 1.2.9.1 ftp download. The effect that this has is that asterisk will return a 200 OK that indicates in the SDP that only inband DTMF is supported. My carrier detects this and their NexTone Session Switch sends out dtmf inband. It sucks having to force asterisk to operate in this manner, but hopefully asterisk implementation of rfc2833 will get the bugs worked out, if they are in fact bugs, and not design desicions. Stagg Shelton wrote: I'm trying to figure out a way around a problem that I'm having. My carrier sends me a SIP INVITE that indicates that the dtmf modes available are inband (0), and rfc2833 (101). My asterisk server (1.2.9.1) sends back a 200 OK message and shows in its SDP Media Description that we accept inband (0) and rfc2833 (101). My carrier therefore sends all DTMF via rfc2833 which obviously causes problems since asterisk is configured for inband. I've tried going pure rfc2833 with the carrier, and am having DTMF related problems. From the research that I have done with my issue it seems to be a problem with the way asterisk sends the rfc2833 packets out at nearly the same time. Altering the timing that asterisk uses to send the rfc2833 packets seems too deeply seated in asterisk. I therefore have settled on the idea on using inband for dtmf. My termination tests using inband have been successful. So here is what i think will solve my particular problem. I just want to respond with a 200 OK that does not contain anything about rfc2833 in the SDP. Is this fairly doable. I've been diging through chan_sip.c and think I could just make a couple of modifications to make asterisk do what I need it to. I'm hoping to have someone who is familiar with chan_sip.c enlighten me as to whether or not this can be done, and what functions I would need to modify in order to make it happen. Thank You Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P VPM
If you want to try and get it working try reading through http://lists.digium.com/pipermail/asterisk-users/2006-February/147198.html. I had a tough time stamping out echo to far-end analog pstn connections. Stagg Shelton www.oneringnetworks.net Imran Ahmed wrote: Use: modprobe wct4xxp vpmsupport=0 On 3/1/06, Aaron Daniel [EMAIL PROTECTED] wrote: Does anyone know how to disable the VPM in software rather than removing the card altogether? The canceler isn't working as well as the software cancelers were. Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Echo
I recently dealt with echo issues and the polycom phones, specifically when using headsets. I used the following settings in my sip.cfg to fix the problem. voice.aec.hs.enable=1 voice.aec.hd.enable=1 voice.aec.hf.enable=1 voice.aes.hs.enable=1 voice.aes.hd.enable=1 voice.aes.hf.enable=1 Also I using sip version 1.6.3.0067 and bootrom 3.1.2 Stagg Shelton www.oneringnetworks.com Anton Krall wrote: What We are hearing here is that our local users say when they raise their voice, sometimes they hear themselves back.. So I guess.. If We have RX and TX setting on the polycoms.. Lowering the TX setting might help.. For those screaming users :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Martin Joseph |Sent: Tuesday, February 28, 2006 10:53 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] Polycom Echo | | |On Feb 28, 2006, at 7:14 PM, Anton Krall wrote: | | Anyway the phone can compensate? I don't think it works that way but | worth asking.. | |If the phone has an input gain (for phone users voice) then |adjusting it down can help echo that is being generated at the |far end. ie if it's too loud coming out the speaker at the |far end, the far end mic might send it back to you. | |Sorry I know nothing about polycom in particular... | |Marty | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Over WiFi
My company One Ring Networks Inc. is a wireless ISP / carrier providing VoIP services using asterisk to our customers in the metro Atlanta area. Powering the radios has never been an issue. There is always power within 30 to 50 meters of where you will want to mount a radio. If you indeed are becoming a carrier, your biggest challenges will almost always be securing the realestate where you intend for your radio to live, gaining access to the riser core in the structure you plan to provide service to, and lastly but not least the amount of money required to turn up a customer. If you are planning to build a substantial network using wifi, the technical list of issues that await you are too long to list. All that negativity being said, I've had a blast doing my part to help build out our network. Best of luck to you. Stagg Shelton www.oneringnetworks.com Juergen K. Zick wrote: Hi there, Well, looks like you are going to start a new telephony company ,-) ...When it is a quite dense populated area then there should already be enough