Hi
I know of a product called a Parlay which does this, but
its expensive. Someone on the list said that asterisk could
do this with a quad T1 card.
I think that would be very nifty if asterisk could transfer
the isdn calls based on CLID or DNIS before the call is
actually answered.
If you get
I agree entirely! I've been told it's not possible, but I'd love for
someone to prove me/them wrong.
-d
At 11:54 PM 8/10/2004, you wrote:
Hi,
I was wondering if any CISCO users out there knows if it is possible to
Change the locations of the BUTTONS along the bottom of the screen.
I ask this as
Mike Coakley wrote:
Anyone have any ideas.
I have a 12SP+ working as a basic one line fone using chan_skinny.
Jeremy McNamara
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On Tue, 10 Aug 2004, Scott Laird wrote:
Why stop there--you can beam pre-recorded messages to phones without a
person or phone line ever being involved. You could send hundreds of
[...]
That's right. Here in Hamburg, Germany one day before our elections my
phone rang and there was a
Dan Mahoney, System Admin wrote:
You start up the phones, they register, all is good. They show up in
sip show peers like thus:
danm/danm65.125.237.91D N 255.255.255.255 5060
OK (29 ms)
We pass a few calls in and out, and asterisk deadlocks (not a true
deadlock, see
On Wed, 11 Aug 2004 [EMAIL PROTECTED] wrote:
I know of a product called a Parlay which does this, but
its expensive. Someone on the list said that asterisk could
do this with a quad T1 card.
I think that would be very nifty if asterisk could transfer
the isdn calls based on CLID or DNIS
Walt Reed wrote:
On Tue, Aug 10, 2004 at 02:12:51PM -0700, Scott Laird said:
On Aug 10, 2004, at 1:14 PM, Loek Gijben wrote:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for
Chris, While you are thinking logically, This will just as
un-effective as putting them all in the dialplan, as the DBGet() and
DBPut() functionality deals with the internal astdb (db1 database).
DBGet and DBPut work with Berkely DB 1.85.
Althought this DB185 is a little outdated, it can
hi,
kann a sip user login two times from different clients? if he can, how
does asterisk handle the call in this case?
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail: [EMAIL PROTECTED]
At 12:22 PM -0400 on 8/10/04, drodden wrote:
Anybody have any experience with blocking numbers in the U.S's Do Not
Call list?
We have a customer that will be getting their own Asterisk server from
us, and they want it to be check outbound numbers against the do not
call list; this is for a backup,
At 10:09 PM +0200 on 8/10/04, Soren Rathje wrote:
John Todd wrote:
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
[snip]
/Soren
It is the mark of an educated mind to be able to entertain a thought
without accepting it.
- Aristotle
Ok, so we moved here from *-dev, no problem... ;-)
Hi all,We use tiff version 3.5.7-2
and spandsp-0.0.1k with asterisk version 0.7.2-4 on debian unstable. Our
asterisk server receives calls and faxes from a mobile operator. We are
connected to their Cisco router which connects to our asterisk over VoIP that
goes inside a Vlan over 100Mbit fiber
[EMAIL PROTECTED] wrote:
At 12:22 PM -0400 on 8/10/04, drodden wrote:
Anybody have any experience with blocking numbers in the U.S's Do
Not Call list?
We have solution developed (based on *) to handle this scenario. Please
contact me off the list for details.
Ta
SJ
Hi,
I compiled the newest asterisk cvs and chan_oh323.
Calls coming via chan_oh323 are not listed in the cdr
file.
How can I fix this?
Roger.
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At 10:14 PM +0200 on 8/10/04, Loek Gijben wrote:
hank [EMAIL PROTECTED] wrote:
voip spam?
I have never gotten any yet.
It's is just waiting for the first one to arrive..
The mechanics are just too appealing for spam-like businesses.
Imagine a telemarketeer script that dials lists of VoIP
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 11 August 2004 05:06 am, Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
At 12:22 PM -0400 on 8/10/04, drodden wrote:
Anybody have any experience with blocking numbers in the U.S's Do
Not Call list?
We have solution developed
I am running asterisk 1.0-RC1 with zaptel 1.0-RC1 on Trustix 2.1 on with
a TDM400P with one FXS and three FXO modules. I am having intermittent
problems when I try to dial out from any of my Cisco 7960 or 7940 SIP
phones. I have my two analog lines configured in zapata.conf as follows:
Generaly, voip server like asterisk, sip proxy, or h.323 proxy should
never reach load of 1.0. Should always be less than 1.
