Third, those complaining of low volume in emailed files are usually
using a compressed format. In the uncompressed wav format, the volume is
effectively doubled by shifting the audio data to the left one bit. This
is done at the format level. Of course on playback via asterisk, it
checks to see if
Hi,
We have an installation that is expanding rapidly. We are about to move from
1 x E1 towards 4 x E1 (PRI CPE) and therefor we bought a TE405P card.
Having set up a test mashine based on a 2,4GHz Celeron CPU and 256MB RAM,
all is fine while modprobing a single E1 span (30 B lines + 1 D) on our
Colin Anderson wrote:
I have 4 gig in my * box. I'm tuning for performance and I'd like to ask
opinions:
1. asterisk -p == renice -20 ??
What asterisk -p does is mark the aterisk process as a POSIX real time
priority process. Unless you have other process marked in the same way,
the scheduling
Me wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and
TELUX wrote:
Redhat 9 is running 100% cpu usage. I had a couple boxes doing this.
upgraded to Fedora and its ok.
I would try running asterisk with LD_ASSUME_KERNEL=2.4.1 if it isn't
already.
Gilad
--
Gilad Ben-Yossef [EMAIL PROTECTED]
Codefidence. A name you can trust(tm)
Web:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and availability etc.
Well, you must be dreaming :)
It all depends on your buying power, if you have at least 2-3 million
minutes goto
[EMAIL PROTECTED] schrieb:
After calling the number and no response of our client the voice-box
gives response. Thats ok... but after the voice-box, which ist self-
configured by our client the server respondes with the notivication to
leave your message please speak after... blablabla
Does
On Sat, 27 Nov 2004 19:37:54 +, Corvin [EMAIL PROTECTED] wrote:
I have very simple question, how to limit SIP phone user making
calls to for example longdistant calls?
This is how I do it -
Thank you very much to all of you.
I have one more question which troubles me.
We have
Soemthing goes wrong with this mail list:
I am getitng something like it:
Sorry. Your message could not be delivered to:
Aster risk (Mailbox or Conference is full.)
??
This is rest of my post.
On Sat, 27 Nov 2004 19:37:54 +, Corvin [EMAIL PROTECTED] wrote:
I have very simple
Hello,
I've got my dialplan configured to do a double ring when a customer
service call comes in, and a normal ring when an extension is dialed
directly. When a customer service call is transferred, I want to ring
to revert back to normal.
In the local extension macro, I have the following
;
Hi,
Reviewing the archives I saw /2004-October/070314.html from Tim Lewis. His
error is almost identical to mine i.e. when make clean; make install in
asterisk sub dir, I get the following:
pbx_dundi.c:54:18: zlib.h: No such file or directory
pbx_dundi.c: In function `update_key':
On Tuesday 16 November 2004 17:12, Jay Milk wrote:
I'm a fairly reasonable person, and I have yet to see one good argument
(and quoting netiquette is not on argument, that's opinion) for
bottom-posting. To me, it is terribly inefficient and wastes time,
especially when you hide your post
Hello list,
I am glad to announce that XC-AST 0.5, released today, offers real time
queue monitoring facilities that let you see the calls flowing through a
set of Asterisk queue(s) and agents logging on and off.
This way, XC-AST provides an one-stop solution to generate reports,
monitor
On 25 Nov 2004, at 21:09, Asterisk wrote:
I've just got a Snom 190 phone with which I'm really pleased. I can get
the LEDs on the keys to light in response to an extension being in use
which is cool, but there's a feature I'd like to implement.
i'm very interested in this option, have a snom 190
hello list,
I was wondering: anybody ever wrote an asterisk based bbs? not a bbs about
asterisk, but a vocal bbs that runs on asterisk, so that people can call,
hear the discussion of the day, leave messages, etc.
it seems a rather basic application to me though I cannot find much about.
Soemthing goes wrong with this mail list:
I am getitng something like it:
Sorry. Your message could not be delivered to:
Aster risk (Mailbox or Conference is full.)
??
Regards,
Corvin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Soemthing goes wrong with this mail list:
I am getitng something like it:
Sorry. Your message could not be delivered to:
Aster risk (Mailbox or Conference is full.)
