Make sure you have
busydetect=yes
busycount=3
somewhere below your [general] context in chan_dahdi.conf (or zapata.conf
depending on your asterisk version) and restart the the service.
This should be enoough to do the magic.
Alyed
2010/3/21 Daniel Bareiro daniel-lis...@gmx.net
-BEGIN
On Sun, 2010-03-21 at 19:00 -0400, Zeeshan Zakaria wrote:
Good to know. I'll try that. I needed such solution for a client few
months ago.
On 2010-03-21 6:06 PM, Gordon Henderson gordon
+aster...@drogon.net wrote:
On Sun, 21 Mar 2010, Zeeshan Zakaria wrote:
Virtual machine
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Alyed.
On Mon, 22 Mar 2010, Alyed wrote:
I was with the following situation: if I call from a cell phone, my
Asterisk take the call, it presents to the caller the possibility to
dialing an extension number and, in case of not doing it, it
On Sun, 2010-03-21 at 21:57 +0100, mickael wrote:
I have a problem to install asterisk with ldap.
. / configure
That one should have found libldap
make menuselect
LIBS =- lldap
make This is where my error
..
/ usr / bin / ld: can not find-lldap
Do you have an /usr/lib/libldap.so ?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Mike Diehl
Sent: Thursday, March 18, 2010 1:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Polycom DHCP
Hi, I am having some trouble setting up Caller id on my asterisk system, I need
to know if there is anything special that needs to be done for an australian
connection specifically as I have tried what most web sites on google reccomend
but without success. I have not had much experience with
Hi people!
I am running Asterisk 1.6.1.12 and I set up the voicemail. That far I
have set upt the voicemailbox with my personal greeting message. If
somebody calls me and is forwarded to my mailbox, my personal recorded
greeting is played back +
the default message please record your message
Exactly what is the problem you've having with CallerID? Are you not
receiving it, or are you not able to send it? Which carrier are you
using and what make and model card is the line connected to?
For incoming calls on ISDN-10/20/30 lines, no special configuration is
required to receive caller
Since the application just does a playback of the canned sounds in
/var/lib/sounds/asterisk, you can use SOX, Audacity, etc. to mix and chop
these sounds in whatever way you see fit. Do a core set verbose 10 on the
CLI and watch the output as you leave a voicemail to see which files to
tweak.
On 3/20/2010 10:34 AM, bruce bruce wrote:
Hi Everyone,
I have a provider who is asking me to send SIP signals through
111.111.111.111 and then media through Media 1: 222.222.22.222 and Media
2: 244.244.244.244. This provider authenticates by IP and I think is
using Sonus gear and hence they
Hi
Use the us option and not b
_/List with the possible options/_
/*s* - _without_ this option a message will be played. The message by
default is: Please leave your message after the tone. When done, hang
up, or press the pound key. If you _set_ this option, the message
won#8217;t be
On 3/21/2010 8:52 AM, Leo Burd wrote:
Hello there,
I'm currently building a PHP-based software to help users make batch
calls. Basically, users provide a script and list of phone numbers.
The system calls those numbers and plays the script to whoever picks up
the phone.
Currently,
Hello,
I'm trying to call different SIP-accounts to connect them to a
conference.
This is my call-file :
Channel: SIP/test3SIP/test1
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: from-conf
Extension: 1000
I get the following in the CLI :
[Mar 22 14:40:26] -- Attempting call on
Call files are the simplest, but not the best answer. The best answer is to
use the AMI in Asyncronous mode (Don't remember the exact syntax, but it
is in the last 2-3 months of messages). The default behavior of AMI is one
at a time calling, but this flag let's you push several calls through at
On 3/21/2010 4:05 AM, Olle E. Johansson wrote:
17 mar 2010 kl. 16.37 skrev Kevin Sandy:
We're having an odd issue with codec negotiation from one of our
SIP providers. Here's the basic situation.
We receive an invite from them advertising support for G711, G729,
and G723. In our
Dear all, if I use the CustomContext module in Asterisk in order to create
new customized contexts for my extensions to managed outbound/inbound calls,
do these custom contexts replace the original context defined in sip.conf,
like context=from-internal ???
