This cuts all of the download time from the install process minimizing
service downtime to a fraction of what it would othewise be.
an other way to minimize it is, creating RPM-packages for Asterisk (plus
addons and zaptel) and install them, with this the downtime for updating
is just some
Hi,
what software do i need to convert an mp3 to a g729 format?
I have a portal where a user can upload their own MP3, but when a user
is using a g729 codec, the music on hold has a crackly sound. using g711
it's very clear.
so what i'd like to do is when they upload an MP3 i will make a copy
Hello,
Thanks for your output.
I can apply a fix or the only issue consiste in updating Asterisk ?
Regards,
Adrien
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
Schmidt
Envoyé : lundi 22 juin 2009
Hi
The calldate column is the date and time of the call, here's the
definition of the cdr table
CREATE TABLE `cdr` (
`calldate` datetime NOT NULL default '-00-00 00:00:00',
`clid` varchar(80) NOT NULL default '',
`src` varchar(80) NOT NULL default '',
`dst` varchar(80) NOT NULL default '',
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu and play that file.
I tried
exten=s,4,Playback(/record/deneme.gsm)
hello,
if you can live with hundreds of unclosed bugs and some great security
holes still open in ver. 1.2.7.1
but if you dont have patched your asterisk version i would really
recommend you to update.
best regards
steve
Adrien Lemoine schrieb:
Hello,
Thanks for your output.
I can
Hi
Drop the '.gsm' from the filename.
Ish
Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu
Thanks for your reply. I saw that info also on voip-info.org.
I was wondering if I could define other columns, like those used for
billing (as defined in my sip.conf).
Jonas.
On Tue, 2009-06-23 at 09:22 +0100, Ishfaq Malik wrote:
Hi
The calldate column is the date and time of the call,
hello,
you should try it with the following:
exten=s,4,Playback(record/deneme)
the .gsm is not necessary
best regards
steve
Oguzhan Kayhan schrieb:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it
exten=s,4,Playback(/record/deneme.gsm)
should be
exten=s,4,Playback(/record/deneme)
so without a format.
Thank you.
That worked :)
On Tue, 2009-06-23 at 11:31 +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice
There's a couple of columns you can write to in the table, most notably
accountcode and userfield. There is more info here.
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
I'm not sure about defining additional columns and writing to them
through the dialplan but I don't think
Hi, also with your template I have always the same problem !
Thanks.
--
Salvatore.
- Original Message -
From: David Gibbons d...@videon-central.com
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Monday, June 22, 2009 2:41 PM
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters connected to
a VoIP gateway. Should we be concerned with such a set-up as far as voice
quality and other issues are concerned? Any experiences with GSM
What do I define as amaflags in my sip.conf ??
Can I leave it to be amaflags=billing ? What use of defining it as
billing or documentation when the number of columns are pre-defined ?
Jonas.
On Tue, 2009-06-23 at 10:39 +0100, Ishfaq Malik wrote:
There's a couple of columns you can write to
On Tue, Jun 23, 2009 at 11:31:54AM +0300, Oguzhan Kayhan wrote:
Hello,
I am trying to create a simple IVR for testing..
What i did is to create a voice file from asterisk-gui.
And i saw it created that under /var/lib/asterisk/sounds/record/ as
deneme.gsm
Then i tried to make a IVR menu and
On 23 Jun 2009, at 11:18, jonas kellens wrote:
What do I define as amaflags in my sip.conf ??
Can I leave it to be amaflags=billing ? What use of defining it as
billing or documentation when the number of columns are pre-defined ?
Eh? amaflags just sets a field in the database called
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters connected to
a VoIP gateway. Should we be concerned with such a set-up as far as voice
quality and other issues
The price difference is HUGE. Analog i about 66% cheaper.
On Tue, Jun 23, 2009 at 12:44 PM, Gordon Henderson
gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote:
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
Yip the VoiceBlue SIP units are very good but a bit pricey
Gordon Henderson wrote:
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Hi all,
We have been planing for a long time to set up GSM mobile trunks for
termination, and were planing on going with analog GSM adapters connected to
a VoIP
Yup, even costlier than Portech. And we also tried chan_mobile as a low
cost alternative, but that seems to be very buggy.