??
Probably nothing to do with you.
A lot of people run mail software that isn't fit for consumption even
by
On Sun, 28 Nov 2004 15:37:44 +, Corvin [EMAIL PROTECTED] wrote:
Soemthing goes wrong with this mail list:
I am getitng something like it:
Sorry. Your message could not be delivered to:
Aster risk (Mailbox or Conference is full.)
This is a problem on your end. I replied yesterday
In particular, people like to run autoresponders and the like which
completely ignore the envelope sender (which is where all backchannel
communications, such as errors, ought to go) and instead target the
listed From: address in the body of the message, which doesn't
necessarily have
On Sun, 28 Nov 2004 16:25:15 +, Corvin [EMAIL PROTECTED] wrote:
Could you reccomend me something better than handbook, and vo-ip wiki?
If you haven't seen it already, you can also try http://www.asteriskdocs.org.
Leif Madsen.
http://www.asteriskdocs.org
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable/Redundant network and
On Sun, 28 Nov 2004 11:23:01 +, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
pbx_dundi.c:54:18: zlib.h: No such file or directory
I'm going to make an educated guess that you don't have zlib and
zlib-development packages installed. These are now required to be
installed for the CVS branch.
Steven Critchfield wrote:
Second, read the rants on licensing. Unless you find a BSD licensed mp3
encoding library and convince Mark of it's need, it is unlikely to make
it to the core code base.
When snackAmp blew up on GSM-encoded wav files I did some cursory
research and found FLAC:
lenz schrieb:
I was wondering: anybody ever wrote an asterisk based bbs? not a bbs
about asterisk, but a vocal bbs that runs on asterisk, so that people
can call, hear the discussion of the day, leave messages, etc.
It doesn't really make sense to me. It only makes sense for some very
limited
Hi,
I search How To transfer call between my SIP phone.
I have an PSTN line (X100P) and 10 grandstream budge tone phone.
For example I want :
- Reveive an external call and send it to SIP/phone1. At this point no
problem.
- After my receptionnist want transfert extern call at SIP/phone2... I
Jeremy SALMON wrote:
I have an PSTN line (X100P) and 10 grandstream budge tone phone.
Jeremy,
Receive call, press flash, call other party, wait for answer, press
transfer, hangup.
I believe that is what I saw on an earlier post.
Doug
___
Has anyone successfully built Asterisk with linux 2.6.9 kernel? It fails
in my zaptel build trying to find a Makefile in the
/lib/modules/2.6.9/build directory - thanks.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Perhaps I could pay you to teach me how to use google properly just send me
Steve's terse post does actually contain the answer to your how to
google question:
http://www.google.com/search?q=%22ignoring+signalling%22+site%3Alists.digium.com
Reverse engineering the above gives this: (entered in
Hello.
I have this problem. In my asterisk box, I was running debian woody with
asterisk package from backports.org. Last friday I upgraded from debian
to sarge and change from kernel 2.4.18-1-686 to kernel 2.6.8-1-686,
rebuild zaptel kernel module and also upgrade to asterisk 1.0.2. But
now
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
Symlink /lib/modules/2.6.9/build to /usr/src/linux
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven P. Donegan
Sent: Sunday, November 28, 2004 10:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk/linux 2.6.9 kernel
Title: OS Choice ?
Do I have any other options besides RH 9.0 ?
Best Regards,
Alex Brecher
Visit us at http://www.Successfulhosting.com
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Title: Phone Selection
I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you suggest and why please ?
Best Regards,
Alex Brecher
Visit us at http://www.Successfulhosting.com
___
Asterisk-Users mailing list
[EMAIL
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Its very helpful if you actually set them to
You always have a choice.. Gentoo, Debian... and as always RedHat is NOT an
OS. It's a Distro.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Alex Brecher
Sent: Sunday, November 28, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven P. Donegan
Sent: Sunday, November 28, 2004 10:30 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
Oh stop messin with that conf file
exten = 555,1,MeetMe(|dM)
NEXT!!!
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Saturday, November 27, 2004 8:48 AM
To: Asterisk Users Mailing List - Non-Commercial
On Sun, 2004-11-28 at 08:29 -0800, Steven P. Donegan wrote:
Has anyone successfully built Asterisk with linux 2.6.9 kernel?