cables and operting providers ... If you can find support by cable owners then you could indeed start to setup some WiFi cells using their net for building up an own VPN. But these cells will probably have just 100m radius w/o directional antennas on the client side. Therefore, roughly per hotspot you get an operation area of about 0,032 km^2 and you have about 1260 km °2 to cover (a 20km radius). Guess how many hotspots you need and how many would overlap and create interference ??? And what happens, when somebody in this area has an own hotspot? How should all the hotspots be managed and the traffic routed ??? Sorry, with 801.11a/b/g this is definitely not feasable at all. Probably with bigger cells your idea could be realized using a UMTS network ... In any case, you must have a BIG investment before that can be done (and much money for radio licenses ...) However, small 801.11a/b/g islands work out for customer access in general with customer side directional antennas. But not with desktop or mobile WiFi phones ... -- Jürgen this is not really an * question but it is somehow related, i am trying to develop a working proposal for cheap and quick telephony services using Voip running over *. By running a wireless network (over 802.11 a/b/g devices), i plan to be able to reach customers directly with eithe table top or handheld 802.11 sip enabled phones. But the disadvantage is that how do i power each radio and back haul the connection (this is to be able to cover at least 20KM radius of densley populated region). During my research, i have fould out that PoE switches exists and can be used to supply both data access and power to the wifi radios, but these only run for 120meters tops b4 signall loss sets in Is there a sort of high grade cat5 cable that can propagate signals for up to 1Km? or does anyone have any idea that i could effectively cover 20Km radius wing WiFi? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: auto provision of IP501 polycom
I've been using vsftpd since fedora core 2. There was a period of time in FC3 when linux wouldn't let me create usernames with capital letters. No biggie though I just created the user with all lowercase and then went back and edited the /etc/passwd file to change the username to the appropriate case. So far in FC4 I haven't had any problems creating mixed case usernames as required for easy polycom mass deployment. Stagg Shelton www.oneringnetworks.com Noah I. Miller wrote: Hi Matt - I have the same problem. I'm running CentOS, which comes with vsftpd, do you know of anyway to do this using vsftpd? I know what you mean. I run TaoLinux on all our * machines, so they all had vsftpd installed with the OS. I had to replace it with ProFTPd because I just couldn't figure out how to make vsftpd do the capitalized usernames. A little annoyance, but easy to fix. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
I forgot about one other issue we had with the 2850. The integrated NICs caused interrupt issues with the TE411P. We had to disable the integrated NICs, and installed dual port gigabit intel NIC. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Alexander, Perhaps I'm wrong, but I have a server here next to my desk (IBM e325) and I tried to fit a normal pci card into it. The slots are completely different and the card would not fit.. this was just a pci dvi video card. The server specs say that it is using PCI-X technology for the slots so this leads me to believe that they are not compatible as one would think. Cory Andrew, I will look into the supermicro servers again, I'm not keen on the handles up front on them though, that makes for awkward handling (imo). Wow Stagg, Thank you for that first hand knowledge. These are things you just can't learn until you buy a product and experience it first hand. I'm not so sure that we want to Frankenstein our own cable for this configuration though! (yikes!) Hopefully some other people will pipe up too with some more server suggestions! On 2/22/06, Stagg Shelton [EMAIL PROTECTED] wrote: We just installed asterisk for a customer using a Dell 2850. It has 3 pci slots. My customers configuration contained a TE411p Quad Span PRI, and a TDM400P with 4 FXS Modules. The only problem that we had with the 2850 was getting power to the TDM400P. We located a power connector on the backplane that supplied the required 12v. I think it was originally intended to power a tape backup drive. We ultimately sacraficed a power supply to get at it's 12V P4 connector. We then used a voltmeter to put together a pinout for the dell power port, and frankensteined together a cable that could be used to power the TDM400P.Aside from the power issue, the platform seems rock solid. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recommended rack-mountable server anyone?
