Ok,
I have spent that past 4 - 5hrs working (trying to) figure some AGI
syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I
am lost. I need to figure out how to send STDIN into the script. I
understand the concept of it, but lost when it comes down to it. No
matter what I
I have this http://www.doorbellfon.com installed on fxo ports
supports 2 doors $300us for
-1 door box, 1 ctrl panel, 130
-2 Outbound Relay Trigger Controller to open standard electric locks, 100
-1xtra door box, 50
http://www.marko.net/asterisk/archives/0209/0077.html
On Thu, 27 Nov 2003, PBX wrote:
I have spent that past 4 - 5hrs working (trying to) figure some AGI
syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I
am lost. I need to figure out how to send STDIN into the script. I
understand the concept of it, but lost when it
http://asterisk.gnuinter.net
Jeremy McNamara
PBX wrote:
Ok,
I have spent that past 4 - 5hrs working (trying to) figure some AGI
syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I
am lost. I need to figure out how to send STDIN into the script. I
understand the concept
Peter Zeltins wrote:
I do not have a hardphone to play around with, but the echo is there both
with built-in audio card (SigmaTel) and Bluetooth headset. There are no
mixer settings than I can adjust as well. I'll try disabling AGC and/or
lowering mike sensitivity.
Peter
According to the
Thanks for all the help and I found the different cadences in chan_zap.c.
Richard
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Walker Haddock wrote:
On Wed, Nov 26, 2003 at 08:33:13PM +0100, Olle E. Johansson wrote:
Asterisk wrote:
Hello!
Does anyone know where I can find out about the CDR fields?
I know most of them are self expiatory, but what is disposition for?
I've done a search in Google, I even went to
Ok
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with status or IM. So maybe there is some quasi presence
avaible? I
Thank you.. I had found this package earlier in the evening... I guess I
just need a Big Jolt of Caffine... I got it rollin now...
Thanks again
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
McNamara
Posted At: Thursday, November 27, 2003
Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish to test and or make this better?
(I know some of the code can be put into functions)
bkw
Do you think this will be merged into the CVS (seeing as its based on
LGPL) or will it be an addon?
Later..
Hi all,
I haveVOIP network built with MGCP endpoints.The
manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and
found it very useful for me. I configured it and it seems taht everything works
OK when I am testing it with one or two endpoints. After that I tried to
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 27 November 2003 05:57, Patrick Cantwell wrote:
Just for the record, I'm using a slackware 9.0 and a slackware 9.1 box
with asterisk, and I had no problems obtaining, building/compiling, or
running asterisk with a fresh install.
I'll
I have a queue defined as
[blabla]
member = SIP/101
member = SIP/102
and in extensions.conf
this:
exten = 101,1,Queue(blabla,t)
exten = 101,102,Congestion
but when both Agents are busy then still the called party does not get a
busy signal.
What I`d want is when both Agents are busy that the
Hi,
I forgot tosay that I have about 300 MGCP endpoints in
my real network.
Best regards,
Sergi Gabunia
- Original Message -
From:
Sergi Gabunia
To: [EMAIL PROTECTED]
Sent: Thursday, November 27, 2003 12:05
PM
Subject: [Asterisk-Users] MGCP
problem
Hi
slack here too - * is working STABLE
Lubo
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 27 November 2003 05:57, Patrick Cantwell wrote:
Just for the record, I'm using a slackware 9.0 and a slackware 9.1 box
with asterisk, and I had no problems obtaining,
Hi,
Calling the FWD, i see a feature a little different.
I don't call any number, but TALK with the system and they go to others parts of the showed menu.
There are any way to make the same with * ?
Where are the link that i'm talking about.
http://fwd.pulver.com/callme.php?userid=5
Hi,
We will shortly launch a sip service. Architecture is:
SER: for SIP registration and IP call routing, incoming number
termination, STUN, Nat traversal etc.
Asterisk: outgoing call routing, calling card platform, billing,
extended facilities e.g. voicemail etc.
Works well.
Tan
The following transfer led to a crash of asterisk, without leaving a core
or any utterances in messages or debug file. It looks like the zombie which
was created during the MASQ-transfer was not cleaned up... But why did
it start
a Dial??? And... why does Asterisk die when this happens??
Yes.