Yes.
It fails
in my zaptel build trying to find a Makefile in the
/lib/modules/2.6.9/build directory - thanks.
Someone posted a patch for the zaptel Makefile and it
On Sun, 28 Nov 2004, Brian West wrote:
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Hi,
Anyone know how can I send a username or account id (h.323) and a
password to register on a remote Gatekeeper. I am using the Nuphone
channel with the h323.conf. I tryed everything but Asterisk always
send root as account id and the Gatekeeper rejected me.
Thank you very much...
Richard Lyman wrote:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steven P. Donegan
Sent: Sunday, November 28, 2004 10:30 AM
To: [EMAIL PROTECTED]
Subject:
Richard Lyman wrote:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
shouldn't that be 'to /usr/src/linux-2.6'
Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp
(error with a reference to non-existent sk_buf-ethernet.mac or similar)
but there is a
Andy Burns wrote:
Richard Lyman wrote:
Brian West wrote:
Symlink /lib/modules/2.6.9/build to /usr/src/linux
shouldn't that be 'to /usr/src/linux-2.6'
Yes, also FYI I had problems building zaptel 1.0 on 2.6.9-1.681_FC3smp
(error with a reference to non-existent sk_buf-ethernet.mac or
On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote:
Fair enough. If my unserstanding is correct perhaps someone can add a
note
to the wiki? It is not totally obvious.
Peter, why don't *you* add a note to the Wiki?
This is a community-supported project, and you're the community.
On Sun, 28 Nov 2004 11:28:05 -0500, Doug Lytle [EMAIL PROTECTED] wrote:
Jeremy SALMON wrote:
I have an PSTN line (X100P) and 10 grandstream budge tone phone.
Jeremy,
Receive call, press flash, call other party, wait for answer, press
transfer, hangup.
I believe that is what I saw
Brian West wrote:
; make sure ring is set to default
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
exten = s,n,SetVar(ALERT_INFO=Bellcore-r3)
exten = s,n,SetVar(_ALERT_INFO=Bellcore-r3)
exten = s,n,NoOp(${ALERT_INFO})
exten = s,n,NoOp(${_ALERT_INFO})
Its very helpful if you
I don't agree with this patch yet... It's the distro's fault for doing this
wrong and I don't feel we have to work around it. The few people I talked
to have Symlinks the build to /usr/src/linux or the like. Then again I
may be wrong anyone know what is the right(tm) thing to do here is?
bkw
Well, being a dinosaur (i.e. a very long UNIX/Linux experience person) I
was not happy when it went from just a symlink of linux-kernel - linux
to the current practice (RedHat style) of linux-kernel -linux-X.Y
Just my .02$
Brian West wrote:
I don't agree with this patch yet... It's the distro's
- Original Message -
From: Joe Greco [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 9:54 AM
Subject: Re: [Asterisk-Users] am i baned or something?
Sadly, the small portion of the Internet community that has a clue does
not seem to care
I have a weird problem and I cannot put my finger on it.
I hope somebody can help me out.
The quick way to solve this problem: Get an HFC-PCI card. It'll cost
you 20-30 euros and with that you can use bristuff from
http://junghanns.net/. This makes the HFC-PCI card a zaptel device.
The other
Hi.
I'm really new.
I was just wondering if it is possible at all to do a IP to IP call
without a * server (or as a matter of fact, any other kind of server)?
say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at
hisdomain.com's 192.168.0.3. Is this sort of things possible? Or must
Hi,
I'm slowly getting to grips with *. My next quest is to get IAX2/FWD
calls working.
I've setup a fwd account and added IAX capability to it via the website.
I got the email saying it had been done with some welcome text and sample
phone numbers to try, such as 10001 for the answer phone.
I
Mike Dent [EMAIL PROTECTED] writes:
I'm slowly getting to grips with *. My next quest is to get IAX2/FWD
calls working.
[...]
Basically when I make a call it rings out but no answer.
FWD's IAX gateway isn't working these days. Noone seems to know why.