We just installed asterisk for a customer using a Dell 2850. It has 3 pci slots. My customers configuration contained a TE411p Quad Span PRI, and a TDM400P with 4 FXS Modules. The only problem that we had with the 2850 was getting power to the TDM400P. We located a power connector on the backplane that supplied the required 12v. I think it was originally intended to power a tape backup drive. We ultimately sacraficed a power supply to get at it's 12V P4 connector. We then used a voltmeter to put together a pinout for the dell power port, and frankensteined together a cable that could be used to power the TDM400P.Aside from the power issue, the platform seems rock solid. Stagg Shelton www.oneringnetworks.com [EMAIL PROTECTED] wrote: Hey everyone, I've been doing a lot of research into a decent server for Asterisk but I seem to be running and circles and now I am turning to you. The issue I have is it needs to be rack mountable (so a Dell SC430 isn't going to work) and preferably have 3 pci ports. The problem that I seem to be running into is that when I look at servers from Dell or IBM or the like they only seem to support PCI-X which (from what I understand) does not support the Digium cards that we already have and that they still make. So if anyone has a suggestion or has a server they rather prefer for it's reliability, expandability, etc, please recommend it! Thank you in advance, Mitchel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
This is with a TE411P (Digium Quad Span PRI with Voice Processing
Module). When I say pulled the zaptel trunk source, I mean I issued the
following command which pulled the latest available source code at that
time:
svn checkout http://svn.digium.com/svn/zaptel/trunk zaptel
I followed up to a message previously with the exact trunk version that
I downloaded: SVN-trunk-r941
Stagg Shelton
www.oneringnetworks.com
Eric Bishop wrote:
Is this with the TE411P? Also what do you mean by "pulled
the zaptel trunk source"?
On 2/17/06, Stagg Shelton
[EMAIL PROTECTED] wrote:
This
is my last update to my issue.Finally my echo problem is
resolved.On Monday morning 2/13/06 I pulled the the zaptel trunk
source.That night after my customers core business hours we built the
new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons.My echo
disappeared almost entirelywe made a few tweaks with the tx and rx
gain settings.My echo problem disappeared completely with the
additional tweaks to txgain.Occasionally at the very beginning of a
local call echo will exist for a second or so, but then it goes away.
In two operating days there has only been one notice from a user about
experiencing an echo.All the users were informed that they should
notify us of any echo experiences.
Here are my final configurations
zaptel trunk pulled 2/13/06 approx 10:00am est.
Asterisk 1.2.4
LibPri 1.2.2
Asterisk-Addons 1.2.1
Asterisk-Sounds 1.2.1
/etc/zaptel.conf
=
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
#bchan=25-47
#dchan=48
#bchan=49-71
#dchan=72
#bchan=73-95
#dchan=96
fxoks=97-100
loadzone = us
defaultzone=us
/etc/asterisk/zapata.conf
context=from-pstn
switchtype=national
pridialplan=national
signalling=pri_cpe
resetinterval=never
faxdetect=incoming
usecallerid=yes
echocancel=yes
echotraining=800
rxgain=4.5
txgain=-13.5
group=0
channel=1-23
Thank You for all of your pointers and support in this issue.