Jan.
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
/O
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Hi!
but when both Agents are busy then still the called party does not get a
busy signal.
What I`d want is when both Agents are busy that the caller gets a busy,
not the long tones, like the phone is ringing, but nobody answers it (
this is how this works now)
Any Ideas?
Method 1:
I posted my solution yesterday and it is $50 so not sure why people are
still asking for a cheap solution.
I have a cheap disposable walmart phone on the channel bank, in
immediate mode so when it is picked up it jumps immediately to the default
context.
I can also dial it like any other
I just fixed a problem with Asterisk where it would not fill in all the
headers correctly in the 407 - Proxy Authenication Required message
that was causing some sip phones not to work with Asterisk.
That fix might fix your problem as well. :)
Have a good one.
--Greg
On Sun, 23 Nov 2003
Jan Janak wrote:
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
Yes.
Followup question:
Both as a SIP UA (Client) and as a SIP proxy?
/O
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On 27-11 15:14, Olle E. Johansson wrote:
Jan Janak wrote:
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
Yes.
Followup question:
Both as a SIP UA (Client) and as a SIP proxy?
I don't know, I tried asterisk as a SIP UA
Jan Janak wrote:
On 27-11 15:14, Olle E. Johansson wrote:
Jan Janak wrote:
On 26-11 22:16, Olle E. Johansson wrote:
Anyone that knows if the Asterisk SIP channel supports symmetric RTP?
Yes.
Followup question:
Both as a SIP UA (Client) and as a SIP proxy?
I don't know, I tried asterisk as a
Philipp von Klitzing wrote:
Hi!
but when both Agents are busy then still the called party does not get a
busy signal.
What I`d want is when both Agents are busy that the caller gets a busy,
not the long tones, like the phone is ringing, but nobody answers it (
this is how this works now)
In my extensions.conf file I'm attempting to distinctively ring one of my
zap channels with a different ring depending upon whether the call is
received from the DID or inside extension. The DID extension looks like
so:
exten = 5551236543,1,Dial,Zap/28r1|20
However, when I dial in on
[EMAIL PROTECTED] wrote:
My problem is the tutorial left out how to configure a SJPhone so
that it connects to my asterisk server not directly FWD. I've tried
everything I can think of, I must be missing something simple.
I'm using SJPhone with the following config:
sip.conf:
[markspc]
Have you tried SER to * in the same setup?
David
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Jan Janak
Verzonden: donderdag 27 november 2003 15:26
Aan: [EMAIL PROTECTED]
Onderwerp: Re: [Asterisk-Users] Symmetric RTP
On 27-11 15:14, Olle E. Johansson
I tested the following scenario:
private network| public internet
SIP Phone 1 --- Asterisk --- NAT --- SER --- SIP Phone 2
and it worked. I was able to make calls from phone 1 to phone 2 and vice
versa.
Jan.
On 27-11 16:37, David Luyens wrote:
Have you tried SER to * in
Small tutorial:
these errors are too generic to be solved in such way...
hey my asterisk crashed, why it did?... there're many
reasons...
First: set ulimit -c unlimited on the console
from which * starts, to let it dump cores.
Then start it with 'g' in his parms , like
asterisk -vvvgc, to enable
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the
Anton Yurchenko wrote:
I just wanted to say, that I patched the code like I wrote below, and it
works. When all the operators are busy, then it drops to priority + 101.
If that would break something please write me ASAP ;)
Philipp von Klitzing wrote:
Hi!
but when both Agents are busy then
Matteo,
I AM running -gc and ulimit -c unlimited (from safe_asterisk) on RH7.2
Thats the weird thing... it crashed without any message. And looking
through the
source I still don't see how the Dial could start on a Zombie channel...
But you are right, I'll try to reproduce it tomorrow
Hello everybody,
I have started using Asterisk in a call center with ACD.
I have noticed something and I wonder if anyone knows whether it is a
bug or a feature!
I am using Queue application to ring a number of agents that have logged
on using AgentCallbackLogin.
Now, while an agent receives
Olle E. Johansson [EMAIL PROTECTED] wrote:
And now, Marks information on SJphone and Asterisk is appended to the Wiki:
http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone
Thanks for posting that for me - I'm honored! :) I've touched it up a bit
to improve my writing. I'm no
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with status or IM. So maybe there is some quasi presence
avaible? I
Assuming you haven't cvs updated yet I can look at this problem but I need
matching sources/binaries/cores. If you've cvs updated, there isn't much
I can do.