-tih
--
Tom Ivar Helbekkmo, Senior
check out skype
- Original Message -
From: nkb [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 2:07 PM
Subject: [Asterisk-Users] IP to IP call without server?
Hi.
I'm really new.
I was just wondering if it
On Sun, 28 Nov 2004, Brian West wrote:
I don't agree with this patch yet... It's the distro's fault for doing this
wrong and I don't feel we have to work around it. The few people I talked
to have Symlinks the build to /usr/src/linux or the like. Then again I
may be wrong anyone know what
On Sun, 28 Nov 2004, Chad Scott wrote:
On Nov 28, 2004, at 9:45 AM, Peter Svensson wrote:
Fair enough. If my unserstanding is correct perhaps someone can add a
note
to the wiki? It is not totally obvious.
Peter, why don't *you* add a note to the Wiki?
This is a community-supported
Peter Svensson wrote:
On Sun, 28 Nov 2004, Brian West wrote:
I don't agree with this patch yet... It's the distro's fault for doing this
wrong and I don't feel we have to work around it. The few people I talked
to have Symlinks the build to /usr/src/linux or the like. Then again I
may be
So I reached the point where my PRI is accepting incoming calls, but I
cannot dialout. I must be doing something stupid, but I can't figure it
out. The Asterisk box is sitting between the Mitel and the phone company,
and has PRI lines to each. Asterisk was built from CVS r1-0
Log for a call
On Sunday 28 November 2004 19:01, [EMAIL PROTECTED] wrote:
So I reached the point where my PRI is accepting incoming calls, but I
cannot dialout. I must be doing something stupid, but I can't figure it
out. The Asterisk box is sitting between the Mitel and the phone company,
and has PRI
On Sunday 28 November 2004 19:25, Steven P. Donegan wrote:
Peter Svensson wrote:
On Sun, 28 Nov 2004, Brian West wrote:
I don't agree with this patch yet... It's the distro's fault for doing
this wrong and I don't feel we have to work around it. The few people I
talked to have Symlinks the
[EMAIL PROTECTED] wrote:
Hi.
I'm really new.
I was just wondering if it is possible at all to do a IP to IP call
without a * server (or as a matter of fact, any other kind of server)?
say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at
hisdomain.com's 192.168.0.3. Is this sort
nkb wrote:
Hi.
I'm really new.
I was just wondering if it is possible at all to do a IP to IP call
without a * server (or as a matter of fact, any other kind of server)?
say I'm at mydomain.com's 10.0.0.1 and I want to call my buddy at
hisdomain.com's 192.168.0.3. Is this sort of things
*Smack*, you're right, changing the g3 to g2 help nicely.
But now the PRI seems to be refusing the call (Channel 0/1 got hangup):
--snip--
-- Executing Answer(Zap/38-1, ) in new stack
-- Accepting call from '' to '15123455476' on channel 0/14, span 2
Nov 28 16:08:14 DEBUG[1894]:
On Sun, 28 Nov 2004, Bob Goddard wrote:
On Sunday 28 November 2004 19:25, Steven P. Donegan wrote:
Well - if 2.6.etc did adopt this it isn't reflected in actual make/make
install world - i.e. nothing gets installed in /lib/modules/anywhere...
And this is with kernel source from kernel.org
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my PC-Asterisk
server (which I yet have to install) and use them as 4 lines in case
anyone has to call me in / I have to call out using
In data Sun, 28 Nov 2004 16:55:06 +0100, Michael Vogel [EMAIL PROTECTED] ha
scritto:
lenz schrieb:
I was wondering: anybody ever wrote an asterisk based bbs? not a bbs
about asterisk, but a vocal bbs that runs on asterisk, so that people
can call, hear the discussion of the day, leave
[EMAIL PROTECTED] wrote:
*Smack*, you're right, changing the g3 to g2 help nicely.
But now the PRI seems to be refusing the call (Channel 0/1 got hangup):
--snip--
-- Executing Answer(Zap/38-1, ) in new stack
-- Accepting call from '' to '15123455476' on channel 0/14, span 2
Nov 28 16:08:14
Tomasz Chmielewski wrote:
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my
PC-Asterisk server (which I yet have to install) and use them as 4
lines in case anyone has to call me in
From what I have been told on this very list you can only use Diva
Server cards with asterisk because the 'cheaper' diva cards do not
support some stuff called 'capi'.
