I downloaded asterisk-sounds-1.2.0-beta1,
superused, then typed "make install". The installation stopped with
the following error:
No description for sounds/access-code.gsmmake:
*** [datafiles] Error 1
Does anyone have any useful tips? I'm running
Debian 3.0.
Thanks, WILL
Hi all!
How do I create a dialpattern in an outgoing rout
that sends all calls starting with 00 AND calls starting with 1-9 to same
trunk
Regards
Anders Svensson
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I wrote a very very simple shell script and an even simplier macro to
use the IBM TTS engine within asterisk for prompts. While its free you
are limited on the number of requests you can do within a day.
If anyone is interested its available at
To read a database entry from the asterisk database using the new DB
function the syntax is this:
Set(DB(Family/Key))
And to write to the database:
Set(DB(Family/Key)=Value)
BUT how do I delete a database entry?
Regards
Thorben
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Hiyall,
been following this for a while, just thought I would add a bit to the
debate, but doesn't the Cisco system (Call Manager?) run on an Windows
2000 based server - if it was that bad why would Cisco choose to run it?
Also 3Com use NT/2000 to run the H323 gateway. Admittedly the call
Hi!
Cant get my Grandstream GXP 2000 to register on my
AAH. All other Grandstream units work fine. Something extra to think about?
Regards
Anders Svensson
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On Sun, 2005-10-02 at 10:21 +0100, Wayne wrote:
Hiyall,
been following this for a while, just thought I would add a bit to the
debate, but doesn't the Cisco system (Call Manager?) run on an Windows
2000 based server - if it was that bad why would Cisco choose to run it?
Politics and
Cisco seem to be moving their CCM users to Linux. At least I have heard
of a few users going that way, after Cisco recommended it.
CCM doesn't usually handle anything near to hard real-time, so it is a
lot less demanding than something like Asterisk.
Regards,
Steve
Wayne wrote:
Hiyall,
Cisco Call Manager does indeed run on Windows 2000.
There are positive and negative facets with this arrangement.
Postive:
- Easier for your average IT engineer to install
- Easier for the same person to maintain
- Using MS SQL Server allows for replication and
On Sun, 2005-10-02 at 10:21 +0100, Wayne wrote:
Hiyall,
been following this for a while, just thought I would add a bit to
the
debate, but doesn't the Cisco system (Call Manager?) run on an
Windows
2000 based server - if it was that bad why would Cisco choose to run
it?
Politics and
Steve wrote:
Cisco seem to be moving their CCM users to Linux. At least I have
heard
of a few users going that way, after Cisco recommended it.
There have been unofficial statements that CCM would move to a
Unix-like OS, but that would be in the next major release, still
some time off. Over
On Sun, October 2, 2005 12:07, Patrick said:
On Sun, 2005-10-02 at 10:21 +0100, Wayne wrote:
Hiyall,
been following this for a while, just thought I would add a bit to the
debate, but doesn't the Cisco system (Call Manager?) run on an Windows
2000 based server - if it was that bad why would
On 30 Sep 2005, at 18:14, Anders Svensson wrote: Hi all! We have to setup 2 *servers. Now I am interested in possible capacity. Server 1. Should be used for getting traffic from our Telco using IAX and send it out using SIP. No transcoding, ulaw both ways. What is possible capacity on 1 server
Hi
Having issues configuring voicemail box
Ran command /usr/src/asterisk/addmailbox .gets
error No such file or directory
Asterisk was installed without any error message,
Running on R.H Fedora all packages installed
Please assist
Thorben Jensen wrote:
BUT how do I delete a database entry?
Thorben,
You still have to use the old DBdel command.
Doug
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Asterisk-Users@lists.digium.com
Wayne wrote:
Hiyall,
been following this for a while, just thought I would add a bit to the
debate, but doesn't the Cisco system (Call Manager?) run on an Windows
2000 based server - if it was that bad why would Cisco choose to run
it? Also 3Com use NT/2000 to run the H323 gateway.
Omar McKenzie wrote:
Hi
Having issues configuring voicemail box
Ran command /usr/src/asterisk/addmailbox … .gets error ‘No such file
or directory’
Asterisk was installed without any error message,
Running on R.H Fedora – all packages installed
I don't know what that command is, to add
I installed a Marquee sign (aka reader board), which was sent emergency
information via an RS-232 serial port. It was pretty nifty, as it was
during to 'everywhere must have caller ID' phase in the 90s.
