Fixed it, had to modify my config, it now reads:
fxsks=1
fxoks=2-3
fxsks=4
loadzone=au
defaultzone=au
--
jeremy bogan[ [EMAIL PROTECTED] ]
segment publishing - design.develop.host
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On Tue, 13 Apr 2004, Alex Brett wrote:
> Has anybody got any experience using an X100P on an NTL phone line in
> the UK (I'm in an ex Cable & Wireless area if that makes any difference).
>
> The problem I'm having (and judging by the number of references to it
> I've found searching it is a c
On Tue, 13 Apr 2004, Jeremy Bogan wrote:
> Fixed it, had to modify my config, it now reads:
>
> fxsks=1
> fxoks=2-3
> fxsks=4
> loadzone=au
> defaultzone=au
This looks wrong. What is the full output of ztcfg -vvv?
I'd be surprised if this worked as expected once you got * started...
Out of cu
This looks wrong. What is the full output of ztcfg -vvv?
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXO Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default)
Thank you. This explains it.Nathaniel Powning <[EMAIL PROTECTED]> wrote:
On Mon, 12 Apr 2004, Ron McMillin wrote:> Is it possible to dial an OUTSIDE SIP address while inside AGI application? For example, within extension context, I could use> [from-sip]> exten => 7723,1,Dial(SIP/[EMAIL PROTECTED])
On Tue, 13 Apr 2004, Stephen Davies wrote:
> I did some work a while back to add detection of the UK busy/hangup
> signal on the line, but I never got it working well enough to depend
> on it. The problem is that it is a single frequency tone. (The US
> one is dual-tone). Women's voices used to
On Tue, 2004-04-13 at 02:19, Alex Brett wrote:
> Has anybody got any experience using an X100P on an NTL phone line in
> the UK (I'm in an ex Cable & Wireless area if that makes any difference).
>
> The problem I'm having (and judging by the number of references to it
> I've found searching it i
On Tue, 13 Apr 2004, Vic Cross wrote:
> On Tue, 13 Apr 2004, Stephen Davies wrote:
>
> > I did some work a while back to add detection of the UK busy/hangup
> > signal on the line, but I never got it working well enough to depend
> > on it. The problem is that it is a single frequency tone. (
HI,
quick and simple question: is it possible to use inband dtmf with g729?
What I would like to do is to have sip clients connected to asterisk and a zaptel
card to make pstn phone calls.
My concern is to allow sip users to use digits for call destinations that
do require menu actions while
Just a note regarding this issue.
I'm using RH9, two X100p and one TDM400
Loading them in this order:
zaptel
wcfxo
wcfxs
zapata.conf like this:
fxsks=1
fx
Good day.
I'm looking for a sip client 4 fedora???
Frdora?
Thanks
Altus
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Hi,
Can anyone please tell me what the leds (and colours ) mean on an E110P
Thanks Jon
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Stephen Davies wrote:
>Hi Alex,
>
>Indeed the call end termination doesn't work on an NTL line. I'm not
>so sure it works too well on other lines either.
>
>I did some work a while back to add detection of the UK busy/hangup
>signal on the line, but I never got it working well enough to depend
>o
Hello all
I would like to use two Asterisk Server as Redudancy System.
If we use Load Balance Server, we will be able to reach this system.
But I don't know what kind server can abailable on this system.
Vocal have Load Balance Server.
SER have SER Media Proxy by AG Project.
Who knows does its
On Sat, Apr 10, 2004 at 08:39:53AM -0600, Michael Welter wrote:
> However, when I put this stair-stepped signal to the channel bank, it
> converts it back into a digital signal. I'm thinking that, because it's
> not a smoothed signal, the analog-to-digital process injects hum and
> buz. Does _
>
> Does anyone know of Phone that supports G.723 on H.323.
>
Innovaphone tiptel 200 for example.
http://www.innovaphone.com/webneu2/products/en_IP200.asp
One of the nicest phones I've seen so far, h.323 only though.
Bye, Martin
___
Asterisk-Users
In laymans terms.
To use your telco's T-1 as the timing source
span=1,1,0,esf,b8zs,yelllow
To use the internal clock of the card you would use (I'm pretty sure that this would
only be used for channel banks, or connections to other PBX hardware. I don't think a
telco is going to use you
Hello!
