Yes - the aggressive suppressor does tend to clip speech although I don't
think it is half duplex.
The MEC3 echo suppressor seemed to be heading in the right direction but
last time I tried it it went funny after a while causing speech
interruption.
Iain
--On Saturday, November 15, 2003
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hi,
i got he following error while trying to install
digium cards in red hat linux 7.3. please help.
[EMAIL PROTECTED] root]# modprobe zaptel
lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved
symbol cpu_raise_softirq_Rd01f3ee8
lib/modules/2.4.18-3bigmem/misc/zaptel.o: unresolved
symbol
It appears you don't have the same modversions.h file as your kernel was
compiled with.
Search the archives for messages like
http://lists.digium.com/pipermail/asterisk-users/2003-February/007588.html
On Sun, 2003-11-16 at 06:43, C M wrote:
hi,
i got he following error while trying to
I'v got the following scenario: Two clients (ohphone) are calling (one
at a time) the host with asterisk, which then connects to the SIP
client.
One of these hosts let's asterisk crash with a segmentation fault (i can
provide the core file, if needed) in the second, the SIP client accepts
the
The X100P cards have horrible echo problems. I've heard talk about this
being fixed, but havent seen anything done about it.
Depends on the installation; I have a half dozen of these cards with very
very little echo problem. You might want to reverse tip and ring in your
install and see if
PLEASE!
Do *NOT* reply to a list message, erase the body, change the subject and
start a new discussion! It completely destroys the list threading for
people with mail clients which can properly thread messages.
Isn't it far more work to do what you're doing instead of just clicking on
the
I think there are two ways of doing it.. Either I can create an AGI that
will run on the h extension and will lookup the last entry that
matches the account code of the call that just ended in the MySQL CDR
and calculate the call cost immediately..
Use the database. I'd recommend Postgres
But this doesn't work! As soon as we pass a number into the context,
it matches successfully against _., and we get our sorry-no-match
recording and the line hangs up. Here's how we force the ordering by
using include to regulate order of matching:
Thanks John, that's a great explanation!
On Sat, Nov 15, 2003 at 07:59:02PM +0100, Peer Oliver schmidt wrote:
What is your reason to use i4l instead of the chan_capi driver
(http://www.junghanns.net/asterisk/)? Did you try both, and found i4l
perform better?
In short: bad reason (the ability to see the AT commands). I will
try CAPI
Sorry I was not blameing the hardware I feel it is a problem with
something I am doing I am very new to this I realy want this phone to
work as they are the only cost effective Hardware sip phone I have
found.
The echo is a local echo on the phone and the user I dial gets choppy
sound.
Larry
On Monday 27 October 2003 06:15 am, [EMAIL PROTECTED] wrote:
TOP POSTING MADNESS continues...
you need to be part of the WORLD context, and not just NANPA, otherwise
011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify
-wasim
Wasim.
Can you please elaborate on this
Hi,
I have little problem and it is so embracing when u r talking to some one
and line get hang-up.
When some one calls from out of state or out of country my calls gets
randomly hang-up with in few seconds and it happens with most of the calls.
It's happening randomly I got few calls, which
After testing and playing around it seems that AST sends
what is in thesip.conf:fromuser field as the VM box.
Or SNOM is reading the wrong field in the SIP packet.
If I set sip.conf:fromuser=*98 for my SNOM phone then
when pressing MWI on that phone will ring voicemail.
From looking at
hi all
When calling (SIP|MGCP) - * - (CAPI|ZAP) - PSTN, users complain
about the receiving end gets echo, especially cellular phones. Any idea
why this may happen?
roy
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If I am dialing with a Bt101 or something that sends all the digits in a
single packet, it works great. It fails miserably, however, if I'm dialing
from a phone on an FXS port, or if I'm trying to do this on an answered
call.
Zap devices should handle this fine (maybe even MGCP), but SIP
On Sun, 16 Nov 2003, marrandy wrote:
On Monday 27 October 2003 06:15 am, [EMAIL PROTECTED] wrote:
TOP POSTING MADNESS continues...
you need to be part of the WORLD context, and not just NANPA, otherwise
011+COUNTRY+AREA+NUMBER works as my numerous jerjer bills will testify
Can you
Zap devices should handle this fine (maybe even MGCP), but SIP should
fail
with that sort of a configuration since we cannot differentiate
between Number valid, but more could be useful and Number incomplete,
therefore once we reach a match, we have to take it.
