Hello,
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
can't be eliminated.
I came to know
Thanks for all the replies.
So it seems auto-configuration code is a feature for ITSP, not for system
integrators looking for an easier way to configure each DECT base.
Too bad, as I'm sure this auto-configuration feature relies on standard
protocols we could play with (DHCP, TFTP, HTTP, ...).
Hi all,
I have to play a alaw file with .wav ext. How can I do this?
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On Wed, 21 Jul 2010, Olivier wrote:
Thanks for all the replies.
So it seems auto-configuration code is a feature for ITSP, not for system
integrators looking for an easier way to configure each DECT base.
Too bad, as I'm sure this auto-configuration feature relies on standard
protocols we
On Wed, Jul 21, 2010 at 12:02:03PM +0530, MohammedShehzad wrote:
Hello,
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the
Hi,
My Asterisk is not running on a virtual machine, and Debian does not have an
X Server.
I have no value with Kernel Timing enabled. Do you think it may be bound for
the proper functioning of chan_local? I have no problem with the Dial
(SIP/XX), but only with the Dial (Local/XX) :-(
Do you
On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote:
Hi,
My Asterisk is not running on a virtual machine, and Debian does not have an
X Server.
I have no value with Kernel Timing enabled. Do you think it may be bound for
the proper functioning of chan_local? I have no problem
Upgrading the kernel to 2.6.23 did it and now the results are far better.
The sound aint choppy no more.
dahdi_test -v -c 6 yeilds..
8192 samples in 8194.952 system clock sample intervals (100.036%)
8192 samples in 8222.504 system clock sample intervals (100.372%)
8192 samples in 8190.120 system
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
can't be eliminated.
I came to know
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type sip show peers in the asterisk
CLI . i am unable to see any failure logs when i do a reload
i can able to connect to the data source through odbc show in the
CLI,
On 21 July 2010 10:59, MohammedShehzad pmh...@gmail.com wrote:
I have been facing an issue that voice is getting distorted sometimes in
MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the
2.6.30-2-686 (Debian)
2010/7/21 Tzafrir Cohen tzafrir.co...@xorcom.com
On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote:
Hi,
My Asterisk is not running on a virtual machine, and Debian does not have
an
X Server.
I have no value with Kernel Timing enabled. Do you
Hi ,
I am trying to add an operator assistance feature to meetme , when the
user dials '0' ,support / help desk personnel should be added to the live
conference for live support / troubleshooting.
How can i do this ? Can I edit the meetme * menu and add a new menu item '
Press '0' for
Hi All,
I ' m using Cisco 7970 IP Phone and Asterisk 1.6.0.10-FONCORE-r40
(Tirxbox). My problem is that I upgrade my phone to SIP image but now
this phone is not registering.
The error likes this :
SIP/2.0 403 Forbidden (Bad auth)
The phone and Trixbox are in the same network. There arenot
Hi,
we are facing the problem , that we cannot distinguish between a trunk
an an extension.
On our trunk side, if the remote user puts us on hold the same
Musiconhold is played as if we would call another extension on the sam
Asterisk PBX.
Asterisk should play the music from the remote End not
My debug output is :
--- Transmitting (no NAT) to x.x.x.a:5060 ---
SIP/2.0 403 Forbidden (Bad auth)
Via: SIP/2.0/UDP x.x.x.a:5060;branch=z9hG4bK809cbff8;received=x.x.x.a
From: sip:5...@x.x.x.b;tag=0019065ca2d258b8c134-b34f8821
To: sip:5...@x.x.x.b;tag=as4099c235
Call-ID:
Hi ,
I am trying to add an operator assistance feature to meetme , when the user
dials '0' ,support / help desk personnel should be added to the live
conference for live support / troubleshooting.
How can i do this ? Can I edit the meetme * menu and add a new menu item '
Press '0' for
Hi, thanks a lot by the answers. But without the application Answer() the
problem remains.