On Tue, Jun 23, 2009 at 1:05 PM, Duncan Turnbull dun...@e-simple.co.nzwrote:
Yip the VoiceBlue SIP units are very good but a bit pricey
Gordon Henderson wrote:
On Tue,
On Tue, 23 Jun 2009, Sasa Bobek wrote:
The price difference is HUGE. Analog i about 66% cheaper.
But you then need some sort of analogue adapter/interface to feed the
analogue GSM module...
Although if you've already got this, it's a obviously a cheaper option.
However, here in the UK,
Thanks for the info Gordon. Just what I was looking for. I think I have
seen one of the telecom FM units, it actually has a whole phone inside :)
In my end of the world things are quite different :) Portech costs an
average of 160E per port, and the cost of the GSM adapter including the cost
of
-- Executing [0473775...@intern:1] NoOp(SIP/twinkle-088e6ea8,
conversation to GSM) in new stack
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-088e6ea8,
SIP/3starsnet/0473775006) in new stack
-- Called 3starsnet/0473775006
-- Got SIP response 482 Loop Detected back from
I believe this is more a human resources problem than a technical one. You
will first need some sort of CDR analysis tool to spot calls to expensive
destinations, and then you wil track back who was the agent in change of the
call.
I am sure that if there is word out that you are tracking these
On Tue, 23 Jun 2009, Sasa Bobek wrote:
Thanks for the info Gordon. Just what I was looking for. I think I have
seen one of the telecom FM units, it actually has a whole phone inside :)
They're designed for remote areas - they are wall mountable and have an
external antenna socket and an
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Karl Fife wrote:
I discovered that after running make, you can run 'make sounds' before
shutting down the service. This cuts all of the download time from the
install process minimizing service downtime to a fraction of what it would
othewise
On Tue, Jun 23, 2009 at 8:09 AM, jonas kellens jonas.kell...@telenet.bewrote:
-- Executing [0473775...@intern:1] NoOp(SIP/twinkle-088e6ea8,
conversation to GSM) in new stack
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-088e6ea8,
SIP/3starsnet/0473775006) in new stack
--
As I see it, even though this is an HR problem, there are some technical
solutions. The first one would be to show the Agent the CDR record. As
Abraham Lincoln said, Fool me once shame on you, fool me twice, shame on
me. The next one would be to block some exchanges/numbers. For example,
Do you understand what is happening ?
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-08de0490,
SIP/3starsnet/0473775006) in new stack
-- Called 3starsnet/0473775006
-- SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
-- Got SIP response 500 Service
Karl Fife wrote:
I was about to ask this question when I figured out the answer by combing
through the makefile.
I am posting this anyway because I think it's good to know, and I didn't
find any threads that speak to it when I searched the list history.
My Question was:
When updating
In sip.conf :
[general]
canreinvite=no ; Asterisk by default tries to redirect
the
; RTP media stream (audio) to go
directly from
; the caller to the callee. Some
devices do not
;
Dave,
I am very interested in seeing these scripts as well... Could you please
forward them my way as well...
Thanks,
Jeff Phelps
IT Support Specialist
IRS Circular 230 Disclosure: To ensure compliance with requirements imposed by
the IRS, McConnell Jones, LLP informs you that any U.S.
Hello -
We at Digium talk to reporters regularly, and we often are asked
for contacts in the Asterisk community who might be interested in
talking about their experiences or industry perspectives. We have a
fairly decent list of possible contacts in various sectors of the
industry,
Try resetting the phone to factory defaults. I have had some odd issues
when moving phones between CallManager and Asterisk that this was the
easiest fix. It might be worth a shot. Here are the directions:
Have you seen the new GSM module for the TDM-400 card? It fits right in
where you would normally have an FXO or FXS daughter card. Not sure
about the price but it stands to be cheaper than any external device.