I think if load is 2.0 than your voice quality should be bad.
Spandsp is CPU sensitive. Especialy if you will try to send faxes with
cpu load of 100% I think you will
Hi,
On Wed, Aug 11, 2004 at 09:49:29AM +0200, Thomas Kuepper wrote:
kann a sip user login two times from different clients? if he can, how
does asterisk handle the call in this case?
a single user can only do one login concurrently. But its possible to
create two accounts for the user and
Good day to
all,
Does anyone
know whether there is an option in * to save (and retrieve) massages in a
different hard disk from that of the computer running *?
Eran
Hi all,
I have one asterisk server with one ISDN BRI connection to PSTN,
with h.323 support (oh323)
I buy some voip phones, and I connect them to the same switch as asterisk
server is; all is at the same TCP network.
I need to route some extensions from my DDI (DID) line at asterisk to
some
[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 11 August 2004 05:06 am, Senad Jordanovic wrote:
[EMAIL PROTECTED] wrote:
At 12:22 PM -0400 on 8/10/04, drodden wrote:
Anybody have any experience with blocking numbers in the U.S's Do
Not Call list?
We
Hello,
I am trying to setup the call on asterisk with one user to another. But when i call from my end to any another asterisk user, the call is going to that end..but when that user receives the call, it gets HANG UP. I think this is CODEC PROBLEM. Currently i am using g723 CODEC. When i checked
John Todd wrote:
At 10:09 PM +0200 on 8/10/04, Soren Rathje wrote:
John Todd wrote:
At 7:14 PM +0200 on 8/10/04, Soren Rathje wrote:
Gang,
[snip]
/Soren
It is the mark of an educated mind to be able to entertain a
thought without accepting it.
- Aristotle
Ok, so we moved
uhh ya, I'll get a pri for home use, sounds like a smart idea
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also frastrated by this... 7905g are laid out a lot better
On Wed, 11 Aug 2004 01:07:04 -0500, denon [EMAIL PROTECTED] wrote:
I agree entirely! I've been told it's not possible, but I'd love for
someone to prove me/them wrong.
-d
At 11:54 PM 8/10/2004, you wrote:
Hi,
I was
Good day to all,
Hello,
Does anyone know whether there is an option in * to save (and
retrieve) massages in a different hard disk from that of the
computer running *?
You can just mount the /var/spool/asterisk/voicemail directory
via nfs or smb from another machine.
Stefan
Hi *,
I want start with a setup of Asterisk with a clean PC.
This PC is a SMP-Machine with two 466MHz CPUs, a Acer ISDN card and a
AVM Fritz! PCI card.
Which Kernel is better for my constellation (Asterisk with SMP, CAPI and
ZAPHFC)?
Kernel 2.6.x or Kernel 2.4.x?
Regards
Bastian
I am currently a new asterisk user and new to telephony in general. I
have been looking around to implement a solution with asterisk that has
many of the nice features of a proprietary PBX for a small office. The
features that I am looking for that I haven't been able to find any
information
hi,
i want to use mystun because off nat problems by more than one device
behind one nat gw. i think it is the only solution to solve the nat
problem.
what i do not understand is why needs the stun server two ip addresses?
thx for any hints.
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Christoph Rothe [EMAIL PROTECTED] wrote:
[...]
That's right. Here in Hamburg, Germany one day before our elections
my phone rang and there was a recording from one of the big parties
that reminded me to vote the right ones ;-(
It could of course have been a joe-job by another party...
--
On Wed, 2004-08-11 at 06:17, Jeremy Lowery wrote:
I am currently a new asterisk user and new to telephony in general. I
have been looking around to implement a solution with asterisk that has
many of the nice features of a proprietary PBX for a small office. The
features that I am looking
On Wed, 11 Aug 2004, Nilesh sonavani wrote:
I am trying to setup the call on asterisk with one user to another. But
when i call from my end to any another asterisk user, the call is going
to that end..but when that user receives the call, it gets HANG UP. I
think this is CODEC PROBLEM.
mattf [EMAIL PROTECTED] wrote:
[...]
As for speed, AGI scripts that we use on a daily basis do thousands
of searches a day through a 800,000 record table in less than a
second(on a dedicated 3.2GHz MySQL DB machine) so looking through a
million shouldn't be too bad. Asterisk will wait for the
Hi,
I have a problem with a Digium quad E1 card. It seems
when I make outgoing calls to any party, when that person talks on the line,
they hear scratching and static (theres also background static, but less
of it). The person making the call from asterisk (via the E1) doesnt
hear any
I think I'm going slightly mad.