Most signs are cheap, and can just be placed in the clubhouse window. You
could even
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05, Federico Alves [EMAIL PROTECTED] wrote:
I am sending calls using Oh323 to a Cisco Gateway (AS5300), and although I
set the caller id correctly in my perl AGI script
$AGI-set_callerid($ani); , the
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from
obj-sys. This driver is included into asterisk-addon package.
Adam.
Cytowanie Asterisk guy [EMAIL PROTECTED]:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05,
I have the same problem with asterisk CVS HEAD and asterisk-ooh323 module from
obj-sys. This driver is included into asterisk-addon package.
Adam.
Cytowanie Asterisk guy [EMAIL PROTECTED]:
I get the same problem. ( asterisk1.2.0beta1+oh323 0.73),
any suggestion for this ?
On 6/13/05,
Trying again, this never made it to the list for some reason.
Thanks..
- Original Message -
From: Mojo Jojo [EMAIL PROTECTED]
To: Asterisk Users List asterisk-users@lists.digium.com
Sent: Thursday, September 29, 2005 10:26 PM
Subject: Channel Banks, what are they for?
Can someone
Any of the more current Win32 systems can be programmed to handle near
real-time events (eg, sip, rtp) just like linux, bsd, and other O/S's.
Obviously, Call Manager is one such system. It's really not an O/S
religious war/discussion, but rather a lack of knowledge (on any O/S
that a poster might
/usr/src/asterisk/contrib/scripts/addmailbox
On Sun, 2005-10-02 at 08:22 -0400, Doug Lytle wrote:
Omar McKenzie wrote:
Hi
Having issues configuring voicemail box
Ran command /usr/src/asterisk/addmailbox … .gets error ‘No such file
or directory’
Asterisk was installed without
Mojo Jojo wrote:
Can someone explain to me what a channel bank is used for?
Multiple of things, but the primary one being, handling a lot of analog
phone lines. We currently have 16 lines in my test facility. These
lines are Centrex lines and we get them very cheap ($10 per line), I
On Sunday 02 October 2005 09:09, Mojo Jojo wrote:
Can someone explain to me what a channel bank is used for?
For example, if I had a 24 channel PRI setup with an Asterisk box
attached to it via a TE110P, how would a channel bank make my life
better?
Anyhow, I just have not clue and
I installed a Marquee sign (aka reader board), which was sent emergency
information via an RS-232 serial port. It was pretty nifty, as it was
during to 'everywhere must have caller ID' phase in the 90s.
Most signs are cheap, and can just be placed in the clubhouse window. You
could
Hi;
I've got an AAH installation where a customer wants to install an active
Eicon DIVA BRI card. AAH is built on Centos 3.5 which is currently at
kernel 2.4.21.37. Support for Eicon active cards is built-in.
I've debugged and run the [EMAIL PROTECTED] install-Eicondiva script but when I
On Sun, 2 Oct 2005, John Daragon wrote:
Hi;
I've got an AAH installation where a customer wants to install an active Eicon
DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel
2.4.21.37. Support for Eicon active cards is built-in.
I've debugged and run the [EMAIL
Armin Schindler wrote:
On Sun, 2 Oct 2005, John Daragon wrote:
Hi;
I've got an AAH installation where a customer wants to install an active Eicon
DIVA BRI card. AAH is built on Centos 3.5 which is currently at kernel
2.4.21.37. Support for Eicon active cards is built-in.
I've debugged and
Thanks lots
Editing the voicemail.conf did work, I was actually using the
VoiceMail2 command in the extensions.conf file , changed to VoiceMail and
all is well.
I tried the command /usr/src/asterisk/contrib/scripts/addmailbox - that
however did not work - same error 'No such file or
I have an adit 600 with one fxo card connected to a Digium single span T1 card.
CallerID, disconnect supervision work perfect, however the users
complain that they have some sound quality issues, after testing it I
realized that whenever one is in a phone call they get like silence
between the
On Sun, Oct 02, 2005 at 06:15:43AM -0700, Derek Whitten wrote:
/usr/src/asterisk/contrib/scripts/addmailbox
As people have stated before: it is no longer needed. Asterisk will
add the mailbox files on its own, given the write permissions.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
Sounds like no comfort noise.do you see this as well on VOIP trunks? sign
up for goiax or something like that to testI tend to think it is between
asterisk and the polycoms.
