I may have found a solution to my zap dialing problem: disabling the
MMX option in the zaptel drivers. After disabling MMX and recompiling
(I had dome some recompiles earlier to adjust tone length and such,
with the problem still appearing), it's been over 36 hours since the
last no-dial
Hi
Il sab, 2004-07-10 alle 05:43, Kevin Walsh ha scritto:
Joe Baptista [EMAIL PROTECTED] wrote:
Is there alist of codecs asterisk actually has per version number - i.e.
0.7, 0.9 etc?
if you have it installed, do show translation on the cli
and you'll see all codecs supported, along with
I got the everyone's busy at this moment too , I have one hfc isdn
card and one x100p , it was because of me using one channel of my isdn
for internet , so i just switched to second channel and it dialed out
fine , you need to edit this in extensions.conf for outgoing dial out.
You can also solved
Has anyone tried to run Asterisk connected to a Central Office using an ISDN
National ISDN 3 PRI trunk?
George Pajari
netVOICE communications
www.netvoice.ca
www.ip-centrex.ca
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Hello
On 10/07/2004, at 5:45 AM, Soren Rathje wrote:
Based on extensions.conf.sample from CVS-HEAD...
Thank you very much for this information ; for some reason I can't seem
to get other version newer than June,29th
As a side note, it's pretty
Jean-Yves Avenard wrote:
There's just what thing I can't figure out.
What is the action for s-.
It's the better safe than sorry option... :-)
Basically it's a wildcard option, anything beginning with s- will go there...
-- Soren
___
On 9 Jul 2004 at 14:08, Chris Shaw wrote:
Thx Jay, I hope this is not a too FAQ... I really did try to look it up
first but I saw s many conflicting things about timing... one person
says no you absolutely do not need ztdummy or a digium card to make
IVR/Voicemail work, others say you
Hello,
We have been using a Diva 4BRI with our Asterisk PBX through the capi
interface for almost a year now with good results. However, recently we
started to hear heavy clicking sounds in our phones when two
simultaneous incoming calls are processed by the card. The clicking does
not originate
Hi,
my setup:
Client: Win/linux client running x-lite or linphone
Server: debian running asterisk
on connect, incomming works well but outgoing to POTS has a lot of bad
sound (no, the mic is ok, using logitec usb headset). to ensure proper
work, tried normal p2p, worx well
the sound is nearly
I've had the same idea about using SendeText for call
call progress information.
Unfortunately, snoms currently do not support SIP MESSAGEs.
I have recently contacted their excellent support about this.
They said that they would consider it.
regards
martin
Brady Alleman wrote:
Is there a list of
Hello Asterisk Users,
is there a good german site for
asterisk?
Moritz
Brent,
Best think through what you're trying to do here. You have multiple
choices on how to interface * to the traditional pstn world, including
the x100p, tdm cards, multiple T1 interface types, external gateways,
etc.
The x100p and tdm cards have an internal 2-wire to 4-wire hybrid that
is
Hi,
I've just added an X101P FXO card to my asterisk machine. The card is
detected ok but zttool always shows it in a RED alarm state, which I
understand means that it doesn't detect the line.
I've connected the card via the line socket to my analogue line using
the cable that came with the card
Rich Adamson wrote:
[...]
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo can in the first place (and
Hi Rich,
Thanks for your heads up. See comments below.
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Rich Adamson
Best think through what you're trying to do here. You have multiple
choices on how to interface * to the traditional pstn world, including
How I can do to use the g.726 on asterisk ?
Miguel
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On Saturday 10 July 2004 11:21, Rich Adamson wrote:
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo
On Fri, 2004-07-09 at 13:55, [EMAIL PROTECTED] wrote:
To me it's a error if I can't complete calls using the ATA configured to use
the g726 codec.
I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
received NOTICES and WARNINGS, but I can't complete a call.
It looks to
You may need the 3-wire (UK) to 2-wire (US) adapters for the jack the
X100P is plugged into. I've heard you can get them most anywhere.
On Sat, 2004-07-10 at 09:40, Richard Airlie wrote:
Hi,
I've just added an X101P FXO card to my asterisk machine. The card is
detected ok but zttool always
Andrew Kohlsmith wrote:
On Saturday 10 July 2004 11:21, Rich Adamson wrote:
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1
On Sat, 2004-07-10 at 10:11, [EMAIL PROTECTED] wrote:
How I can do to use the g.726 on asterisk ?
Use Asterisk CVS -head.
http://www.asterisk.org/index.php?menu=download
--Eric
--
Useful Asterisk Docs (BOOKMARK THEM!):
http://www.digium.com/index.php?menu=documentation (look at the
Unofficial
We have a strange echo problem. Maybe echo isn't the correct term.