Also allows for considerable density. Not sure about availability.
Steve Casto escribió:
I am trying to retrieve the cause code of a outgoing call over a PRI
where the number called is out of service. When an out service number is
called I get a recording that the number dialed is not a working
number. I see cause code 1 in in the CLI as soon as the call is
Hello
I noticed a small bug in the way my extensions.conf work:
Users can choose extensions 1-4 or 9 to tell why they're calling, and
I'll send an e-mail to the person(s) to whom is involved. Extension 4
is actually for personal messages for User1, and extension 9 is for
everyone (User1, User2,
First of all, this is a PHP problem, not an asterisk one. That being said,
could the empty case 2: be causing the problem? Have you run the php from a
command line to see what happens (php send_call_notification.phpcli 3)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Steve Casto escribió:
/ I am trying to retrieve the cause code of a outgoing call over a PRI
// where the number called is out of service. When an out service number is
// called I get a recording that the number dialed is not a working
// number. I see cause code 1 in in the CLI as soon as
Hi,
I am attempting to implement Answering Machine Detect and have also played
with using BackgroundDetect instead. Does anyone recommend one over the
other? Here is the code I am using for the BackgroundDetect method (from
voip-info.org).
Thanks.
[detect]
exten = s,1,Set(MACHINE=0)
I do see that AMD.conf offers more parameters, but my initial testing has
had vastly varying results. (I meant AMD in the subject line. :/)
Thanks again.
_
From: John Regal [mailto:jre...@gmail.com]
Sent: Tuesday, June 23, 2009 12:03 PM
To: 'Asterisk Users Mailing List -
Alex Samad wrote:
I am having some problem forcing my tdm410 to alaw over ulaw...
You will want to set the alawoverride module parameter to 1. i.e.
'modprobe wctdm24xxp alawoverride=1' or alternatively, edit your
/etc/modprobe.d/dahdi file and add a line options wctdm24xxp
alawoverride=1
jonas kellens wrote:
Do you understand what is happening ?
I don't understand what this sentence means :
SIP/3starsnet-08d70ea8 is making progress passing it to
SIP/twinkle-08de0490
Pretty simple really. Your SIP trunk 3starsnet is making progress with
the call and Asterisk is passing that
On Tue, Jun 23, 2009 at 9:36 AM, jonas kellens jonas.kell...@telenet.bewrote:
Do you understand what is happening ?
-- Executing [0473775...@intern:2] Dial(SIP/twinkle-08de0490,
SIP/3starsnet/0473775006) in new stack
-- Called 3starsnet/0473775006
-- SIP/3starsnet-08d70ea8 is
Thank you for your answer.
Could you explain why the call fails ?
Connected to Asterisk 1.4.25.1 currently running on asterisk (pid =
17936)
Verbosity is at least 25
Core debug is at least 5
-- Executing [0473775...@intern:1] NoOp(SIP/twinkle-0a0567f8,
conversation to GSM) in new stack
Calls succeed now because I have added in sip.conf :
[3starsnet]
type=peer
host=85.119.188.3
username=username
secret=
fromuser=username
fromdomain=sip.3starsnet.com
What does this 'fromdomain'-parameter do ?? So I can understand why this
is so important.
Jonas.
On Tue, 2009-06-23 at
Dave Fullerton dfullertaster...@shorelinecontainer.com wrote:
You can also just grab and un-tar the sound files by hand from:
http://downloads.asterisk.org/pub/telephony/sounds/
Good point because the MOH files would need to be bulled down manually if
you want to minimize downtime AND you
On Tue, Jun 23, 2009 at 11:32:08AM -0500, Shaun Ruffell wrote:
Alex Samad wrote:
I am having some problem forcing my tdm410 to alaw over ulaw...
You will want to set the alawoverride module parameter to 1. i.e.
'modprobe wctdm24xxp alawoverride=1' or alternatively, edit your
Barry L. Kline blkl...@attglobal.net wrote:
BTW, I just implemented my first system using the Polycom config system
you spoke about on VUC. I appreciate you taking the time to do that.