Realized over a battery of tests and refined the problem. Follows:
A = External link that came with my Voip number.
B = Operator.
C = The extent to which A want to speak.
A called my number and B answer.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Quy Pham Sy
Subject: [asterisk-users] play alaw file with .wav extension
I have to play a alaw file with .wav ext. How can I do this?
Use the asterisk convert command or
Hi,
The files are actually alaw file (i check by file command). they're,
however, named with .wav extension, and these file are inherented with
current system I'm not allow to change these.
Quy
On Wed, Jul 21, 2010 at 8:12 PM, Danny Nicholas da...@debsinc.com wrote:
Asterisk won't be happy trying to play foobar.wav if it is actually a
.alaw file. Since you can't rename the existing files, there's no law that
says you can't copy them and play them correctly.Assuming that your
calls are using the alaw codec, this snippet would do the trick
Exten =
On 07/21/2010 04:35 PM, Danny Nicholas wrote:
Asterisk won’t be “happy” trying to play foobar.wav if it is actually a
.alaw file. Since you can’t rename the existing files, there’s no law
that says you can’t copy them and play them correctly.Assuming that
your calls are using the alaw
On Wed, Jul 21, 2010 at 3:09 AM, Murali Vasu vimurli@gmail.com wrote:
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type sip show peers in the asterisk
CLI . i am unable to see any failure logs when i do a reload
Op 21-07-10 08:32, MohammedShehzad schreef:
Hello,
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen
This sounds like a QOS issue (quality drops during heavy usage). Since it
was more prominent when you were on ISDN, that pretty much verifies it for
me. Can you prioritize voice traffic?
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Exten = 1234,n,System(/bin/cp foobar.wav /tmp/foobar.alaw)
it actually works, I made a link to the .wav file instead of copying it
ln -s foobar.wav foobar.alaw, and it works well.
No, that won't work either, because a WAV file has a header, and a raw
alaw file does not... so Asterisk will try
Op 21-07-10 17:18, Danny Nicholas schreef:
This sounds like a QOS issue (quality drops during heavy usage). Since it
was more prominent when you were on ISDN, that pretty much verifies it for
me. Can you prioritize voice traffic?
QOS on ISDN? Don't know how to do that.
Ron
--
NeoNova
Hi again
today when i was doing my research on this issue i found that even dialing a
sip user by it's IP also raises this problem. here is what i did,
First I dialed my registered user in normal way like this,
Dial(SIP/XYZ,30,rtT)
and during conversation audio was OK in both ways. Then I
After reading some docs about compiling external kernel modules:
- [Kernel source dir]/Documentation/kbuild/modules.txt
I saw some things that guided me to solve the issue:
Note: ...the kernel must have been built with modules enabled.
1. Check if the 'echo' module does not has a 'Kbuild' file
Op 21-07-10 18:05, Danny Nicholas schreef:
QOS on ISDN? Don't know how to do that.
Ron
NeoNova BV
innovatieve internetoplossingen
It's not the ISDN part, it's the internal SIP part controlled by the
network. Most calls on Asterisk consist of two or more legs; one or more
of these
Hi,
We are experiencing this issue of redial dtmf tones generated randomly in
the Voip calls, we have asterisk 1.4.21.2, dahdi 2.0.2.2 and we have dtmf as
rfc 2833, we have a cisco router at the location Cisco 2431, 8FXS (only one
FXS is used for Fax and rest are empy) connected to the netgear
Quy Pham Sy qu...@vega.com.vn writes:
they've just named as xxx.wav so I guess there is no problems with copying
or linking solutions.
You're simply lucky that the header is short enough to not sound too
bad.
/Benny
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1. Sometimes it's ok to be lucky
2. my SOX (1.14.0) on CENTOS doesn't handle alaw files.
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das sandesh sandesh...@gmail.com writes:
In the wireshark capture attached we could see the random dtmf
digits have been sent from the server side.can anyone share your
thoughts in regards to this...