Mark
On Thu, 27 Nov 2003, Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 27 November 2003 05:57,
Hello again,
I have noticed with Queues and roundrobin policy that if even if a
timeout is set for a queue, Asterisk keeps ringing an available member
of the queue after the timeout expires. This continues a few times
before the next available agent is tried.
I am using CVS of August 17 but I
That was the whole reason I did this. Since the unixODBC stuff is LGPL we
can side step all the drama. :P
I still wanna clean it up a bit more
bkw
On Thu, 27 Nov 2003, WipeOut wrote:
Brian West wrote:
http://bugs.digium.com/bug_view_page.php?bug_id=586
woop... Anyone wish to
Hi all,
We would like to increase the size (sample length) of RTP packets sent by
Asterisk to SIP phones. I gather that Asterisk currently always uses 20ms
packets for RTP, although I can't find in the source where that's defined,
unless it's in chan_zap.c.
I'm guessing from
Hi!
Now, while an agent receives a call from the Queue they cannot logoff
using AgentCallbackLogin. Instead the Agent is asked for their agent no
and their password and after that they get the agenty-alreadyon
message!!!
When the call goes away they are able to logoff!!
I remember having
who says they have a channel bank, or a zaptel card for that matter.
your $50 solution figure would then be a bit skewed, eh?
Jon Pounder wrote:
I posted my solution yesterday and it is $50 so not sure why people are
still asking for a cheap solution.
I have a cheap disposable walmart phone on
who says they have a channel bank, or a zaptel card for that matter.
your $50 solution figure would then be a bit skewed, eh?
so use a port on one of the 4port fxs cards then.
no matter what solution you use, you still need a port on the pbx to
connect to.
or pickup one of the $10 x100p
Now, while an agent receives a call from the Queue they cannot logoff
using AgentCallbackLogin. Instead the Agent is asked for their agent no
and their password and after that they get the agenty-alreadyon
message!!!
When the call goes away they are able to logoff!!
when the nice lady
Hello,
I've been lurking around the mailing list and browsing around on
Asterisk-related links while I wait for my X100P to come in the mail.
So far I haven't found very much information related to what I want to
do with Asterisk. I was wondering if someone could point me in the
direction
I have an order for an SPA-2000 through them, and they won't respond to
any email I send them. I've also tried calling them, but I can never
get a human. I've left voice messages, but they haven't responded.
Does anyone know any other way I can get in contact with them?
Thanks!
Steve
--On Wednesday, November 26, 2003 9:03 PM +0500 [EMAIL PROTECTED]
wrote:
On Wed, 26 Nov 2003, Angel Gabriel wrote:
I've just been informed, in an IRC room, that it is possible to use a
modem card with *. Can someon please confirm this for me? Thanks in
advance.
why don't you also mention that
Is it just me or do we have this same modem x100p clone conversation on
here at least once in two weeks literally ?
For anyone who doesn't know the facts, look at the past emails on this
subject on google.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
I've added '\\1' and Debug echos 1
I've added '1' and debug echoes \1, but regexp fails to work.
The example above is from the nic.at presentation, I can't
Thats because thats not correct.
show me your full NAPTR record.
bkw
On Thu, 27 Nov 2003, Olle E. Johansson wrote:
Olle E. Johansson wrote:
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
A working ENUM record:
; Mecosta Test routing
*.2.7.9.1.3.2.1 IN NAPTR 100 10 u E2U+X-IAX2
!^\\+(.*)$!iax2:gw-mecosta/\\1! .
Jeremy McNamara
Olle E. Johansson wrote:
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't
Brian West wrote:
On Thu, 27 Nov 2003, Olle E. Johansson wrote:
Olle E. Johansson wrote:
Anyone succeeded in using regexp replacements in ENUM, like
!\\+421257296(.*)$!sip:[EMAIL PROTECTED]
I can't get it to work in ASterisk.
I've added '\\1' and Debug echos 1
I've added '1' and debug
Hcqm wrote:
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
Yes, please go ahead.