You could also test with a Sipura SP3000 as a replacement to see if you have
the same issue, or if you have
Thank you
Doug Lytle [EMAIL PROTECTED] skrev i en meddelelse
news:[EMAIL PROTECTED]
Thorben Jensen wrote:
BUT how do I delete a database entry?
Thorben,
You still have to use the old DBdel command.
Doug
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--Bandwidth and Colocation
Good explanation Rich. Unix was built for the riggers of the Telecomm
industry. You won't find Windows running the PSTN. Unix and Linux are
used where their needed for real time processing and the highest
reliably. Windows is a productively OS that is easy to use for non
technical people. I
I am trying oh323(version 0.67) , make call from sip UA to h323 gateway,
can't get Call-id pass from sip UA to h323 gateway, h323 always gets
call-ID sent from Asterisk as *. are there any configure to pass
the correct call-id from sip UA to h323 gateway? or this is a bug in
oh323 0.67?
I installed this card, everything work, i can make call and receive
call with no echo and great sound quality, but after between 5 to 50
secs the call disconnect by itself, in the log i don't see nothing
revelant.
I don't share any IRQ, zttest show me values between 99.98 and 100.
The only thing
Quit aware of the telecomm industry; spent
21 years in buried in techie detail as an engineer and had
a ton of fun. Not sure the overall
programming community would agree with real-time vs productivity
assessment; lots of folks out there
writing production systems on Win32 systems that
Works with 1.2 bets!
Voice is MUCH better than festival!
John Novack
trixter http://www.0xdecafbad.com wrote:
I wrote a very very simple shell script and an even simplier macro to
use the IBM TTS engine within asterisk for prompts. While its free you
are limited on the number of requests
Doug Lytle wrote:
Omar McKenzie wrote:
Hi
Having issues configuring voicemail box
Ran command /usr/src/asterisk/addmailbox … .gets error ‘No such file
or directory’
Asterisk was installed without any error message,
Running on R.H Fedora – all packages installed
I don't know what that
Kind of the idea I got..
So then I could take a channel bank and feed my PRI from the telco into it
then on the other side I would have a bunch of RJ11 ports to use as normal
analog type phone lines?
What about this situation... I have a PRI going directly into an Asterisk
box now in a
Corey S. McFadden wrote:
We've been experiencing an odd issue lately. I'm not sure when it started
because it's not happening on most calls--it seems confined to a couple of
our queues. It's consistent though.
Here's the CLI output:
-- Got SIP response 400 Bad Request back from
Olle E. Johansson wrote:
Corey S. McFadden wrote:
Here's the CLI output:
-- Got SIP response 400 Bad Request back from 192.168.249.94
-- SIP/502-9a58 is circuit-busy
I've tried a few different Asterisk versions CVS-HEAD, stable, even 1.2
beta. I've also bounced between SIP firmware
i have opened an account with callshopcompany,and
when ive tried to send calls by the sip i had a
message show an asterisk invitation problem i had
these sip configuration:
sip.conf
[callshop]
type=peer
host=sip.callshopcompany.com
username=X
secret=XXX
Then i
Doug Lytle wrote:
Olle E. Johansson wrote:
Corey S. McFadden wrote:
Here's the CLI output:
-- Got SIP response 400 Bad Request back from 192.168.249.94
-- SIP/502-9a58 is circuit-busy
I've tried a few different Asterisk versions CVS-HEAD, stable, even
1.2 beta. I've also
On Sun, 02 Oct 2005 00:53:03 -0700, you wrote:
I wrote a very very simple shell script and an even simplier macro to
use the IBM TTS engine within asterisk for prompts. While its free you
are limited on the number of requests you can do within a day.
If anyone is interested its available at
I am upgrading to Asterisk-Realtime and stumbled upon a problem
converting my existing sip.conf register command to the RealTime
format. It seems that sip_friends table setup doesn't allow for such
thing to happen. So far the only way I see to do this is dumping the
sip_friends table setup in
where can i find these properties:
answeronpolarityswitch and hanguponpolarityswitch.