In other words, does a custom context
Not too long ago I needed to do the same thing but apparently you need to
have a separate call file for every call. The dial command didn't work with
an '' separating multiple destinations. I did it through a php script
running via agi.
On 2010-03-22 9:56 AM, jonas kellens
Dear community,
Please help. I've been looking around the internet (and in this great forum)
for help with DUNDi setup between servers (I'm using Elastix) and while I
can get my servers to lookup extensions on each other very well, I have not
been able to successfully make calls between servers.
Alejandro Cabrera Obed wrote:
Dear all, if I use the CustomContext module in Asterisk in order to
create new customized contexts for my extensions to managed
outbound/inbound calls, do these custom contexts replace the original
context defined in sip.conf, like context=from-internal ???
Yes, Custom Context is a module from FreePBX in order to define calling
routes.
Thanks.
2010/3/22 Leif Madsen leif.mad...@asteriskdocs.org
Alejandro Cabrera Obed wrote:
Dear all, if I use the CustomContext module in Asterisk in order to
create new customized contexts for my extensions to
Hello:
于 2010年03月20日 23:21, Leif Madsen 写道:
CHEN XUEQIN wrote:
I have a similar problem when using AGI for call control. Also
udp port leak for some incomplete call. I wonder if the problem
is related to issue 16774.
Only way to know would be to reproduce on a development machine, and then
We are getting ready to install a client that uses g729 when talking to their
SIP provider to minimize bandwidth usage. We are going to want to be able to
record the calls using AMI monitor actions into wav sound files. All the phones
are soft phone running on Windows XP systems.
Questions I
Hi All;
I am configuring IAX endpoint, I just need to understand why I have to set
requirecalltoken = no to be able to register because the following message is
displayed:
[Mar 22 12:25:39] ERROR[2297]: chan_iax2.c:4251 handle_call_token: Call
rejected, CallToken Support required. If
There was some security patch that changed some details of the IAX protocol.
There are now some tokens used to make the connection more secure or some such
thing. If you have some newer and older versions of asterisk that want to talk
to each other you need to tell the newer versions to not
On 22 Mar 2010, at 17:26, bilal ghayyad wrote:
From the other side: if I make requirecalltoken=no , what does it mean?
I am afraid if I make requirecalltoken=no then I will not be able to receive
a calls on my IAX client, any advise?
Alejandro Cabrera Obed wrote:
Yes, Custom Context is a module from FreePBX in order to define calling
routes.
I'd suggest using the FreePBX forums as I imagine the majority of people
responding on this list are vanilla Asterisk users.
Leif.
--
you are right, under [channels] is where it's supposed to be my mistake, i
guess i was thinking in sip.conf :)
However, the following doubt arises to me: it would also have had this
problem for some originating call from a telephone that is not a cell
phone?
yes, and this can be a really
I would like to play music to an inbound caller, AFTER asterisk answers the
call, but before the call is bridged by DIAL. Is there a simple way to
achieve this?
MD
--
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-- Bandwidth and Colocation Provided by
On 20/03/10 9:47 AM, Tzafrir Cohen wrote:
On Fri, Mar 19, 2010 at 10:50:17AM -0400, Zeeshan Zakaria wrote:
Hi Matt,
This is very useful. But what about android platforms? Will it run on it?
Just use an RSS reader. I guess browsers and RSS readers on the iPhone
are too limited.
There are a
On 20/03/10 3:46 AM, Zeeshan Zakaria wrote:
Actually I might be wrong but haven't tried it yet because the download
page is not available or the link is broken. I have however an iPhone
too to try it.
Which link is broken?
I just clicked on it from the original email, and then clicked the
This might be your answer:
Exten = s,1,answer
Exten = s,n,wait(10,m)
Exten = s,n,Dial.
This would wait 10 seconds playing MOH before dialing.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent:
There's something weird at you linux system.
Your kernel sources are i686 sources, but the output of your ls shows you
are on a X86-64 system. (?)
lrwxrwxrwx 1 root root 54 Nov 6 23:31 build -
../../../usr/src/kernels/2.6.18-164.6.1.el5-xen-x86_64
Look here!
-bash-3.2# ls
I know for a fact that you can provision a Polycom via ftp.