I can't get anything to
Has anyone seen an FXO converter for a Cisco ATA. There is someone
selling a device on Ebay that claims to convert a Cisco ATA FXS port to
an FXO.
FX-200 VOIP PORT CONVERTER FXS to FXO
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388
Kevin wrote:
Has anyone seen an FXO converter for a Cisco ATA. There is someone
selling a device on Ebay that claims to convert a Cisco ATA FXS port to
an FXO.
FX-200 VOIP PORT CONVERTER FXS to FXO
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3057281388
Don't bother. Support Asterisk
I do support Asterisk. I have a TDM40B and X100P from Digium, I can't
take the echo on the X100P.
-Original Message-
From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 16, 2003 1:27 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS
Kevin wrote:
I do support Asterisk. I have a TDM40B and X100P from Digium, I can't
take the echo on the X100P.
I've got dozens of X100P based systems and have only had echo trouble on
4 systems. All of them were solved by tweaking the various settings in
the Zaptel Makefile and in
The echo issues are line or PSTN.
make sure your tip and ring are correctly wired. Polarity
does matter and teh X100P does not do polarity fixing like
most consumer phones today.
john brown
chagres technologies, inc
http://www.chagres.net/products/voip/
On Sun, Nov 16, 2003 at 01:35:13PM
Excuse my ignorance, but could someone explain what tip and ring is and
how I ensure/test that it is wired correctly?
Thanks,
Ed
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Brown
(CV)
Sent: Sunday, November 16, 2003 10:54 AM
To: [EMAIL
Just to be sure again, I did a reversal on the tip and ring with no
improvement.
-Original Message-
From: John Brown (CV) [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 16, 2003 1:54 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] FX-200 VOIP PORT CONVERTER FXS to FXO
The echo
On Sun, Nov 16, 2003 at 11:11:06AM -0800, Ed Rubright wrote:
Excuse my ignorance, but could someone explain what tip and ring is and
how I ensure/test that it is wired correctly?
In the old days, phone plugs looked like 1/4 phono jacks
There was the TIP of the jack and the RING at the base
of
Ok, then I'd suggest that you need to carefully comb thru
your echo settings.
Keep in mind that you will need to STOP NOW your server for
hardware changes. Some will disagree with this, but thats been
my experience.
Like JerJer, I've got boxes with X100P cards and no echo issues
once I got
Hi,
I have two X100P cards in the same system.
I can use both of them to initiate and/or receive PSTN calls.
I want now to define separate context for each of them, in oder to route
inbound calls to different extensions.
This is what I have now in zapata.conf file:
[channels]
language=en
based on below, you have them in the same context
insert a context=foo line after channel =1
if you want channel 2 in a different context
On Sun, Nov 16, 2003 at 09:32:42PM +0200, Dan wrote:
Hi,
I have two X100P cards in the same system.
I can use both of them to initiate and/or
Hi John,
Thanks for the info. I'll be stopping by Radio Shack to pickup a
polairy tester. I have 2 X100P and 1 TDM400 card. I will be adding a
SIP phone here in the next week or so.
What do you recommend for the following:
- What echo cancellation settings in the zaptel makefile?
- What
Hi,
- Original Message -
From: John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 16, 2003 9:42 PM
Subject: Re: [Asterisk-Users] two X100P cards, different context
based on below, you have them in the same context
insert a context=foo line after
On Sun, 2003-11-16 at 19:50, Jeremy McNamara wrote:
I've got dozens of X100P based systems and have only had echo trouble on
4 systems. All of them were solved by tweaking the various settings in
the Zaptel Makefile and in zapata.conf or calling the telco and
bitching, loudly.
Earlier
I am attempting to contact John Brown from Chagres
Technologies, I know he watches this list. Please contact me ASAP John, I
have been trying to get hold of you for the last few weeks regarding an order
but so far havent had any luck!