I'm running Asterisk on both LInux and FreeBSD servers
without any PSTN or ISDN hardware.
Have fun!
/Olle
___
CAN I USE/COMPILE ASTERISK
without any telephone/sound card?
I only want to use it as a IP PBX.
Yes.
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Hi,
I am thinking of starting a project at Work to pilot to use of Astrisk
and VOIP, and would like to tie in several projects together.
Currently we are looking at purchasing an ISDN and H.323 based video
conferencing system from Polycom. It suggests the use of 3-4 BRI lines
(ie 6-8 B
Hi people.
The latest version of my Ethereal plugin for IAX2 is now available here:
- http://almaw.com/ethereal-iax2-plugin-0.3.zip
A screenshot showing what you're missing is here:
- http://almaw.com/ethereal.png
The new version adds the following features/bugfixes:
- Decomposes the CODEC
'These' people don't bother reading, cause when they search for list
messages about modem cards, there's one real post, and a dozen smart
remarks about the fact they we've been there and done that already.
If all you people put your efforts together to create an easy to use web
front for
Hi!
You (not the flamers) who really want to look for information - try looking
at the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk as it
is easy to use, and most of the important posts from this list end up there.
Question:
Why does the Wiki search return nothing when I
On Thu, 2003-11-27 at 12:03, Mark Spencer wrote:
Yea, cause I used both Kphone and Windows messenger, and they
successfully registered (and subscribed i think) towards asterisk. Using
Kphone I even get a online status on all other users on the asterisk but
no interaction with status or IM.
Hello,
I was successfully using the BT-100 phone with CVS 11/10. Now that I've
upgraded to 11/27, I can't place an outbound call. However the phone is
registered and works well with inbound calls. Any suggestions will be
appreciated. Thank you.
Regards,
Christopher
So, anyone got a solution for under AUD$100 ?
Surely this is really just a bunch of cheap/commodity electronic
components?
[EMAIL PROTECTED] wrote:
I posted my solution yesterday and it is $50 so not sure why
people are still asking for a cheap solution.
I have a cheap disposable
Hi All,
It's a long shot but may be someone has CISCO-ATA-186 for sale in
Australia?
Pls contact me off list if you do.
Thanks
Alex.
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I was successfully using the BT-100 phone with CVS 11/10. Now that I've
upgraded to 11/27, I can't place an outbound call. However the phone is
registered and works well with inbound calls. Any suggestions will be
appreciated. Thank you.
Hi!
I encounter similar problems. But in my case also
netstat -nap and see if any ports are open by asterisk, run asterisk by
doing asterisk -vc and see if there's any error messages. (don't quit
asterisk to do netstat -nap) Should work fine on RH9
- Original Message -
From: Hcqm [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday,
On Thu, 2003-11-27 at 18:49, Arnold Ligtvoet wrote:
Hi,
I have started development to import the mailinglist archives into a MySQL
database and creating a full text search possibility on this. My questions;
1) Is this already done somewhere else ?
2) Is this of interest ?
I actually just
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
See page 27 of the Asterisk Handbook, version 2 for further details.
Steve.
On Friday 28 November 2003 01:53, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to
If you go to google and add site:lists.digium.com then your keywords..
you can search the list.
bwk
On Fri, 28 Nov 2003, Arnold Ligtvoet wrote:
Hi,
I've been on the list for slightly under a month now and noticed;
a) a fairly high amount of traffic,
b) a lot of questions which come up more
exten = _0119.,1,blah
exten = _011.,1,blah
would that work?
On Fri, 28 Nov 2003, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to specific number pattern.
Now, I need the following:
Calls to 0119XXX - Blocked the calls
Calls to 011 - Route the
I believe there's a cool site that indexes many mailing lists, including
asterisk. google for mailing list archives or similar. Sorry i can't
remember the name atm.
- Original Message -
From: Leif Madsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, November 28, 2003 12:13 PM
On Thu, 2003-11-27 at 16:19, Harry McGregor wrote:
Hi,
I am thinking of starting a project at Work to pilot to use of Astrisk
and VOIP, and would like to tie in several projects together.
Currently we are looking at purchasing an ISDN and H.323 based video
conferencing system from
Ok I guess I need to stop being lazy, and actually finish this project.