Asterisk box receiving calls. Is there some way to get information about
current calls from external or AGI application?
I'm interested in:
- duration, how long calls already in the system (billing and actual time);
- source/destination phone numbers;
- etc.
In other words can I receive i
Hi,
On Mon, 12 Apr 2004 at 19:49, asterisk wrote:
[...]
> chan_capi.c: In function `pipe_frame':
> chan_capi.c:1187: error: too many arguments to function `ast_dsp_process'
> make: *** [chan_capi.o] Error 1
This looks like you are trying to compile chan_capi against a version
of Asterisk (or ins
In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having trouble. We use VoicePulse connect, which gives us
local DID for inbound and outbound calls (even though DTMF tones are no
Hi,
bristuff 0.0.2rc20 will add support for HOLD/RETRIEVE, SUSPEND/RESUME
and isdn transfers in an "experimental" way.
It also features a zaptel that works on 2.6 (and does not freeze),
together with optimized qozap drivers. Load tests have shown that it
is possible to have 6 quadBRI cards in a d
Don & others,
Thank you for your answer. The fog maybe lifting ;).
The zaptel.conf file has the following in its comments:
#
# The timing parameter determines the selection of primary,
secondary, and
# so on sync sources. If this span should be considered a
primary sync
# source, then give it a v
Well, Once upon a time, I had problems receiving callerid, and then one
day, Mark was logged into my asterisk box helping with something else,
and I asked him about this, and he showed me a nice tweak to some source
file that made it work.
Some time later, I must have done hundreds of CVS updates,
On the other end of our PRI line would be a telco switch.
On Mon, 12 Apr 2004, Scott Stingel wrote:
> Mike-
>
> You didn't say what's at the other end of your PRI line, but you might try
> having the other end be the timing sync source. Try: span=1,0,0,esf,b8zs
> instead. Maybe that will help.
Dear Nick
Very usefull function driver how can i try it?
Thanks in advance
Dimitri
On Monday 12 April 2004 21:00, Nick Knight wrote:
> Hello all,
>
>
>
> Just a quick note, I have been putting together a TAPI driver for
> Asterisk, this enables the user to perform things like click to dial
Ahh, the {EXTEN:1} must serve to skip the 1st character of the extension,
stripping the 9 off
That's what I needed, thanks!
-Original Message-
From: Austin M. Brower [mailto:[EMAIL PROTECTED] On Behalf Of
Austin M. Brower
Sent: Tuesday, April 13, 2004 9:11 AM
To: Nik Martin
Subject: Re: [A
Some people have some really wacky ideas about how sampled systems work :-)
Regards,
Steve
Michael Welter wrote:
Just when I thought I couldn't be wrong, I was wrong. We have
woodpeckers that drill into the arial telephone cables, and water
seeps through the holes and partially grounds the tip
Try something like this:
exten => _9NXX,1,Dial,IAX2/[EMAIL PROTECTED]/1${NPA}${EXTEN:1}
...
-brian
Nik Martin wrote:
In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd
like to do something possibly unique with the formatting of extensions in my
dial plan, and am having
[EMAIL PROTECTED] wrote:
> Another observation of something which doesn't work:
>
> exten => 3200,1,Dial(SIP/3200,20,tTr)
> exten => 3200,2,Playback(tt-weasels)
> exten => 3200,3,Hangup
> exten => 3200,102,Dial(SIP/3201,20,tTr)
> exten => 3200,103,Playback(tt-weasels)
> exten => 3200,104,Hangup
>
have to dial the entire number, like 1 + area code + number. I'd like to
eliminate this by having the user just dial 9 + 7 digit number, and have
asterisk put the 1 + area code (which is in a variable in
extensions.conf)
in front of it prior to sending the request to Voice Pulse. Is this
possib
Stephen Davies wrote:
Hi Alex,
Indeed the call end termination doesn't work on an NTL line. I'm not
so sure it works too well on other lines either.