Or off course you can buy digium cards. They look pretty cool anyway -
can't wait to receive the onces I have ordered :-)
Which Distro is the most commonly used distro with Asterisk please ?
Best Regards,
Alex Brecher
-Original Message-
From: Brian West [mailto:[EMAIL PROTECTED]
Sent: Sunday, November 28, 2004 12:37 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Title: Phone Selection
Anybody here have suggestions on these phones please
?
Best Regards,
Alex Brecher
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
BrecherSent: Sunday, November 28, 2004 12:36 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users]
See the reply below yours.
I would hazard a guess that Redhat and SuSE, followed by Debian, are
probably the top three (RH and SuSE because of market share, and
enterprise server distros thbey have).
Greg
Alex Brecher wrote:
Which Distro is the most commonly used distro with Asterisk please ?
This looks like a config issue, class of service barred but getting
config information out of verizon is nearly impossible. I compared what
the Mitel is sending to asterisk (since the mitel does work with the PRI)
with what asterisk is sending and do not see any large differences.
debugging spam
Do I have any other options besides RH 9.0 ?
You always have a choice. Most distros provide some form of download for
their media. RH/FC, regardless of version, is easiest IMO because of
simple ISO image availability.
If you really wanted, you could build up a Linux machine based only on a
Jean-Michel Hiver wrote:
Tomasz Chmielewski wrote:
Hello,
I'm thinking of deploying Asterisk.
I already have a handful of EICON Diva 2.01 PCI ISDN cards.
I was thinking if it's possible to insert 4 such cards to my
PC-Asterisk server (which I yet have to install) and use them as 4
lines in case
[EMAIL PROTECTED] wrote:
In data Sun, 28 Nov 2004 16:55:06 +0100, Michael Vogel
[EMAIL PROTECTED] ha scritto:
lenz schrieb:
I was wondering: anybody ever wrote an asterisk based bbs? not a
bbs about asterisk, but a vocal bbs that runs on asterisk, so that
people can call, hear the
I've been looking at the wiki and the source for a long time now and I
just can't seem to get this straight... is VM config by PostgreSQL
functional?
From what I've seen it looks like it isn't. I've noticed:
1) Setting USE_POSTGRES_VM_INTERFACE to 1 in apps/Makefile sets CFLAGS
to include
If you are using GnuGK, i think this should do,
in your h323.conf file, configure an asterisk endpoint as follow for
instance
[time] Username
type=h323
e164=99
context=test
K.
- Original Message -
From: Nahuel Alejandro Ramos [EMAIL PROTECTED]
On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Only way that I know is to open the case and look at the slot to see if
there are two dividers. I would be interested in knowing this as well.
I've seen many motherboards that claim to be PCI 2.2 compliant, but
they
This looks like a config issue, class of service barred but getting
config information out of verizon is nearly impossible. I compared what
the Mitel is sending to asterisk (since the mitel does work with the PRI)
with what asterisk is sending and do not see any large differences.
debugging spam
On Sun, 28 Nov 2004 [EMAIL PROTECTED] wrote:
This looks like a config issue, class of service barred but getting
config information out of verizon is nearly impossible. I compared what
the Mitel is sending to asterisk (since the mitel does work with the PRI)
with what asterisk is sending and
On Sun, 28 Nov 2004, Lee wrote:
On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Only way that I know is to open the case and look at the slot to see if
there are two dividers. I would be interested in knowing this as well.
I've seen many motherboards that
I agree you can do this with SIP. but I would use skype, msn, yahoo or
VOIP blasters (get on ebay) for a simple call to call without a
server. its too much effort and too much to learn for a simple call.
On Mon, 29 Nov 2004 04:07:43 +0900, nkb [EMAIL PROTECTED] wrote:
Hi.
I'm really new.
I
Does soxmix works with asterisk ver. 0.9?
I have ver. sox-12.17.5 on Gentoo but the option m does not combine
two WAV files (In and Out) into one file. I have two separate files
in /monitor folder.