Regards,
Aaron Martin.
unsubscribe
I have thought about doing this as well, for what may be the
same application. The easiest way to do it would be to use the
Console channel and audio drivers and use a mixer -- keep in
mind, I'm thinking of a radio talk show, presumably with a mixer,
other audio sources, etc. It would look
Make sure you have at least one blank line at the bottom of your
meetme.conf..
sorry but this isn't true mine doesn't... I have checked in vi
If yours has drama.. what editor are you using?
bkw
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Having spent 21 years in a telephone company as an engineer, reversing
tip ring will have zero impact on any 2-wire fx pstn line. The equipment
in the central office (regardless of who the manufacturer happens to be)
is balanced and supplies -48 volts that is fed through the outside plant
to
Also keep in mind if you don't come straight from the dmarc to the x100p
you might have echo also:
PSTN == X100P == * SERVER
|
|
PHONE
If you do the above you will get mad echo in some cases. :P
I have 3 x100p's with only about 3-5 seconds of echo at the begining of
Having spent 21 years in a telephone company as an engineer, reversing
tip ring will have zero impact on any 2-wire fx pstn line. The equipment
Why in some cases does it infact fix the echo issues?
bkw
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I posted this earlier on the development
list. For those of you who watch both lists, please pardon the
duplication.
Currently,in our*lab
weuseall SIP phonesso the MWI NOTIFY works
perfect.
I would like to do a pilot with some legacy gear,
however. Accordingly, I'd like to be able to have
On Sunday 16 November 2003 15:23, Brian West wrote:
Make sure you have at least one blank line at the bottom of your
meetme.conf..
sorry but this isn't true mine doesn't... I have checked in vi
If yours has drama.. what editor are you using?
What this calls to is not that you have a
http://bugs.digium.com/bug_view_page.php?bug_id=156
Anyone else try this? Feedback.. gripes.. nitpicks? Please test it out
and post to the bug note.
bkw
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On Sun, 16 Nov 2003, Josh J. Zuerner wrote:
[...]
For example, if I currently dial 1000400 on my * SIP phone, the MW lamp on legacy X
400 is flipped on by the PBX. If I dial 1001400 on my * SIP phone, the MW lamp on
legacy X 400 is flipped off.
Does this dialing capability already exist?
This is off-topic for Asterisk, but since echo problems with the
X100P cards seems to be a common issue and one which people blame on
telco loops, I'd suggest getting the following gear: a 3M Dynatel
subscriber loop test unit.
The 745 is the one I have experience with, and if you have a fairly
That might just very well be it. :P
On Sun, 16 Nov 2003, Tilghman Lesher wrote:
On Sunday 16 November 2003 15:23, Brian West wrote:
Make sure you have at least one blank line at the bottom of your
meetme.conf..
sorry but this isn't true mine doesn't... I have checked in vi
If
Hi All,
This topic has come up before in the Asterisk mailing list many times,
so I know that a lot of people have given up in waiting for a FXO card
to be approved by the Australian telecommunications authority. My
question is: all legalities aside - is anyone using a FXO card in
Australia
I replied privately back to Aaron. Seems our Spamassissin software
tagged his messages as spam. With the volume of spam email I've been
getting I haven't reviewed the spam folder in a bit.
I've noticed a couple of other emails got tagged as well and I'll
reply to those off list.
john brown
On Mon, Nov 17, 2003 at 12:13:09PM +1100, Gonzalo Servat wrote:
Hi All,
This topic has come up before in the Asterisk mailing list many times,
so I know that a lot of people have given up in waiting for a FXO card
to be approved by the Australian telecommunications authority. My
question
The answer is yes.
Peter
At 12:13 17/11/03 +1100, you wrote:
Hi All,
This topic has come up before in the Asterisk mailing list many times,
so I know that a lot of people have given up in waiting for a FXO card
to be approved by the Australian telecommunications authority. My
question is: all
I am sure that others have used it directly...
I have used it indirectly hanging off PABX extensions and even tested
them on emulators... not a problem...
The x100p in their current form will never pass a-tick and even c-tick
might be questionable.