I did some work a while back to add detection of the UK busy/hangup
signal on the line, but I never got it working well enough to depend
on it. Th
On Tue, 13 Apr 2004 [EMAIL PROTECTED] wrote:
> I have no reason to doubt what you wrote, so I already
> changed the timing parameter for my system ;). I did have
> it set as span=1,0,0, ... Now, please, in what scenario
> would one select option '0' and win what scenario would one
> use option
Scott Laird wrote:
> Since the system clock ticks at 1 kHz in 2.6, is there any reason why
> it can't be used (more or less) directly for timing in 2.6? That'd be
> a lot easier then hooking into a 1 kHz USB interrupt source.
Would someone who is familiar with the 2.6 series kernel please comment
I have two grandstream budtetone-100 and cisco 7960g phones. When I talk via speaker
phone on either of the phones I get a lot of echo. Any suggestions? Also how do I turn
on the mark echo canceller.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [E
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
From what I've been able to guess at Telstra sends a short ~50ms chirp
to the phone, the caller id and th
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch cleanly
against current CVS...
http://b
Hello Pertti,
we would be interessted to, if you could send further informations...
Thanks
Regards
Felix Deierlein
[EMAIL PROTECTED]
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Pertti
Pikkarainen
Gesendet: Samstag, 10. April 2004 11:26
An
Hi Mike-
It sounds like, from the discussion here, that your setup is already
correct. Must be something else causing the occasional red alarm! Should
not occur...
Cheers
Scott
-Original Message-
From: Mike Sturdee [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 7:08 AM
To: S
I had exactly the same question. There is some really useful documentation
on voip-info.org regarding the extensions.conf syntax.
-Original Message-
From: Brian Cuthie [mailto:[EMAIL PROTECTED]
Sent: 13 April 2004 15:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Dial Plan For
> -Original Message-
> From: Gregory Junker
>
> On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote:
> > At this point, I'm using straight Asterisk, with a a PSTN
> gateway at a data
> > POP passing calls via IAX to my PBX here in the office.
>
> Who is the PSTN gateway provider?
>
G'day Adam,
This drove me nuts for a few days just recently (only fixed it yesterday
in fact, and I've not had a chance to update any doco anywhere yet).
On Wed, 14 Apr 2004, Adam Goryachev wrote:
> Actually, now that I look at the file again, I can also see:
> Line: 80
> /* Typically, how man
That makes sense. I've always found the following wording (from the sample
file) confusing, not clear about whether "this" and "this span" referred to
the span connection on the card, or the span itself:
# The timing parameter determines the selection of primary, secondary, and
# so on sync sourc
On Tue, 13 Apr 2004, Andrew Thompson wrote:
> Two things:
>
> 1) Have you looked at call queue's?
>
> 2) I think you should have been looking at incominglimit, not outgoinglimit,
> or possibly both of them together in some combination.
>
In response to [EMAIL PROTECTED], who wrote:
> >
> > The
Andrew Thompson wrote:
[EMAIL PROTECTED] wrote:
Another observation of something which doesn't work:
exten => 3200,1,Dial(SIP/3200,20,tTr)
exten => 3200,2,Playback(tt-weasels)
exten => 3200,3,Hangup
exten => 3200,102,Dial(SIP/3201,20,tTr)
exten => 3200,103,Playback(tt-weasels)
exten => 3200,10
Good Morning,
I'm working with a queue at the moment and I having trouble with my
digits.
Australia is my example.
On the tiki it says for international digits, I can dump them in the
"digits/au" directory.
I tried that -- just because, I also made a copy in "au/digits".
When the queue announc
Hi List,
I have asterisk running on my server and work with 2 cisco ata und 1x
snom device. I can intern call it´s fine. But wenn i make a extern call,
I have many quality troubles. The extern user hear me good, but I hear
him bad (robotics). I work with SIP an ALAW protocol.
Where can i look thi
If you're reading the digest of the Asterisk-users mailing list:
* Please always strip the parts of the message you're not replying to
- do not resend the whole digest!
* Please always change the subject so it reflects your message
- Do not send a message with a subject of
RE: Asterisk-Use
Alessio Focardi wrote:
HI,
quick and simple question: is it possible to use inband dtmf with g729?
What I would like to do is to have sip clients connected to asterisk and a zaptel
card to make pstn phone calls.