Which end hears the DTMF, the SIP phones or the phones on the PSTN?
When you say
Un-top-posting and trying to regurgitate into a cohesive thread...
On Wed, 21 Jul 2010, Quy Pham Sy wrote:
I have to play a alaw file with .wav ext. How can I do this?
On Wed, 21 Jul 2010, Danny Nicholas wrote:
Asterisk won’t be “happy” trying to play foobar.wav if it is actually a
.alaw
Hi!
2. my SOX (1.14.0) on CENTOS doesn't handle alaw files.
It surely does, only that you need to tell it explicitely to:
Use -t ul or -t al and you are fine.
Philipp
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On Wed, 21 Jul 2010, Danny Nicholas wrote:
2. my SOX (1.14.0) on CENTOS doesn't handle alaw files.
Do you mean read or write?
Do you mean a raw (header-less) file containing A-LAW encoded data or
A-LAW encoded data in a WAV formatted file?
While some of the options are a bit obtuse (like
Hi!
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I will
transfer this connection, the call is mute for the extension I have
Hi Benny...
DTMF tones are heard at the SIP phones side and not the other
party...'server side' means from the Asterisk side.from the
wireshark captures it appeards that the dtmf digits were sent from the
asterisk server ip to the phone ip randomly through Cisco(just passes the
SIP packt)
Hi Elder,
I would first check the behaviour of your PSTN lines (i.e. nothing to do with
Asterisk). In many places PSTN companies allow between 30 to 90 seconds of
connection to remain open even if the -called- party, NOT the calling party,
has hung-up. Normally this is to allow putting down the
Can any good men on this group share me the firmware of a Cisco 7960 Phone?
Currently this one has Call Manager Firmware installed, I am trying to
convert it into SIP.
Much appreciated.
Apu
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On Wed, 21 Jul 2010, Apu Islam wrote:
Can any good men on this group share me the firmware of a Cisco 7960
Phone? Currently this one has Call Manager Firmware installed, I am
trying to convert it into SIP.
Wrong list. I think you were looking for Requests for Bootlegs of
Copyrighted
Apu, your best bet would be to purchase a service contract from Cisco. Then,
you'll have access to all of the firmware files that you'll ever need! :)
Rick
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
On 10-07-16 02:38 PM, Anita Hall wrote:
Is it possible to receive video calls using Asterisk and then process
them as an IVR ? One of our clients wants to set-up a video IVR system
in the US and we are evaluation possible options.
Also, what is the bandwidth of receiving a video call in US ?
Hello to all. I have succesfully received fax by app_fax, but tif files are
weird.
There a faxes sended by several fax machines to asterisk.
http://filebin.ca/hnnumf/122.tif
http://filebin.ca/ospmn/151.tif
http://filebin.ca/fzuknc/151_.tif
Any ideas how to fix this?
debug log:
On 09:06 Thu 22 Jul , Alexander Aksarin wrote:
Hello to all. I have succesfully received fax by app_fax, but tif files are
weird.
There a faxes sended by several fax machines to asterisk.
http://filebin.ca/hnnumf/122.tif
http://filebin.ca/ospmn/151.tif
http://filebin.ca/fzuknc/151_.tif
Sorry, I should have said 'Bad Men'. There are a few really bad men out
there...
:-)
On Wed, Jul 21, 2010 at 8:03 PM, Steve Edwards asterisk@sedwards.comwrote:
On Wed, 21 Jul 2010, Apu Islam wrote:
Can any good men on this group share me the firmware of a Cisco 7960
Phone? Currently
I have been facing an issue that voice is getting distorted sometimes in
MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
can't be eliminated.
I came to know
Something you may want to try (its fixed it for us) is putting an I
(uppercase I) on the asterisk invocation line.
We run servers in the cloud and can't get reliable timing from ISDN
cards etc so this instructs asterisk to generate its own internal
timing. If you have ISDN you probably don't want
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