I've got a Dell PowerEdge 400SC (Cel 2.4GHz, 256MB RAM, 40GB HDD) that I'm
using to set up a * PBX for a (very) small startup.
It's running Debian Sarge with the stock 2.4.26 kernel (I know it's still an
unstable release, but I'd need to jump through all
John Todd wrote:
and deep pockets to champion something for a monetary loss. So, Duane,
want to put your ENUM tools to good use? (see my post of a few minutes
Would be happy to if there was funding in it for the other enum
activities we're currently under taking, then of course not being based
On Wed, 2004-08-11 at 12:24, Gary Pigott wrote:
I've got a Dell PowerEdge 400SC (Cel 2.4GHz, 256MB RAM, 40GB HDD) that I'm
using to set up a * PBX for a (very) small startup.
It's running Debian Sarge with the stock 2.4.26 kernel (I know it's still an
unstable release, but I'd need to jump
I
have:
RedHat 9.0
TDM40B
asterisk-0.9.0 compiled from sources
zaptel-0.9.1 likewise
/etc/zaptel.conf contains
fxoks=1-4
loadzone = us
defaultzone=us
loaded
modules zaptel and wcfxs
/etc/askterisk/zapata.conf
contains
[channels]
language
= en
signalling
= fxo_ks
Hello,
I have installed 4 port FXO card.
I have a problem when I am calling from SIP device to ZAP channel (calling
to outside line)
First couple of seconds calls is normal but when I stop speaking to
microphone for a couple of seconds (so for a couple of seconds it is
silence) then when I start
On Wed, 11 Aug 2004 02:46:14 -0700, lists-jmhunter
[EMAIL PROTECTED] wrote:
also frastrated by this... 7905g are laid out a lot better
Irrelevant - you still need to press More to get transfer on the 7905Gs as well.
I have Smartnet on my phones - I wonder if it's worth putting in a
feature
[EMAIL PROTECTED] schrieb:
...
My question is how can I do outgoing calls?
Need I call firstly IP of asterisk and then to enter phone # to PSTN?
respectively, how can voip phone knows to which IP to connect?
Is this problem solved by gatekeeper? need I one.
...
Hi,
this depends on your H323
Although I don't think the error message would indicate something like
this... but... are you sure you have a ptp isdn line (bri_cpe versus
bri_cpe_ptmp).
Gary Pigott wrote:
I think I'm going slightly mad.
I've got a Dell PowerEdge 400SC (Cel 2.4GHz, 256MB RAM, 40GB HDD) that
I'm using to
Carlos,
As far as I can tell from pulling the phone apart I don't see an
EEPROM. There is one chip that could be an EEPROM (has a sticker on it
with the MAC address). But from the shape I believe it is a NVRAM type
chip (just a guess though). (Probably is a boot ROM though.) Could be
an
Jeremy,
I can get them working for a day or two but then they go south and
don't come back.
Thanks,
Mike
On Aug 11, 2004, at 2:23 AM, Jeremy McNamara wrote:
Mike Coakley wrote:
Anyone have any ideas.
I have a 12SP+ working as a basic one line fone using chan_skinny.
Jeremy McNamara
Yes I'm sure, although just in case, I tried both point to point and point
to multipoint modes in zapata.conf
Gary
- Original Message -
From: Michael Sandee [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 11, 2004 1:55 PM
Subject: Re: [Asterisk-Users] zaphfc problems...
Hi,
I have two zap lines into the asterisk box, however when the line is dialled
asterisk waits 3 rings before it picks the line up and deals with the call.
Is there any way of changing that? Or is it just built in?
Also with a queue is there anyway of getting it to goto voicemail if no-one
try
exten = 101,2,Dial(Zap/1,10,r)
in stead of
exten = 101,2,Dial(Zap/1,10)
BC
From: Warren Burstein [EMAIL PROTECTED]
Reply-To: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Ringing() doesn't play sound while phone is
ringing
Date: Wed, 11 Aug 2004 15:22:45 +0400
I have:
Hi,
I have a problem with a Digium quad E1 card. It seems when I make
outgoing calls to any party, when that person talks on the line, they
hear scratching and static (there's also background static, but less of
it). The person making the call from asterisk (via the E1) doesn't hear
any of
On Wed, 11 Aug 2004, Martin List-Petersen wrote:
Shouldn't it be possible to pipe the channels for the MAX through the
Asterisk box ?