My concern is to allow sip users to use digits for call destinations that
do req
Duane wrote:
Adam Goryachev wrote:
I am in Australia, which I think expects callerid at a different time to
other countries Although other people have told me callerid is
working correctly for them
Forgot to mention there is a patch for this, but it won't patch
cleanly against current
On Mon, 12 Apr 2004, Dragan Mickovic waxed:
> Is it possible for asterisk to do an sql query in order to
> get the member list of a call queue?
No, you will have to write code besides SQL in order to do
it. To go the C route, try modifiying app_queue. To use a
different language, you could code
According to voip-info.org,
"3 way calling: Normally implemented by the phone"
I am using a Grand Stream 100 and not able to make this work. I can dial out
to 1st number then with the flash button I am able to dial out again to a
2nd number. I am not able to bind them together into 1 conversation
On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
> In other words can I receive information which we are usually getting in CDRs
> during the time when the call is still active?
Yes, via the manager interface. Check manager.conf, it
lets * talk on port 5038.
--Chris
--
Chris Maj, Rochester
cma
Stephen Davies [EMAIL PROTECTED] wrote:
> > Has anybody got any experience using an X100P on an NTL phone line in
> > the UK (I'm in an ex Cable & Wireless area if that makes any
> > difference).
> >
> Indeed the call end termination doesn't work on an NTL line. I'm not
> so sure it works too we
Since * does video over sip has anyone tried to configure voicemail2 to be able to
leave a video message?
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Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need this functionality before I can migrate to * completely.
2) Are there currently any problems with inbou
Hello:
I have a ATA 186 and a FXS => FXO converter so I will like to program a
extension that can be dialed and it will dial the ATA extention #, wait for
dial tone and then dial the phone number.
Thanks
Erick
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On Fri, 2004-04-09 at 02:56, Alessio Focardi wrote:
> HI,
>
> quick and simple question: is it possible to use inband dtmf with g729?
Absolutely not.
> What I would like to do is to have sip clients connected to asterisk and a zaptel
> card to make pstn phone calls.
>
> My concern is to allow
Assuming that it is a Welltech gw, the setting is peer-to-peer mode.
Jorge
John Bittner wrote:
Hi,
Anyone get the Immix C3-FXO Sip gateway to work with asterisk. I have it
working for outbound calls but cant get it to work for inbound calls. The
unit has an built-in greeting and it keeps picking
Il 17:28, martedì 13 aprile 2004, Robert Siedl ha scritto:
> Hi List,
>
> I have asterisk running on my server and work with 2 cisco ata und 1x
> snom device. I can intern call it´s fine. But wenn i make a extern call,
> I have many quality troubles. The extern user hear me good, but I hear
> him b
The GS phones do not currently support conferencing on the phones using
the conference button. You'll probably have better luck setting up a
conference room, help with which I'm absolutely worthless... The
on-phone conferencing should be addressed in a future GS firmware
revision.
HTH,
Ryan
On
The lastest CVS's versions (both stable and head), the 'r' option in
app_dial doesn't work with SIP and Re-invites. I've heard reports that it's
not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and
I am doing re-invites, and it's worked up till this point.. What's going on?
That would be a valid configuration, and yes yellow is an option for setting a yellow
alarm when no channels are open.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Scott Stingel
Sent: Tuesday, April 13, 2004 10:57 AM
To: [EMAIL PROTECTED]
Subject: RE: [
> 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
> I have had a terrible time getting a hold of anyone over there, and I
> need this functionality before I can migrate to * completely.
Works just fine for me. Don't send in-band DTMF if you're not using the
alaw/ulaw/sli
Hi,
i install the Asterisk PBX on a linux machine with i4l to connect to PSTN
(EuroISDN). And i configure a very simple dial plan in extension.conf.
After this, i connect with a SIP program to asterisk and would call my
cellular phone, but got this error:
-- Executing Ringing("SIP/ACzerniak
Erick Weber V. wrote:
> Hello:
>
> I have a ATA 186 and a FXS => FXO converter so I will like to program
> a extension that can be dialed and it will dial the ATA extention #,
> wait for dial tone and then dial the phone number.