When we make a call f/ a SIP phone (we have several 7960's, some
3coms, and I've even tried a softphone, all on the same 100BaseTX
network) to the pstn, if the person I'm calling has a PRI or
channelized T1 f/ Bell, then the
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo can in the first place (and
they do exist).
Just
Not from me. I think the more books the better. I'm looking forward to
getting my copy.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
Steve Underwood wrote:
Andrew Kohlsmith wrote:
On Saturday 10 July 2004 11:21, Rich Adamson wrote:
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you
Beierlein Moritz wrote:
Hello Asterisk Users,
is there a good german site for asterisk?
Moritz
Hi Moritz,
there is * dicussion group at the German IP-Phone forum:
http://www.ip-phone-forum.de/
http://www.ip-phone-forum.de/forum/viewforum.php?f=24
jo
Richard Airlie [EMAIL PROTECTED] wrote:
I've just added an X101P FXO card to my asterisk machine. The card is
detected ok but zttool always shows it in a RED alarm state, which I
understand means that it doesn't detect the line.
I've connected the card via the line socket to my analogue line
Eric Wieling [EMAIL PROTECTED] wrote:
You may need the 3-wire (UK) to 2-wire (US) adapters for the jack the
X100P is plugged into. I've heard you can get them most anywhere.
No - steal the RJ-BT cord off the back of a modem if the X100P didn't
come with a BT plug. It'll work without any
Das is aber schöen!
Paul von Wachter Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jo
Sent:
[Please excuse if this is a repeat; I initially tried to send it from a
different account, and it's been held up for a couple of days awaiting
moderation.]
1) What's the absolute minimum required (hardware-wise) in order to get one
in-bound POTS line into Asterisk, and then have IP phones
Remarque : message transféré en pièce jointe.
Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage !
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Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !Téléchargez Yahoo!
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Hello everybody,
Can someone help me to find the right elements and the
right configuration to send the CDRs of Asterisk to a
freeradius servers?
I found some help from the page
http://bugs.digium.com/bug_view_page.php?bug_id=0001193
but It didnt work for me... should I change something
in
Ken D'Ambrosio wrote:
[Please excuse if this is a repeat; I initially tried to send it from a
different account, and it's been held up for a couple of days awaiting
moderation.]
1) What's the absolute minimum required (hardware-wise) in order to get one
in-bound POTS line into Asterisk, and then
1) A digium FXO card ($100) will do. Works great for me, ymmv. ~$100.
My system is running on a Celeron 2.7GGhz with 256 of RAM and the
processor never really blips. I'd say a ~1 GHz would be plenty enough
for one or two channels.
2) $10 Walmart Special connected to a $100 Sipura SPA-2000. The
I am using asterisk as a voicemail server for our IP Centrex SoftPBX.
Umar.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chad Whitten
Sent: 09 July 2004 22:46
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] using asterisk voicemail with a class 5
After trying everything under the sun to get rid of echo on my X100P,
I'm curious if anyone managed to solve the echo issues by switching to a
SPA-3000?
As well, if you have multiple SPA-3000's, can you create dial-out groups
similar to the Dial(ZAP/g1) functionality?
Thanks.
--
Mike Benoit
How can one specify that codec selection should avoid transcoding if
possible?
The reason I ask is that when enum lookup succeeds and the destination only
accepts ULAW, the various transcodings seems to garble the audio if GSM,
ILBC, etc. are allowed.
Cheers,
Rich
Dr. Murphey:
How can one specify that codec selection should avoid transcoding if
possible?
Asterisk only transcodes if the original audio stream needs to be connected
either to (a) another audio stream or (b) an internal Asterisk function --
and the original audio stream is encoded in a
1) X100P from Digium. We currently have four of them in one machine
(started with one, then added a second, then went all the way up to
four) and they're working great. Put the X100P into a cheap PC (we're
using a 1.7 Ghz Celeron system, I believe), add Linux and Asterisk,
and you have a
Rich Adamson wrote:
[...]
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo can in the first
On Fri, Jul 02, 2004 at 02:48:17PM -0700, Chris A. Icide wrote:
2) can you cross connect PRI interfaces?
in other words can you use the dacs functionality to insert a digium card
(on a system running asterisk) in between a pri from a carrier, to a legacy
pbx system?
Yes, you can. But
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
From the WIKI:
Contributions
Manager: Daniel Jimenez (cuban)
Bounty: $50 USD
Date opened: July 10, 2004
Contributors: cuban ($50)
Detail
Yes, Yes I know you could do all sorts of fun with the dialplan to
produce a
On Sat, Jul 03, 2004 at 07:44:54PM +0200, Thilo Salmon wrote:
How would you go about running, 8 or 16 say, E1s over TDMoE? Would you
create multiple dynamic spans or just one large one? How would you
assign d channels to spans, if you had just one large span?