Stagg Shelton
www.oneringnetworks.com
___
--Bandwidth and Colocation provided by Easynews.com
--
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: asterisk t.38 pass
I'm very interested in implementing this patch. Currently I am using code that came from ionidea that I managed to get compiled into asterisk 1.0.9. I've been digging through all the comments in this bugtracker issue for the past week or so, and trying to find the time to give this new code a try. If it works well in my site, I'll put it on a few of our customers sites who complain about faxing the most, and let you know how it goes. We are a wireless ISP delivering voice over our wireless highspeed data network. I can't imagine a worse condition for faxing than long distance point to point wireless. Voice is great though :) Stagg Shelton www.oneringnetworks.com Adolfo R. Brandes wrote: turby wrote: yes, with last patch works well. thanks. Glad to be of service! Adolfo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
I didn't make any changes to any makefiles, just downloaded, built for 2.6 kernel and installed. Then I rebuilt and installed libpri, asterisk, asterisk-addons. Below is the output from ztcfg. [EMAIL PROTECTED] ~]# ztcfg -v Zaptel Version: SVN-trunk-r941 Echo Canceller: MG2 Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) 28 channels configured. Stagg Shelton www.oneringnetworks.com Andrew Kohlsmith wrote: On Thursday 16 February 2006 11:11, Stagg Shelton wrote: Here are my final configurations zaptel trunk pulled 2/13/06 approx 10:00am est. Can you tell us what SVN checkout # and echo canceller you ended up using? 'dmesg' output when you load the module will tell you, as will ztcfg -v. Also, did you manipulate zconfig.h or the makefiles in any way? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
This is my last update to my issue. Finally my echo problem is resolved. On Monday morning 2/13/06 I pulled the the zaptel trunk source. That night after my customers core business hours we built the new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons. My echo disappeared almost entirely we made a few tweaks with the tx and rx gain settings. My echo problem disappeared completely with the additional tweaks to txgain. Occasionally at the very beginning of a local call echo will exist for a second or so, but then it goes away. In two operating days there has only been one notice from a user about experiencing an echo. All the users were informed that they should notify us of any echo experiences. Here are my final configurations zaptel trunk pulled 2/13/06 approx 10:00am est. Asterisk 1.2.4 LibPri 1.2.2 Asterisk-Addons 1.2.1 Asterisk-Sounds 1.2.1 /etc/zaptel.conf = span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #bchan=25-47 #dchan=48 #bchan=49-71 #dchan=72 #bchan=73-95 #dchan=96 fxoks=97-100 loadzone = us defaultzone=us /etc/asterisk/zapata.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe resetinterval=never faxdetect=incoming usecallerid=yes echocancel=yes echotraining=800 rxgain=4.5 txgain=-13.5 group=0 channel=1-23 Thank You for all of your pointers and support in this issue. Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
This is my last update to my issue. Finally my echo problem is resolved. On Monday morning 2/13/06 I pulled the the zaptel trunk source. That night after my customers core business hours we built the new zaptel drivers, rebuilt libpri, asterisk, asterisk-addons. My echo disappeared almost entirely we made a few tweaks with the tx and rx gain settings. My echo problem disappeared completely with the additional tweaks to txgain. Occasionally at the very beginning of a local call echo will exist for a second or so, but then it goes away. In two operating days there has only been one notice from a user about experiencing an echo. All the users were informed that they should notify us of any echo experiences. Here are my final configurations zaptel trunk pulled 2/13/06 approx 10:00am est. Asterisk 1.2.4 LibPri 1.2.2 Asterisk-Addons 1.2.1 Asterisk-Sounds 1.2.1 /etc/zaptel.conf = span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #bchan=25-47 #dchan=48 #bchan=49-71 #dchan=72 #bchan=73-95 #dchan=96 fxoks=97-100 loadzone = us defaultzone=us /etc/asterisk/zapata.conf context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe resetinterval=never faxdetect=incoming usecallerid=yes echocancel=yes echotraining=800 rxgain=4.5 txgain=-13.5 group=0 channel=1-23 Thank You for all of your pointers and support in this issue. Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