Regard;
wassim
__
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http://mail.yahoo.com
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I've just interfaced an Avaya 4621 set to [EMAIL PROTECTED] It's running the
2.2 SIP firmware released August 17th.
I've run into some strange behavior with this phone. 1st, in order to
get two way audio, I had to tell * that it was behind a NAT even
though it is on the same subnet as the *
I'm having a problem with sound output to the console.
My basic dial plan is as follows:
exten =
_1NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},30,A(beep))
exten = _1NXXNXX,2,Playtones(info)
exten = _1NXXNXX,3,Hangup
I get the following output in the console:
___*CLI dial [EMAIL
YUP!!!
I guess working on this at 11:30pm at night, really need your wits about
you... Thanks... did that and it worked like a champ..
One other question I have the app_dtmftotext.c file is located at the
root of spandsp... however, when I had this file in my Makefile to
compile, and then
On Sunday 02 October 2005 15:29, Mojo Jojo wrote:
So then I could take a channel bank and feed my PRI from the telco into it
then on the other side I would have a bunch of RJ11 ports to use as normal
analog type phone lines?
I don't know of any channel banks that accept PRI. Channel banks
[EMAIL PROTECTED] wrote:
One other question I have the app_dtmftotext.c file is located at the
root of spandsp... however, when I had this file in my Makefile to
compile, and then have asterisk load it, asterisk complains on it.. Is it
really needed?
That I couldn't tell you.
Doug
Below is the code that we have. We are getting ready to run a sniffer
and see if/why asterisk is doing the writes separately instead of in one
chunk.
There were some changes in CVS that appear to address this issue.
However, if you are trusting the manager to write full events for your
Im receiving the following error over and over, adnauseam:
Oct 1 23:59:53 NOTICE[3194]:
chan_sip.c:5890 check_auth: stale nonce received from CNAME-CID
sip:[EMAIL PROTECTED]
Does anyone know what stale nonce is?
Thanks!
Paul Conn
app_dtmftotext.c is a separate application. If you don't want an
application which turns DTMF into text, then don't use it.
Regards,
Steve
[EMAIL PROTECTED] wrote:
YUP!!!
I guess working on this at 11:30pm at night, really need your wits about
you... Thanks... did that and it worked like a
Hi Paul,
I'm receiving the following error over and over, adnauseam:
Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID
sip:[EMAIL PROTECTED]'
Does anyone know what stale nonce is?
Thanks!
This is normally not an error.
Digest authentication in
Hi,
I have a analog phone connect to a WCTDM card.
It used to work fine. Now recently, after several conf change and power restart,
it stops working.
Whenever I pickup the phone, instead of hearing the dial tone, I hear a busing beeping
tone, like a machine gun is firing. :) However, from
Does anyone have any success using AudioCodes FXO terminating calls ?
Ehsan
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We use the mp-108 fxs units a lot.. also use mp-2000 units for pri_cpe
end. Probably the closest thing to your situation is our use of the
mp2000 terminating a pri at the z end and sending calls on to asterisk.
While it was not without it's flaws I can say that it worked rather well
just using
Since the search engine on voip-info.org is not working correctly with old
links, etc..
I was curious if there is some hidden talent in the IAX2 outbound dialing?
What I'm asking about is:
Dial(IAX2/g1/${EXTEN})
Is there a way to set up groups like the above command using either SIP or
IAX2
Thank you for your advise, I'll find something with a lot of memory
Adrien
--
Adrien Laurent - CIO
514-284-2020
[EMAIL PROTECTED]
www.modulis.ca
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Saturday, October 01, 2005 12:25 AM
[EMAIL PROTECTED] wrote on 10/01/2005
10:17:47 AM:
Is there a way with either RHEL or CentOS to force it to use an
APIC-enabled kernel? I've tried Googling but no success.
I can find no way of doing this during the install.
If you have a single processor system, AFAIK you are stuck with
Hello!
A small suggestion for an improvement
to zttest: some sort of histogram to show a broader range of the
results that are being returned. For example, on a test machine I
ran each of the following items in separate infinite loops at the same
time:
ssh-keygen -b 8192 -t rsa -f /test.key
dd
Hello!
While performing some zttest's for some
time today, I was also keeping an eye of a top of the machine. While
the zttest was running, I also had a ssh-keygen and a dd creating a 5GB
file on an EXT3 partition running. I noticed that for the most part,
I got a decent number of 100%'s, and a
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