Did any of you guys try this
http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SIP+simultaneous+registry
Updated,
Allow a SIP device to register more than once so a single extension may
exist in multiple locations.
Upped total to $75.
Daniel...
Daniel Jimenez wrote:
Hey Brent,
If you only need a small number of fxo ports, pstn gateways in the
form
of Cisco boxes, Mediatrix 1204, as well as many other products can do
that at a relatively inexpensive cost. The echo can function is built
into those boxes, which do work. Each comes with additional
If you install a T1 card and an external T1 mux (with fxo cards), the
echo can function already exists within the mux and/or cards. Don't
really need 'another' external echo can box unless you actually
purchased a T1 mux that didn't have echo can in the first place (and
they do exist).
On Wed, Jul 07, 2004 at 11:57:31AM -0400, Timothy R. McKee wrote:
This has always been one of my pet peeves, even as I worked in the industry.
A telco switch operating a DS1 on trunk side should enforce caller-id
numbers to be within the range of DID numbers assigned to that trunk. There
Use the NuFone to call numbers 1800,1866 and after 20 seconds the call
isinterrupted. In log of * it writes: Max retries exceeded you host.
Somebody can help me? Thank
you Joao Carlos Moura
You need to send a vallid CALLERID to Nufone.
On Sat, 10 Jul 2004, V59Net wrote:
Use the NuFone to call numbers 1800,1866 and after 20 seconds the call is
interrupted. In log of * it writes: Max retries exceeded you host.
Somebody can help me?
___
No you don't it will just make one up...
bkw
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, July 10, 2004 8:17 PM
Subject: Re: [Asterisk-Users] NuFone Error
You need to send a vallid CALLERID to Nufone.
On Sat, 10 Jul 2004, V59Net wrote:
Use
Hello group
I'm working on getting festival installed and working on my FC1. I ran
into a problem and after searching Google I found this message talking
about a patch for Speech Tools and Festival
http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html
The above site does not have
Hello
Use two Asterisk servers.
I registered Server2 in Server1. When I bind for an extension in the
Server1, the hard call some as and is interrupted.
My configuration:
iax.conf / Server 1
[20001]
type=friend
accountcode=20001
host=dynamic
secret=secret
context=sip
disallow=all
allow=gsm
Look in your /usr/src/asterisk/contrib directory
Hello group
I'm working on getting festival installed and working on my FC1. I ran
into a problem and after searching Google I found this message talking
about a patch for Speech Tools and Festival
On Sat, 10 Jul 2004, Brian K. West wrote:
No you don't it will just make one up...
I beg to differ, Mr Brian sir. I had problems calling 800 numbers with
Nufone, and Jeremy explained to me that they check for a caller id before
sending calls to 800 numbers.
Hi All,
Looking for a VoIP provider for a 2 site IP-PBX deployment to provide DID
numbers for each person in the offices (~75 numbers across the 2 sites) and
outbound VoIP calling. The sites will have sufficient POTS line for backup
outbound and 'main number' inbound calling. I haven't gotten
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
George Pajari
Sent: Saturday, July 10, 2004 4:58 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Avoiding transcoding
Dr. Murphey:
How can one specify that codec selection should
Hi,
T1 is the carrier. T1 provides 24 D channels of 64Kbps each.
Telephone companies provide ISDN (integrated services data network) on top
of T-carrier. Two common flavors are BRI (basic rate interface) and PRI
(Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23
usable
I'm not sure I understand what you are trying to do.
You have an administrative assistant and several other staff. You want the
administrator to be able to take calls directed to the staff extensions?
If I have the requirement right, you could accomplish this by ringing the
staff extension and
Hi Paul, you would know better than I would but I always thought a T1
was 24 channels of voice with the signalling additional like we have in
Australia a Pri or E1 is 30 channels voice channels plus signalling.
Can anyone else clarify?
Cheers,
Dean
-Original Message-
From: [EMAIL
Paul,
The question is very simple.
When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of
On Sun, 11 Jul 2004, Kannaiyan Natesan wrote:
When I call a SIP user, the phone should ring in more than one
extentions. Also more than one phone should be able to register with
asterisk. Right now it is not the case. The last phone which register will
be receiving the calls. This type of
in other SIP proxy server, this can be done easily, i mean its default
1 or more phone could be registered at 1 number (12345) and resulting same effect as u
ask
SER (SIP Express Router, http://iptel.org/ser) can deal with this
SER is a friend to asterisk, i think :), you can accept calls with
Hi Guys,
This topic has become pretty much pointless. CallerID was never designed to
be any kind of authentication scheme. Also, it is very hard for telco to
restrict proper usage of CallerID in PRI or SS7 (Please consider number
protability, etc.)
We all already agreed on fact that author of
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