I am using asterisk 1.2.4 and zaptel 1.2.3. Also, I tried the latest zaptel out of subversion. Stagg Shelton www.oneringnetworks.com Isaac Xiao (KVB Kunlun Pty Limited) wrote: What version of Asterisk and Zaptel you were using? Did you try latest Asterisk 1.2.4 and Zaptel 1.2.3? Anyone has good feedback for TE411P? Isaac Xiao Stagg Shelton wrote: It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight and still the exact same echo problem. At this point I feel like I can rule out failed hardware. I contacted Digium support and now they are telling me it's something with my carrier, and I should call them. I called Bellsouth, and they ran a full stress test on the circuit taking me offline for about 30 minutes. The end result is that the circuit test passed with no errors. Bellsouth says it's not in their network, Digium says its not their card, and I have a te411p with VPM disabled in the wct4xx kernel module because something doesn't work the way it should. My customer is wanting to know about sangoma cards with the echo cancellation, and at this point I'm nervous to recommend any hardware. I'm going to look into the sangoma that you suggested. Are there any other kinds of products that I could look into either Passive or Active. Thanks Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
Yes, right now we are only using span 1 on the quad span card with plans to pull in another T1 PRI when we get this echo problem solved. The echo is only experienced when the call terminates to traditional analog circuits both local and long distance. Calls to cell phones, and other known digital circuits do not exhibit the symptoms. I've been sent a few articles off list which discussed the reasons why this may occur, imperfect impedance where the digital circuit is switched to the analog circuit. In testing I've set echocancel=no made calls, and echocancel=yes and made calls with no real audible difference. I haven't done a zap show channel while on a call with echo, but I plan to this weekend. I was informed that the number of taps would be shown. I used a stop watch last night to try to get the delay, and it was about 1/2 to 1 second delay and was continual so long as I was talking. It wasn't affected by differing acoustical variations. I tested this using handset, headset, and speakerphone. Disabling VPM and recompiling zaptel or removing the VPM off the board completely is the only thing that has any effect on the echo. Hopefully the zap show channel will provide to me another data point to help me determine if the HW module is active. More to come... Stagg Shelton www.oneringnetworks.com Cory Andrews wrote: Stagg - I know you get a full 128ms tail of echo can on the Sangoma. I believe that on the TE411P, the 128ms tail is shared by all (4) spans, and as you add additional spans up to maximum of 4, the echo can tail amount decreases accordingly. If you are running 4 spans, you have 32ms of echo can tail on each span, not the full 128ms. Now that I'm reading back through your post thread, it looks like you were only running 1 span on the TE411P, so you should have been getting the full 128ms of echo can tail. You may not find an improvement with the Sangoma. I have never experience, nor heard of a 1-2 full second delay. Trial and error is likely your course of action. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: Stagg Shelton To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 10, 2006 11:34 PM Subject: Re: [Asterisk-Users] TE411P Really Bad Echo It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight and still the exact same echo problem. At this point I feel like I can rule out failed hardware. I contacted Digium support and now they are telling me it's something with my carrier, and I should call them. I called Bellsouth, and they ran a full stress test on the circuit taking me offline for about 30 minutes. The end result is that the circuit test passed with no errors. Bellsouth says it's not in their network, Digium says its not their card, and I have a te411p with VPM disabled in the wct4xx kernel module because something doesn't work the way it should. My customer is wanting to know about sangoma cards with the echo cancellation, and at this point I'm nervous to recommend any hardware. I'm going to look into the sangoma that you suggested. Are there any other kinds of products that I could look into either Passive or Active. Thanks Stagg Shelton www.oneringnetworks.com Matt wrote: try sangoma carrier grade 104d hardware EC card. we're using it ourself. Best Regards Matt - Original Message - From: "Anthony Rodgers" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Tuesday, February 07, 2006 12:57 PM Subject: Re: [Asterisk-Users] TE411P Really Bad Echo For what it's worth, we have been going through very similar issues with a TE411P - with Digium support, we have basically gone as far as we can with the HW EC, and are now using MG2 with much better results. We have a Ditech EC box on order. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote: On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are onl
Re: [Asterisk-Users] TE411P Really Bad Echo
