Hello,
The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload
to 101.
I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload?
--
Necati DEMİR
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Hi Carlos,
you can get better idea after reading this.
http://lists.digium.com/pipermail/asterisk-users/2007-August/193347.html
Dhaval Indrodiya
On Thu, Nov 1, 2012 at 5:36 AM, Carlos Alvarez car...@televolve.com wrote:
Indeed this is getting ridiculous. This person also called me (!!)
Necati Demir wrote:
Hello,
Hola,
The service provider wants me to setup dtmfmode to rfc2833 and dtmf
payload to 101.
I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload?
Asterisk already uses payload 101 for RFC2833 so you should be fine with
dtmfmode=rfc2833
Tim Nelson wrote:
Greetings-
Hola,
I'm running into an issue as follows, in simplified form:
A remote Asterisk box, when registered/peered via SIP to a central server, and
makes a call to that central server, is *sometimes* authenticated and calls go
through properly (via from-internal
Hi,
I am configuring my asterisk server as below scenario
Jisi Astrisk-Analog PBX- Phones
For that I have Asterisk Server 1.8.1 in my PC
Digium card in my PC
TDM2400P with 5 Red (FXO) module
I install dahdi and modprobe in my system. After that i configured
chan_dahdi.conf file
- Original Message -
Tim Nelson wrote:
Greetings-
Hola,
I'm running into an issue as follows, in simplified form:
A remote Asterisk box, when registered/peered via SIP to a central
server, and makes a call to that central server, is *sometimes*
authenticated and calls go
Hello,
I installed Asterisk 11 via the following command
* svn co http://svn.asterisk.org/svn/asterisk/branches/11*
(as written in asteriskdocs.org
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html
)
But it seems that I have a development version instead of
SVN Version is always development version. Try downloading a stable tarball
archive from http://www.asterisk.org/downloads.
Regards,
Qasim
On Thu, Nov 1, 2012 at 6:52 PM, Thomas Thomas debussy...@gmail.com wrote:
Hello,
I installed Asterisk 11 via the following command
* svn co
On 10/31/2012 04:43 PM, Benny Amorsen wrote:
Jeff LaCoursiere j...@sunfone.com writes:
The basic question was has anyone made a USB FXS device work with
asterisk. Now that I have additionally defended my architecture
decisions, can anyone actually answer the question?
The Open USB FXS
Tim Nelson wrote:
Thanks Joshua-
In this case, we're using SIP registration to peer the remote systems to the
'central system'. In option #1 above, the 'user' portion is always the CID we
set for the outbound call, but the actual SIP user is something different like
'site12' for example.
31 okt 2012 kl. 15:07 skrev Giuseppe Longo giuseppe...@gmail.com:
I don't want update Asterisk configuration, i want to query LDAP only
for name and secret field.
Currently Asterisk can't do that. If you add Kamailio as a proxy in front of
Asterisk, you can
easily authenticate with LDAP
1 nov 2012 kl. 15:13 skrev Joshua Colp jc...@digium.com:
Tim Nelson wrote:
Thanks Joshua-
In this case, we're using SIP registration to peer the remote systems to the
'central system'. In option #1 above, the 'user' portion is always the CID
we set for the outbound call, but the
with this problem/if they have this issue.
Name: Voicemail
Message Number: 5
Mailbox: 1
Caller ID: S X
Caller Name: S XXX
Caller Number: X
Duration: 1:22
Date: 20121101_1116
The voice mail:
http://home.comcast.net/~jpiszcz/20121101/msg0004.WAV
Justin
: 20121101_1116
The voice mail:
http://home.comcast.net/~jpiszcz/20121101/msg0004.WAV
Justin.
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Hello,
Let's say I have a sip client that supports both G711 and G729 codecs and I
have them both enabled in sip.conf and G729 has higher priority.
Can I force the call to choose a different codec based on the dialed number
or other conditions?
For instance I would want to do G711 if the call
If you set up your DAHDI lines as users you should be able to do this.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ali Pey
Sent: Thursday, November 01, 2012 10:49 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
it happened to me once, make sure dahdi is on;
/etc/init.d/dahdi restart
Also
Chckconfig dahdi on
If all fails, I had to re-install dahdi, and Asterisk should not be a big
deal,
cd /usr/lib/asterisk/
mv modules ./old-modules-backup
And re-install Asterisk, make sure you backup your current
I am using two polycom phones to call into an asterisk box and the
console/dsp.
First phone calls in and I get connected just fine.
second phone calls in and I detect the Console/dsp is busy, and i try to use
playtones(busy) and I hear nothing. (see below)
How can I hear the tones? Thanks
i manage to make call and receive call , IVR , GSM gateway less than 100
$.https://www.facebook.com/photo.php?fbid=451312488254098set=a.435627683155912.115286.11260518622type=3theater
see the pic, and let me know if it is useful to you..it is raspberry pi and 3G
usb dongle with voice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell
Sent: Thursday, November 01, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] asterisk
On 11/01/2012 01:00 PM, Mahendra Dobariya wrote:
i manage to make call and receive call , IVR , GSM gateway less than
100 $.
https://www.facebook.com/photo.php?fbid=451312488254098set=a.435627683155912.115286.11260518622type=3theater
see the pic, and let me know if it is useful to you..
On Thu, Nov 1, 2012 at 10:25 AM, Jerry Geis ge...@pagestation.com wrote:
second phone calls in and I detect the Console/dsp is busy, and i try to
use
playtones(busy) and I hear nothing. (see below)
I experienced a similar issue in the past, where Asterisk and DAHDI seemed
to disagree about
Jeff LaCoursiere j...@sunfone.com writes:
Nifty! Love this Raspberry Pi. I keep thinking of new things I want to
do with it. If I could only clone myself. I have a video doorbell
project at the top of the list, if I don't find a USB FXS device :)
The Raspberry Pi has some problems with USB
I experienced a similar issue in the past, where Asterisk and DAHDI seemed
to disagree about my zone. In any case, try using Congestion() instead of
PlayTones(busy).
Chris,
This is very strange. I changed to Congestion() I still get nothing .
I even changed to playback(demo-congrats) and I
On 11/01/2012 04:37 PM, Benny Amorsen wrote:
Jeff LaCoursiere j...@sunfone.com writes:
Nifty! Love this Raspberry Pi. I keep thinking of new things I want to
do with it. If I could only clone myself. I have a video doorbell
project at the top of the list, if I don't find a USB FXS device :)
If I issue a dialplan reload and some AGI starts as its reloading
and directs something into the diaplan that is still reloading
what happens
I presume my context is not there?
What I see is the diaplan is messed up somehow and I goto the default
context
then after that it is messaged up
: S X
Caller Name: S XXX
Caller Number: X
Duration: 1:22
Date: 20121101_1116
The voice mail:
http://home.comcast.net/~jpiszcz/20121101/msg0004.WAV
Justin.
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