The whole PRI into Asterisk and a PRI cable from a second port to the
MAX.
I haven't looked much at data calls from Zap to Zap, but it looked like
it was
On Wed, 11 Aug 2004 [EMAIL PROTECTED] wrote:
I know of a product called a Parlay which does this, but its
expensive.
Nifty - I'll take a look. Thanks!
Someone on the list said that asterisk could do this with a quad T1
card.
I think that would be very nifty if asterisk could transfer the
Under Cisco Call Manager you can create soft button templates and move them
around. Of course this is under sccp and also dynamic xml config files.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Gardiner
Sent: Wednesday, August 11, 2004 12:54 AM
Gary Pigott wrote:
I think I'm going slightly mad.
I've got a Dell PowerEdge 400SC (Cel 2.4GHz, 256MB RAM, 40GB HDD) that
I'm using to set up a * PBX for a (very) small startup.
It's running Debian Sarge with the stock 2.4.26 kernel (I know it's
still an unstable release, but I'd need to
Mike Coakley wrote:
Chris,
Actually it is a documented feature of Macro. Macro only executes
extension s there are no other extensions in the macro context. I ran
into this while working through building our dial plan. It was driving
me nutz. (But the WIKI rescued me.) What I had to do is use
Hi,
Just install 6 new polycoms at a customer and all of them have a major echo
issue. Have asterisk connected to the PSTN via digium 4 port fxo card in a
P4 running fedora.
I have tweaked zapata ran ztmonitor... just as a test I attached a cisco
7960 the cisco has no echo problems.
Hello
2.6 scheduler performs in O(1), it will perform much better in
multi-processor environment than the 2.4 series
Jean-Yves
On 11/08/2004, at 8:00 PM, Bastian Schern wrote:
Which Kernel is better for my constellation (Asterisk with SMP, CAPI
and
ZAPHFC)?
Kernel 2.6.x or Kernel 2.4.x?
---
I could be wrong, but according to the Max documentation, drop insert
only works on a channelized T1...not a PRI.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 11 Aug 2004, Nate Carlson wrote:
On Wed, 11 Aug 2004, Martin List-Petersen wrote:
Shouldn't
Michael Welter wrote:
Gary Pigott wrote:
I think I'm going slightly mad.
I've got a Dell PowerEdge 400SC (Cel 2.4GHz, 256MB RAM, 40GB HDD) that
I'm using to set up a * PBX for a (very) small startup.
It's running Debian Sarge with the stock 2.4.26 kernel (I know it's still
an unstable
We're using a TE410P connected directly to Kingston Telecom (C7 -
EuroISDN conversion is done along the route to us).
Unsure what to change within my setup really. I've played around with
the rxgain and txgain, although it's made some difference, nothing
special!
Does anyone know why I would get
hi,
wenn i do a call from gsm to sip endpoint i dont see the gsm telephone
nummber. i can only see UNAVAILABLE.
how can i change this?
--
Thomas Küpper
01063 Telecom GmbH Co. KG
Mottmannstr. 2
53842 Troisdorf
Telefon: 02241-9434-506
Telefax: 02241-9434-846
E-Mail: [EMAIL PROTECTED]
E-Mail:
OK... Now I'm lost. I have no more Cisco 12sp+ or 30VIP that work. So
anyone with any ideas would be appreciated.
Thanks,
Mike
On Aug 10, 2004, at 11:41 PM, Mike Coakley wrote:
I've searched high and lo and googled to I can't google no more... I
knew that Cisco bought Selsius to get their VoIP
I have two questions with using asterisk remote unix connections.
I'm running asterisk as asterisk -gc
Then I have two -r sessions... one just asterisk -r, and another that runs as
watch 'asterisk -rx iax2 show channels'
... so I have two remote sessions going and no verbosity.
First one...
On Wed, 11 Aug 2004, Ben Merrills wrote:
We're using a TE410P connected directly to Kingston Telecom (C7 -
EuroISDN conversion is done along the route to us).
Unsure what to change within my setup really. I've played around with
the rxgain and txgain, although it's made some difference,
Ryan Ayers wrote:
Yes, I read that. It is not in there. It does mention setting the
Auto-Answer for the phone. However, I want an intercom, I don't want
a door
phone. The Auto-Answer feature just sets it so it answers all calls
automatically.
Okay, I finally got the intercom working.