It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight and still the exact same echo problem. At this point I feel like I can rule out failed hardware. I contacted Digium support and now they are telling me it's something with my carrier, and I should call them. I called Bellsouth, and they ran a full stress test on the circuit taking me offline for about 30 minutes. The end result is that the circuit test passed with no errors. Bellsouth says it's not in their network, Digium says its not their card, and I have a te411p with VPM disabled in the wct4xx kernel module because something doesn't work the way it should. My customer is wanting to know about sangoma cards with the echo cancellation, and at this point I'm nervous to recommend any hardware. I'm going to look into the sangoma that you suggested. Are there any other kinds of products that I could look into either Passive or Active. Thanks Stagg Shelton www.oneringnetworks.com Matt wrote: try sangoma carrier grade 104d hardware EC card. we're using it ourself. Best Regards Matt - Original Message - From: "Anthony Rodgers" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Tuesday, February 07, 2006 12:57 PM Subject: Re: [Asterisk-Users] TE411P Really Bad Echo For what it's worth, we have been going through very similar issues with a TE411P - with Digium support, we have basically gone as far as we can with the HW EC, and are now using MG2 with much better results. We have a Ditech EC box on order. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote: On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right now, and we have a tdm400p with 4 fxs modules installed as well. If anyone has experience with this card, can you tell me if I am missing something. 1 to 2 seconds?! That's ridiculously huge. I don't think you'll find a echo canceler anywhere that can fix your echo problem. If it gets better with the VPM disabled, then definitely contact Digium tech-support about it. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE411P Really Bad Echo
Yes, I actually just removed the VPM. After doing so, I had echo at the beginning of the call, but it trained out after a second at least making it usable. I plan on contacting digium support this morning. Do you know if there are any docs specific to this card or the vpm module itself? --Stagg www.oneringnetworks.com Kevin P. Fleming wrote: Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I 1 to 2 _seconds_? There is no echo canceler anywhere that will handle that much echo delay. Did you actually remove the VPM, or just disable it? Please check whether you have this problem with the VPM installed but disabled, because it could be a bad VPM. Finally... I know it's Sunday night, but you should really pursue this with Digium Support tomorrow morning :-) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE411P Really Bad Echo
I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right now, and we have a tdm400p with 4 fxs modules installed as well. If anyone has experience with this card, can you tell me if I am missing something. zaptel.conf = span=1,1,0,esf,b8zs bchan=1-23 dchan=24 #bchan=25-47 #dchan=48 #bchan=49-71 #dchan=72 #bchan=73-95 #dchan=96 fxoks=97-100 loadzone = us defaultzone=us zapata.conf = context=from-pstn switchtype=national pridialplan=national signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=3.0 txgain=0.0 group=0 channel=1-23 ;;[1103] signalling=fxo_ks record_out=Adhoc record_in=Adhoc [EMAIL PROTECTED] echotraining=800 echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callerid=device 1103 busydetect=no busycount=7 channel=98 ;;[111] signalling=fxo_ks record_out=Adhoc record_in=Adhoc [EMAIL PROTECTED] echotraining=800 echocancelwhenbridged=no echocancel=yes context=from-internal callprogress=no callerid=device 111 busydetect=no busycount=7 channel=97 Thanks Stagg www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo while using Headset with Polycom IP 501 / 601 Asterisk 1.2.1
I'm hearing an echo when using a headset with my IP 501 / 601. The phones are using BR 3.1.2 and SIP 1.6.3. I use tftp to configure the phones. The sip.cfg is the default from polycom except for the parameters required to connect the phones to asterisk. I have absolutly no echo with the handset, but do have a slight echo on the speaker phone. I haven't ruled out room acoustics as the source of the echo on the speaker phone. The headset is where the echo is most noticeable, the delay is about 1/2 second. The persons that I'm talking to on the other end of the connection do not hear an echo. I noticed in the sip.cfg that voice.aec.hs and voice.aec.hd are disabled while voice.aec.hf is enabled. Polycom Admin guide says don't mess with the settings -- I'm thinking I will enable the hs, and hd and see if that makes a difference. Does anyone out there have experience with tweaking the echo cancellation, and suppression settings in sip.cfg for the polycom phones. Any help would be appreciated. --Stagg ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