Unfortunately I don't believe there is a concept of "wait for dia
I am confused as well. They also made it clear that the contract terms
included recouping the cost of the ConnectReach for them, so I doubt
that TWTC is offering it at no extra cost. My contention with that, of
course, is, "why not take my ConnectReach and give it to someone else...
you already hav
I have 2 asterisk systems connected with an iax2 trunk. The first has
SIP phones and x100 line cards, the second at a remote location has a
TDM with zap extensions. When calls are parked by the zap extensions at
the second system, the calls are parked on the second system so users at
the first serv
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need this functionality before I can migrate to * completely.
2) Are there currently a
Andrew
Thanks for your answer
I'll test this conf an I'll post it so you know if it works
Thanks
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, April 13, 2004 12:48 PM
Subject: RE: [Asterisk-Users] FXS => FXO Converter Pro
Andrew:
It didn't work, the problem is that * stays on priority 1 until you hangup
and the it pass to priority 2 so what I think is that it has to be all in
the priority 1 line
Hope we can figure it out
Erick
- Original Message -
From: "Andrew Thompson" <[EMAIL PROTECTED]>
To: <[EMAIL P
Scott Laird wrote:
There shouldn't be much that needs tuned, unless your network is
overloaded and dropping packets. If that's happening, then you're
going to need to dig in and take a look at QoS on Linux *and* on your
switches and routers, but odds are that won't be a problem on most
LANs.
Hi,
I just upgraded to the recent CVS, and IAX1 no longer seems to be available.
Is there a way to reenable it?
Tor
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> 2) I think you should have been looking at incominglimit,
> not outgoinglimit, or possibly both of them together in
> some combination.
>
Another perspective issue. Apparantly 'incoming' means into
the [*] box, and outgoing is leaving the [*]. In any case,
I tried both, but 'outgoing' is co
> Another thing to try is to disable call waiting on the
> [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing
> what you've asked it to)...
>
Yep, except on the Polycom, we have found no way to disable
call-waiting.
WW
Willy Wouters
ypOne Publishing
___
Tor Houghton wrote:
Hi,
I just upgraded to the recent CVS, and IAX1 no longer seems to be available.
Is there a way to reenable it?
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
>
> Use IAX2, it is a better IAX protocol.
>
>
> Jeremy McNamara
>
>
> P.S. If you really must have it, dig thru the channels/Makefile, but
> there is zero reason to use it any longer.
>
Well, I use IAX1 between the clients on
Hi!
> On Tue, 13 Apr 2004, Dmitry Mishchenko waxed:
>
> > In other words can I receive information which we are usually getting in CDRs
> > during the time when the call is still active?
>
> Yes, via the manager interface. Check manager.conf, it
> lets * talk on port 5038.
The other option is
Isaac:
The DID is in Ocala, FL. I am not sure if it is a new market or
not. I have not heard anything from their support folks either, but I
just checked the line again and it is working. I did nothing to fix it.
I just don't understand. If you don't mind give yours a try again and
let
On Tue, 2004-04-13 at 16:11, Benjamin Wakefield wrote:
> On the tiki it says for international digits, I can dump them in the
> "digits/au" directory.
> I tried that -- just because, I also made a copy in "au/digits".
> When the queue announces the position I it says:
> -- Started music on hold
Tor Houghton wrote:
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any longer.
Well, I use IAX1 between the clien
On Tue, 2004-04-13 at 20:13, Tor Houghton wrote:
> Well, I use IAX1 between the clients on the inside of the NAT to my local
> Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
> Previously (I have not tried yet with current version), when both clients
> and Asterisk used IAX2,
It works now! I did nothing on my end either. VP must monitor this list.
Isaac
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need
Hello all,
Just a quick note, I have been putting together a TAPI driver for
Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment it
is very basic and can only perform click to dial but further
functionality w
Very cool. I am just glad they got it fixed.
-Original Message-
From: Isaac McDonald [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 13, 2004 3:56 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VoicePulse Connect Problems
It works now! I did nothing on my end either. VP must mon
Mine, too, are fixed...I was in much the same boat as the original
poster...an old DID in 212 worked with DTMF, two much newer ones in 213
and 818 (new markets, apparently) didn't until this morning.