Can't you stop quoting all of the previous text?
:-)
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Does anything have to be done at compile time in order for Asterisk to
take advantage of 2 CPU's?
Thanks
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Or atleast reply above the original, so that you don't have to scroll
down. ;)
Holger Schurig wrote:
Can't you stop quoting all of the previous text?
:-)
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You should post this on the wiki with the agi. I am sure there are allot of
people that would love this patch.
- Original Message -
From: Ryan Ayers [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, August 11, 2004 10:52 AM
Subject: Re: [Asterisk-Users] Snom Intercom
Ryan Ayers
That did it. I thought Ringing() did that, but I guess it's just for when
you want to fake a ringing tone. I'll add a comment to
http://www.voip-info.org/wiki-Asterisk+cmd+Ringing.
thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bart Coppens
On Wed, 11 Aug 2004, Bruce Komito wrote:
I could be wrong, but according to the Max documentation, drop insert
only works on a channelized T1...not a PRI.
The old Max 4000 books I have don't say squat about what it will and won't
do. :( Do you happen to have a link to decent docs about this?
I tried implementing my * and it didn't pass the spouse factor at this
time. I wanted to hook it up for outbound only at this point to get a
better handle on the dial plans and the echo problem.
I thought this might have been done before as a natural part of testing
- but maybe not.
In wcfxo.c I
Jean-Yves Avenard schrieb:
Hello
2.6 scheduler performs in O(1), it will perform much better in
multi-processor environment than the 2.4 series
That's one thing, but what is with the compatibility?
CAPI?
ZapHFC?
And so on.
Regards
Bastian
___
Why hack the code for this? Just implement a wait() in your dialplan.
That way you can switch back and forth between outbound-only and in/out
by just changing the wait(120) to wait(1).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Cook
Sent:
On Aug 11, 2004, at 9:45 AM, Christopher L. Wade wrote:
Mike Coakley wrote:
Chris,
Actually it is a documented feature of Macro. Macro only executes
extension s there are no other extensions in the macro context. I
ran into this while working through building our dial plan. It was
driving me
Mike Coakley wrote:
Anyone out there with * code experience that can put this question to bed.
Actually, it is in bed. Any 'user defined' extension will work inside a
macro. What won't work is any 'system' extension whose execution is
based on an 'exception', which AbsoluteTimeout and the 'T'
If less than a second implies best part of a second, then that's a
bit slow, although probably still good enough for this application.
(But I personally avoid MySQL for security reasons.)
how is SQL insecure?
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Gabriel Millerd [EMAIL PROTECTED] wrote:
[...]
If less than a second implies best part of a second, then
that's a bit slow, although probably still good enough for this
application. (But I personally avoid MySQL for security reasons.)
how is SQL insecure?
SQL itself is just a standard for
On Wed, 2004-08-11 at 15:47, John Bittner wrote:
Hi,
Just install 6 new polycoms at a customer and all of them have a major echo
issue. Have asterisk connected to the PSTN via digium 4 port fxo card in a
P4 running fedora.
I have tweaked zapata ran ztmonitor... just as a test I
Ben Merrills wrote:
We're using a TE410P connected directly to Kingston Telecom (C7 -
EuroISDN conversion is done along the route to us).
Unsure what to change within my setup really. I've played around with
the rxgain and txgain, although it's made some difference, nothing
special!
Does anyone
On Aug 11, 2004, at 12:04 PM, Christopher L. Wade wrote:
Mike Coakley wrote:
Anyone out there with * code experience that can put this question to
bed.
Actually, it is in bed. Any 'user defined' extension will work inside
a macro. What won't work is any 'system' extension whose execution is
Thomas Kuepper wrote:
hi,
i want to use mystun because off nat problems by more than one device
behind one nat gw. i think it is the only solution to solve the nat
problem.
what i do not understand is why needs the stun server two ip addresses?
It needs 2 IPs because the server will attempt to
David Cook wrote:
I tried implementing my * and it didn't pass the spouse factor at this
time. I wanted to hook it up for outbound only at this point to get a
better handle on the dial plans and the echo problem.
I thought this might have been done before as a natural part of
testing - but
That is a matter of opinion and not in any way factual SQL, just as
everything else, is as secure as YOU make it... As you said, it's a language
for querying relational databases, it has no knowledge of security. That's
what firewalls, encryption and strong passwords are for...
However, for
I'm using vanilla bri-stuff-0.1.0-RC2k from www.junghanns.net and
everything looks right. There are no errors or warnings during startup.