If you don't hear from me on the list about it in the next couple weeks,
bug me.
I'll complete the hardware, and take photos of it. (I have several loose
electric strikes besides the one on the front door so I can
On Thu, 2003-11-27 at 13:39, Andrew Nelson wrote:
--On Wednesday, November 26, 2003 9:03 PM +0500 [EMAIL PROTECTED]
wrote:
On Wed, 26 Nov 2003, Angel Gabriel wrote:
I've just been informed, in an IRC room, that it is possible to use a
modem card with *. Can someon please confirm this for
Just a warning that you will not know when the line is picked up with a
X100P. Later when you upgrade to T1/E1 service you will know when it is
picked up.
So I assume that means that I should just wait until I hear some level
of noise and then I know that the line has been picked up?
You may
On Thu, 2003-11-27 at 19:17, Brian West wrote:
exten = _0119.,1,blah
exten = _011.,1,blah
would that work?
Unless you are in a location that needs to support varying length
outbound phone numbers, you should really fully define the pattern. This
will let asterisk know ahead of time how many
Thank you for the reply, please see questions in-line.
On Thu, 2003-11-27 at 18:36, Steven Critchfield wrote:
snip
I noticed that some of those Polycom units supported V.35 connections.
You could possibly get a V.35 card for your asterisk computer and split
the channels off to the V.35
Yes I recall simlar from the handbook.
bkw
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
On Thu, 27 Nov 2003, Steven Critchfield wrote:
On Thu, 2003-11-27 at 19:17, Brian West wrote:
exten = _0119.,1,blah
exten = _011.,1,blah
would that
Come on guys how hard is it to add site:lists.digium.com into the google
search box along with your keywords? Or is that like too hard?
On Thu, 27 Nov 2003, Dustin Knuttgen wrote:
Would really love to see a searchable archive. I think it would be very helpful.
Thanks for taking this project
Carl Youngblood wrote:
What is EAGI? I will probably use festival for the time being, but I
thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the likelike
Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy. It's quite
straight forward.
Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this. I'm sure within the next few
Ok.. I was thinking about this.. It is not a very wise decsion to put
the user input in a loop.. So how could I do some error checking outside
of the loop?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Thursday, November 27, 2003
On Thu, 2003-11-27 at 19:48, Carl Youngblood wrote:
Just a warning that you will not know when the line is picked up with a
X100P. Later when you upgrade to T1/E1 service you will know when it is
picked up.
So I assume that means that I should just wait until I hear some level
of
On Thu, 2003-11-27 at 20:06, Harry McGregor wrote:
On Thu, 2003-11-27 at 18:36, Steven Critchfield wrote:
Another route would be to help hack up the libpri code to support the
3DSO or Brite signaling to support ISDN lines of T1. CAC has ISDN cards
for the ADIT600, and I'm sure there is an
Hi
When i make a call using IAX2, the log of the remote asterisk say
Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389):
RFC3389 s upport incomplete. Turn off on client if possible
Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389):
RFC3389 s upport
Jon Pounder wrote:
I posted my solution yesterday and it is $50 so not sure why people are
still asking for a cheap solution.
I have a cheap disposable walmart phone on the channel bank, in
immediate mode so when it is picked up it jumps immediately to the default
context.
I can also dial it like
On Thu, 27 Nov 2003, Brian West wrote:
Come on guys how hard is it to add site:lists.digium.com into the google
search box along with your keywords? Or is that like too hard?
This doesn't always work well. For example, ten days ago a message came
through the list with the text voicepulse
What is EAGI? I will probably use festival for the time being, but I
thing that I would eventually like to use ScanSoft's RealSpeak SDK
because it is so life-like. Unfortunately our text alerts are fully
customizeable, so we can't pre-record them.
Beware the likelike TTS, that sucks up
Greg Hill wrote:
On Thu, 27 Nov 2003, Brian West wrote:
Come on guys how hard is it to add site:lists.digium.com into the google
search box along with your keywords? Or is that like too hard?
This doesn't always work well. For example, ten days ago a message came
through the list with the
Or maybe noise would have to last for more than a certain period of time
before it triggered another waiting sequence. Like, say, if noise lasts
for longer than 2 full seconds or something.
That may be fine. Although you may have trouble with some line that also
is feeding back echo. That
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