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten =
After lots and lots of digging it appears to me as if support for DB
based configuration is only via ODBC (known as extconfig). Sorry for
the noise. I'll be sure to compile my findings on my site and post a
suitable followup when it is complete.
On Sun, 2004-11-28 at 16:31 -0500, Tim Mattison
I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you
suggest and why please ?
I briefly tested the 480i a couple of weeks ago. Had a problem in that it
would not use the tftp server address contained in the dhcp response, so
had to define everything from the keypad to make
You may want to try calling StopMonitor to see if that forces a merge.
I've used Monitor before on Gentoo and it works with soxmix but I've
never tried to do it without an explicit StopMonitor.
On Sun, 2004-11-28 at 15:36 -0700, Joseph wrote:
Does soxmix works with asterisk ver. 0.9?
I have
Hi,
i am looking for a tool to merge the two wav files of a monitored call
into one. soxmix does that well but actually merges the two channels.
I would prefer a solution that creates a stereo wav file of the two mono
files so you have the called party on one (e.g. left) channel and the
calling
On Sun, 28 Nov 2004 16:32:10 -0600, Rich Adamson [EMAIL PROTECTED] wrote:
I'm looking at the Sayson 480i or the Cisco CP-7960. Which one would you
suggest and why please ?
I briefly tested the 480i a couple of weeks ago. Had a problem in that it
would not use the tftp server address
Joseph wrote:
Does soxmix works with asterisk ver. 0.9?
I have ver. sox-12.17.5 on Gentoo but the option m does not combine
two WAV files (In and Out) into one file. I have two separate files
in /monitor folder.
exten = 711,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten =
269804 Nov 28 16:16 11-20041128-161619|m-in.wav
-rw-r--r-- 1 root root 254924 Nov 28 16:16 11-20041128-161619|m-out.wav
Though that pipe +m is part of the filename, as if the soxmix wasn't
executing. Soxmix is in the path.
--
#Joseph
___
Asterisk
Lee wrote:
On Sat, 27 Nov 2004 20:53:24 -0500, Steve Totaro
[EMAIL PROTECTED] wrote:
Only way that I know is to open the case and look at the slot to see if
there are two dividers. I would be interested in knowing this as well.
I've seen many motherboards that claim to be PCI 2.2
On Sun, 2004-11-28 at 18:11 -0500, Dave DeChellis wrote:
Joseph wrote:
Does soxmix works with asterisk ver. 0.9?
[snip]
I belive you need 1.0 for the m option to work.
That was my initial impression. So need to pull few unstable packages
from portage to compile 1.0.1 or 1.0.2
--
On Sun, 28 Nov 2004 17:25:25 -0500, Steven Kalcevich (Lists) wrote:
I agree you can do this with SIP. but I would use skype, msn, yahoo or
VOIP blasters (get on ebay) for a simple call to call without a
server. its too much effort and too much to learn for a simple call.
I'm not a big fan of
On November 27, 2004 08:53 pm, Steve Totaro wrote:
Only way that I know is to open the case and look at the slot to see if
there are two dividers. I would be interested in knowing this as well.
What exactly do you mean by two dividers? Almost every PCI motherboard I
have has only one, and
A while back, I found a site that had the entire asterisk-users
mailing list archive in mbox format. Does anybody know if and where
such a thing is availible?
PJ
--
All men know the utility of useful things;
but they do not know the utility of futility.
-- Chuang-tzu
I figured it out.
The key was that the mitel was always dialing channel 14, Asterisk was
always dialing channel 1.
When I reconfigure asterisk to round robin dial the PRI group it started
at 1 and worked upwards. If the channel in the debug output is 9 dialing
works. It appears that the bottom
Andres Junge wrote:
snip
It seems that the first FXS module of my TDM22B is broken. Is that
correct? In that case how can I disable it? Just open the case and pull
it out? Or can I apply a configuration parameter to disable it?
You should be able to do so by removing all reference to that
I have already tryed this but asterisk always send root as h323_id
On Sun, 28 Nov 2004 21:31:52 -, kido noagbodji [EMAIL PROTECTED] wrote:
If you are using GnuGK, i think this should do,
in your h323.conf file, configure an asterisk endpoint as follow for
instance
[time]
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