It seems to work correctly (well SIP and IAX2 anyway) until I try to
dial out over the ISDN line.
I get the following:
*CLI -- Executing
If I call a number from my mobile say, it sounds fine! Nothing is wrong
with the call quality at all. If I call asterisk (via the digium card)
then route that call out to another mobile, that sounds just as bad as
making the call from asterisk...
So, to cap off,
OK: Mobile - EuroISDN - Asterisk
Hi,
At my house, I have two POTS lines. Both are connected to my * server
on a TDM400P card. As an example, say the phone numbers are
(919)555-1212 and (919)555-1213. I also have four SIP extensions, an
ATA with a fax machine, and a DID coming in from an ITSP.
I have an autoattendant
Mike Coakley wrote:
Hmm... I couldn't get my Macro to work and your comments have me
thinking I'm stupid again. I think I'm going to setup another test bed
and wack the crap out of this one... next question to ask the
developers... why wouldn't they include the system extensions. That
would
John,
I have a dozen polycom ip 600 phones that are connected to the PSTN with
two, digium 4 port fxo cards. I also had a lot of near side echo, but I
do not believe that it came from the phones. I was able to get rid of it
with these settings in my zapata.conf:
echocancel=yes
On Wednesday 11 August 2004 12:32, Chris Shaw wrote:
However, for the purpose of blocking numbers based on a do-not-call list,
it will work perfectly fine. It's lightweight, fast and relatively
efficient...
If that's all you're using it for you may as well use GDBM; no need for a
full-out SQL
Chris Shaw wrote:
That is a matter of opinion and not in any way factual SQL, just as
everything else, is as secure as YOU make it... As you said, it's a language
for querying relational databases, it has no knowledge of security. That's
what firewalls, encryption and strong passwords are
Hi all,
I've been trying to wrap my mind around this one for several days now.
How can I 'debug' the CallerID reception on a Zap/POTS channel? I have
a POTS line with CallerID and a Digium TDM11B card right now. I have my
signalling set to ks for both sides, can make and receive calls just
Title: [Asterisk-Users] a few question about asterisk
I am currently a new asterisk user I have worked with the old rolm systems in the past. I have been asked to look around and find out how to do a few things in asterisk, either in asterisk itself or with third party software. The features
I have an Asterisk box running happily under Fedora Core 2 with a X100P and a
TDM400P, and now I'd like to integrate it to my ISDN2e connection using
either an AVM Fritz PCI card or an Eicon DIVA passive card, both of which I
have sculling around. I've successfully used the AVM card under
No, there is no echo canceling on the handset. I wasted alot of time on
this before I got the following answer from Polycom:
-
John
it turns out that we do not support Handset AEC at this time. This is
something we plan to support in the
On Wed, 2004-08-11 at 11:32, Chris Shaw wrote:
That is a matter of opinion and not in any way factual SQL, just as
everything else, is as secure as YOU make it... As you said, it's a language
for querying relational databases, it has no knowledge of security. That's
what firewalls,
On Wed, 11 Aug 2004, Olle E. Johansson wrote:
Dan Mahoney, System Admin wrote:
You start up the phones, they register, all is good. They show up in sip
show peers like thus:
danm/danm65.125.237.91D N 255.255.255.255 5060 OK
(29 ms)
We pass a few calls in and out, and
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On Wed, 2004-08-11 at 12:24, Brassfield, Anthony wrote:
I am currently a new asterisk user I have worked with the old rolm
systems in the past. I have been asked to look around and find out how
to do a few things in asterisk, either in asterisk itself or
I have an asterisk server running on Redhat 8.0 with a Digium TDM400P
w/4 FXO modules (TDM04P)
There are 2 lines going into the Digium card. One line is a Vonage
digital line, and the other line is a Comcast voice line. I have a SIP
Grandstream 100 phone connected to the Asterisk server.
The
Ben Merrills wrote:
If I call a number from my mobile say, it sounds fine! Nothing is wrong
with the call quality at all. If I call asterisk (via the digium card)
then route that call out to another mobile, that sounds just as bad as
making the call from asterisk...
So, to cap off,
OK: Mobile -
I have an asterisk server running on Redhat 8.0 with a Digium TDM400P
w/4 FXO modules (TDM04P)
There are 2 lines going into the Digium card. One line is a Vonage
digital line, and the other line is a Comcast voice line. I have a SIP
Grandstream 100 phone connected to the Asterisk server.
If a
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