On Tue, 13 Apr 2004 16:02:37 -0400, "Robert Jackson"
<[EMAIL PROTECTED]> said:
> Very cool. I am
Tor Houghton wrote:
Well, I use IAX1 between the clients on the inside of the NAT to my local
Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
Previously (I have not tried yet with current version), when both clients
and Asterisk used IAX2, the clients would communicate direct
Has anyone had any luck with voice recognisation using sphinx, if yes then
could u please send some pointers.
does the eagi app for sphinx really work cause i'v tierd it and sphix dosent
seem to do anything
--
regards
Vikram (http://www.vicramresearch.com)
__
On Tue, 13 Apr 2004, Eric Wieling wrote:
> Tor Houghton wrote:
>
> > Well, I use IAX1 between the clients on the inside of the NAT to my local
> > Asterisk, and IAX2 between the local Asterisk and my remote Asterisk.
> > Previously (I have not tried yet with current version), when both clients
Since Asterisk 1.0 will be released soon I am wondering if Digium runs
CVS stable on IAXtel and Digium's own PBX. If they are, then great! It
will get a good workout. If not, then WHY? A great way for a product
to get bugs fixed are for the group that codes the product to run it in
a production
Yeah, tried this.
Seems that the Zapateller code is not written correctly.
The problem is that if one does this
exten => s,1,Zapateller(answer|nocallerid)
Then the call is answered by Zapateller regardless of the callerID state.
The tones are played if there is no caller id. The problem with th
On Tue, Apr 13, 2004 at 04:58:19PM -0400, James Golovich wrote:
>
> # If you really want IAX1 uncomment the following, but it is
> # unmaintained
> #
> #CHANNEL_LIBS+=chan_iax.so
>
Thanks all, I'll move to IAX2 after I've tested the notransfer option.
Tor
__
They just updated their software and that seems to have resolved the DTMF
issues, at least for me.
Chris.
On Tue, 13 Apr 2004, Andrew Kohlsmith wrote:
> > 1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
> > I have had a terrible time getting a hold of anyone over there,
Erick Weber V. wrote:
I'll Like to now how to insert a pause on a SIP string. I have a ATA 186 and
a FXS => FXO converter so I will like to program a extension that can be
dialed and it will dial the ATA extention, wait for dial tone and then dial
the phone number.
You cannot put pauses in any dia
Could you post a link?
Thanks!
On Apr 13, 2004, at 2:59 PM, Nick Knight wrote:
Just a quick note, I have been putting together a TAPI driver for
Asterisk, this enables the user to perform things like click to dial
from any TAPI enabled app (such as outlook or ACT etc). At the moment
it
is very
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Thanks
Sean Garland, MCP+I, A+
Siskiyou Technology Consultants
205 N. Mt. Shasta Blvd. Suite 100
Mt. Shasta, CA 96067
Brian Cuthie wrote:
Tor Houghton wrote:
On Tue, Apr 13, 2004 at 02:56:48PM -0400, Jeremy McNamara wrote:
Use IAX2, it is a better IAX protocol.
Jeremy McNamara
P.S. If you really must have it, dig thru the channels/Makefile, but
there is zero reason to use it any longer.
Well, I use
FYI
http://www.itbusiness.ca/index.asp?theaction=61&sid=55298
did not like this by line in the story
t"he CRTC has said it will likely regulate voice over IP the same as other
phone services."
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lis
Sean Garland wrote:
In almost all my calls now, I am getting beeps and loud and soft parts
of a conversation. It is getting very irritating. Has anyone had this
happen? How do I get rid of it?
Set relaxdtmf=no in /etc/asterisk/zapata.conf.
___
Asteri
they actually send the caller-id info after the SECOND ring.
Now of course if the au indications were changed to combine the first
and second ring to appear as one ring, no other changes would be needed
??
Gary
On Wed, 14 Apr 2004 00:36:16 +1000, Duane wrote:
>Adam Goryachev wrote:
>> I am in A
> FYI
> http://www.itbusiness.ca/index.asp?theaction=61&sid=55298
>
> did not like this by line in the story
> t"he CRTC has said it will likely regulate voice over IP the same as other
> phone services."
>
I noticed exactly the same thing and thought the same when I read it earlier.
CRTC knows
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