Thanks Barry. I'm glad you and others have benefitted from that.
There's a quick-and-dirty resource page
Yes it will be beneficial. We have a TDM410P and were getting nominal
performance until we recompiled the kernel. A good use of 6 hours.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Samad
Sent:
Hi
I was reading this article on installing asterisk 1.6 + debian
http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian
and I noticed they suggested to recompile to 1000Hz enable kernel, I
currently have a 250Hz stock
On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote:
Hi,
what software do i need to convert an mp3 to a g729 format?
I'm not aware of a package to do it in one step. Sox can work with a
large number of formats, but mp3 isn't one of them.
I have a portal where a user can upload their
On Tue, Jun 23, 2009 at 5:10 PM, Alex Samada...@samad.com.au wrote:
Is recompiling the kernel to the 1000Hz going to be beneficial to me,
the box is primarily used for firewall router / voip (Asterisk)
I could be wrong, but I think the 1000Hz suggestion specifically
refers to trying to do audio
On Tue, Jun 23, 2009 at 4:15 PM, Danny Nicholas da...@debsinc.com wrote:
Yes it will be beneficial. We have a TDM410P and were getting nominal
performance until we recompiled the kernel. A good use of 6 hours.
Most likely that was just incidental. Having either Digium or Sangoma
hardware
On Wed, Jun 24, 2009 at 07:10:15AM +1000, Alex Samad wrote:
Hi
I was reading this article on installing asterisk 1.6 + debian
http://www.howtoforge.com/installing-and-configuring-asterisk-1.6-and-postgresql-to-manage-cdr-and-realtime-config-on-debian
and I noticed they suggested to
On Mon, Jun 22, 2009 at 1:35 PM, Bryan
Field-Elliotbryan+asterisk-us...@nextalarm.com wrote:
We are trying to get Asterisk to behave correctly when our SIP clients
have Silence Suppression turn on, but are not having any luck.
Basically, there are several apps in Asterisk which won't send any
David Backeberg wrote:
I'm not aware of a package to do it in one step. Sox can work with a
large number of formats, but mp3 isn't one of them.
Yes, sox can work with MP3 files, but not G.729. Not all distributions
include MP3 support in their sox builds due to patent licensing
concerns,
On Wed, Jun 24, 2009 at 01:02:08AM +0300, Tzafrir Cohen wrote:
On Wed, Jun 24, 2009 at 07:10:15AM +1000, Alex Samad wrote:
Hi
I was reading this article on installing asterisk 1.6 + debian
David Backeberg wrote:
On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote:
Hi,
what software do i need to convert an mp3 to a g729 format?
I'm not aware of a package to do it in one step. Sox can work with a
large number of formats, but mp3 isn't one of them.
hi sir, i'm ok if
http://www.voip-info.org/wiki/view/sox
On Tue, Jun 23, 2009 at 8:40 PM, Ronnha...@gmail.com wrote:
David Backeberg wrote:
On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote:
Hi,
what software do i need to convert an mp3 to a g729 format?
I'm not aware of a package to do it in one
On Wed, 24 Jun 2009, Ron wrote:
David Backeberg wrote:
On Tue, Jun 23, 2009 at 3:31 AM, Ronnha...@gmail.com wrote:
what software do i need to convert an mp3 to a g729 format?
I'm not aware of a package to do it in one step. Sox can work with a
large number of formats, but mp3 isn't one of
Hi, all
Are there any patches for chan_h323/chan_ooh323 to support video?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I'm using the latest SIP firmware from Avaya. The phone receives the
46xxsettings.txt OK, and then after entering extension and password it
goes to the home screen saying 'Registering'. When I check
options-ViewIPSettings-IPAddresses on the phone,
the registrar and SIP Proxy fields are blank.
I
Does anyone know if the Avaya 5610 supports SIP? I cannot even find
the phone listed on the Avaya support site. Anyone have an example
config?
Thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing
65 matches
Mail list logo