I've included
much of my dhcpd.conf file below. Pick out what you need.
Let me know if
I can confirm that using option 66 will work with FTP (and HTTP, for
that matter) with newer BootROM versions. I don't know the exact
version
I think I forgot some important information...
I'm actually running an AGI script after the answer (and before the dial).
I would like to play MOH while the AGI script is running, and then perform
the dial (ending the MOH).
This is where I'm stuck
Thanks!
Michelle
_
From:
Hi, it's not that simple.
It requires deep modification on asterisk and dahdi sources to work the way
you want.
Rafael Prado
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Muro, Sam
Sent:
How many simultaneous channels?
Rafael Prado
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: segunda-feira, 22 de março de 2010 2:33
To: Asterisk User MailList
Subject:
Or you can write your own application module.
Try looking at cdr_pgsql sources. ;)
Rafael Prado
+55 (11) 3323-1055
www.practis.com.br
PRACTIS - Comunicação Tecnologia
Av Aquidaban, 766 - Conj 51
CEP 13026-510, Campinas/SP - Brasil
-Original Message-
From:
I am a newbie to asterisk, I have a complete installation of
asterisk running on my ubuntu machine and I have x-lite installed also, I would
like to know if I can call myself on the same machine, because whenever I try
to call myself I get an engaged tone. My configuration file settigns are as
There will be up to 150 phones so there will be 300 channels when they are all
on the phone at one time.
I will be using a current 1.4 version.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On Mar 22, 2010, at 5:05 PM, Rafael Prado Rocchi wrote:
How many
On Mon, 2010-03-22 at 23:40 +, ayodele abejide wrote:
I am a newbie to asterisk, I have a complete installation of asterisk
running on my ubuntu machine and I have x-lite installed also, I would
like to know if I can call myself on the same machine, because
whenever I try to call myself I
I tried port 5061 for my softphone, but the same problem occurs
From: cur...@telecomabmex.com
To: asterisk-users@lists.digium.com
Date: Mon, 22 Mar 2010 17:54:27 -0600
Subject: Re: [asterisk-users] Can I call myself on the same machine
On Mon, 2010-03-22 at 23:40 +, ayodele abejide
Does this help?
The A near the end calls the audio file ginr3
exten = 551,1,Answer()
exten = 551,n,Dial(SIP/callwithus/17025551212,120,A(ginr3))
On Mon, Mar 22, 2010 at 6:41 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I think I forgot some important information...
I'm actually running an AGI
1.6.2:
-- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1,
1...@default,u) in new stack
-- DAHDI/4-1 Playing
'/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en')
[Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File
vm-intro does not exist
No...I need the audio to play while the AGI is running, but BEFORE the dial
command.
I did solve it eventually by using the Dial Local Channel...while the AGI is
running and before the second answer
Thanks
MD
-Original Message-
From: asterisk-users-boun...@lists.digium.com
/var/lib/asterisk/sounds/
On 2010-03-22 9:44 PM, sean darcy seandar...@gmail.com wrote:
1.6.2:
-- Executing [...@incoming-pstn-line:4] VoiceMail(DAHDI/4-1,
1...@default,u) in new stack
-- DAHDI/4-1 Playing
'/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en')
[Mar 22
Hello my friends,
I want to make upgrades for all my software, currently i have the following
versions:
Asterisk 1.4.21.2
Zaptel Version: 1.4.11
WANPIPE Release: 3.4.7
libpri version: 1.4.5
I want to make upgrade for the last version of Asterisk 1.4, the last
version of Zaptel (dahdi will be
If it is a production server, you should not do the upgrade on it. Setup a
new server with upgraded software, migrate all the data, test it and make
sure it works fine. There are things like CDR and voicemail which are
constantly being updated, meaning just before the final migration, you
should
--
祝您愉快！！
Aaron Chen
陈江涛
--
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
Hi!
[directories](!) ; remove the (!) to enable this
You did see the above, did you.
'/var/spool/asterisk/voicemail/default/100/unavail.gsm' (language 'en')
[Mar 22 17:15:46] WARNING[31145]: file.c:650 ast_openstream_full: File
vm-intro does not exist in any format
Did you maybe upgrade
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