The CE version of the card I have never seen
You will need to check with Cisco to see if the ATA188 has the same issues
with faxing as the ATA186.
http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml
Dave Weis wrote:
Should I expect a standard fax machine connected to an ata-188 connected
to an
Hi List,
Here is the config
ext 2601 is a GS BT-101 phone
ext 2062 is a SNOM 200
latest public firmware on both
asterisk is Asterisk CVS-11/14/03-22:55:45
Make a call from 2601 - 2602 life good, call works
have 2602 place call on hold. The music on 2601 IS NOT
my music on hold. It
hi,
i am getting these errors while installing asterisk. i
reconfigured kernel and i have all the modules
installed.
kernel-source
readline
readline-devel
openssl
openssl-devel
this is the error: (at the last part of the
installation)
gcc -g -o asterisk -Wl,-E io.o sched.o logger.o
frame.o
On Mon, 2003-11-17 at 12:20, Anthony Wood wrote:
I have spoken to a number of Australian users who are successfully using:
X100P
NetJet (echo issues)
AVM Fritz!Card
I hope to add myself to their number shortly, since we have recieved our Fritz!es
Also [EMAIL PROTECTED] seems to be
On Sun, Nov 16, 2003 at 08:33:22PM -0800, C M wrote:
hi,
i am getting these errors while installing asterisk. i
reconfigured kernel and i have all the modules
installed.
kernel-source
readline
readline-devel
openssl
openssl-devel
this is the error: (at the last part of the
Hi All,
I was wondering what the status of distinctive ring support in Asterisk
is? I had a google search read and Mark Spencer wrote some support for
it.
Is distinctive ring different in every country or is it pretty standard?
And for my final question, does the Wildcard FXO card support
On Mon, Nov 17, 2003 at 03:49:40PM +1100, Gonzalo Servat wrote:
On Mon, 2003-11-17 at 12:20, Anthony Wood wrote:
I have spoken to a number of Australian users who are successfully using:
X100P
NetJet (echo issues)
AVM Fritz!Card
I hope to add myself to their number shortly,
Has anyone got a mobile wireless phone working with * yet
Is it possible to use the Cisco 7920 with skinny
Regards Mick West
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I do not know the answer for #1, but for #2, I highly doubt it. What you
could do is add something to the callerid to distinguish the calls.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gonzalo Servat
Sent: Sunday, November 16, 2003
On Mon, 2003-11-17 at 16:00, Anthony Wood wrote:
ISDN (telstra Onramp 2) is very similar in price to standard telstra lines.
The only problem is you can't have ADSL ISDN on the same line.
We upgraded from 2 analogue lines to 2 digital (i.e. 4 channels) for $250.
I was a bit turned off by
No
No sip image for it yet
Also is there any way I can change messages and extensions depending on
local time ??
Also is there a way to transfer the call over PSTN if the local
extension is not answered.
Eg to a normal gsm mobile ??
Regards Mick West
NetExpress
Phone 61 08 82420173
Fax
On Sunday 16 November 2003 22:33, C M wrote:
i am getting these errors while installing asterisk. i
reconfigured kernel and i have all the modules
installed.
kernel-source
readline
readline-devel
openssl
openssl-devel
this is the error: (at the last part of the
installation)
gcc -g
On Mon, Nov 17, 2003 at 04:32:50PM +1100, Gonzalo Servat wrote:
On Mon, 2003-11-17 at 16:00, Anthony Wood wrote:
ISDN (telstra Onramp 2) is very similar in price to standard telstra lines.
The only problem is you can't have ADSL ISDN on the same line.
We upgraded from 2 analogue lines
Otherwise, maybe Icecast can be hacked a bit or glued to a
sip client i.e.:
sipclient sip:[EMAIL PROTECTED] | icecast
for some hypothetical sip client that just listens and sends
audio data to stdout.
Fortunately such a SIP client actually exists: playSIP; see
Does anyone know how to make
Calls auto transfer to a mobile if no one answers ??
Regards Mick West
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On Mon, 17 Nov 2003 [EMAIL PROTECTED] wrote:
Does anyone know how to make
Calls auto transfer to a mobile if no one answers ??
suppose your mobile number is +923008508070
exten = 15,1,Dial(IAX/farfon|30) ; try for 30 seconds on IAX
exten = 15,2,Dial(Zap/1/03008508070|45) ; then try for 45
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