Re: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

[Eyal]

Hey,
There is a way to use the new confbridg without installing the new
version of Asterisk,
I'm new in using asterisk, my is version 1.6.2 of Asterisk. 
I would not want just to get into the installation of a new version just
for a one commend which I have very great need.

Thank you for your help.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Development Team
Sent: Thursday, November 10, 2011 6:39 PM
To: Asterisk Development Team
Subject: [asterisk-users] Asterisk 10.0.0-rc1 Now Available

The Asterisk Development Team is pleased to announce the first release
candidate
of Asterisk 10.0.0. This release candidate is available for immediate
download
at http://downloads.asterisk.org/pub/telephony/asterisk/

All Asterisk users are encouraged to participate in the Asterisk 10
testing
process. Please report any issues found to the issue tracker,
https://issues.asterisk.org/jira. It is also very useful to see
successful test
reports. Please post those to the asterisk-dev mailing list.

All Asterisk users are invited to participate in the #asterisk-testing
channel on IRC to work together in testing the many parts of Asterisk.

Asterisk 10 is the next major release series of Asterisk. It will be a
Standard support release, similar to Asterisk 1.6.2. For more
information about support time lines for Asterisk releases, see the
Asterisk
versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

A short list of features includes:

* T.38 gateway functionality has been added to res_fax.
* Protocol independent out-of-call messaging support. Text messages not
  associated with an active call can now be routed through the Asterisk
  dialplan. SIP and XMPP are supported so far.
* New highly optimized and customizable ConfBridge application capable
of mixing
  audio at sample rates ranging from 8kHz-192kHz
  (More information available at
   https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 )
* Addition of video_mode option in confbridge.conf to provide basic
video
  conferencing in the ConfBridge() dialplan application.
* Support for defining hints has been added to pbx_lua.
* Replacement of Berkeley DB with SQLite for the Asterisk Database
(AstDB).
* Much, much more!

A full list of new features can be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/10/CHANGES

For a full list of changes in the current release, please see the
ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-
10.0.0-rc1

Thank you for your continued support of Asterisk!

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] MeetMeAdmin()

[Eyal]

Hi, 

 

I use this command " exten =>
unmute,n,MeetMeAdmin(room_number,m,${caller})",

Caller is the number of the participant that I want to mute, but when
the number of participant is not in the room I get a notice in the CLI:

"NOTICE[17723]: app_meetme.c:4244 admin_exec: Specified User not found!"

And the program continues from there as it should,  

 

My question is, is there a way to use that answer in the program? Is
there a way to get a result of whether it works or not?

 

Thank you for your help.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Get return code from MeetMeAdmin()? did it possible?

[Eyal]

?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal
Sent: Thursday, November 24, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] MeetMeAdmin()

 

Hi, 

 

I use this command " exten =>
unmute,n,MeetMeAdmin(room_number,m,${caller})",

Caller is the number of the participant that I want to mute, but when
the number of participant is not in the room I get a notice in the CLI:

"NOTICE[17723]: app_meetme.c:4244 admin_exec: Specified User not found!"

And the program continues from there as it should,  

 

My question is, is there a way to use that answer in the program? Is
there a way to get a result of whether it works or not?

 

Thank you for your help.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Talk detection in meetme

[Eyal]

Hi,
I create Chat room with MEETME and now I have a problem.
I want that the host of the room could identify the participants in the
room by their speech, so that if a participant uses language the host
could kick him from the room.
Is there a way to do it?

thanks.
Eyal Mahalal

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Talk detection in meetme

[Eyal]

???

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal
Sent: Tuesday, December 06, 2011 11:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Talk detection in meetme

 

Hi,
I create Chat room with MEETME and now I have a problem.
I want that the host of the room could identify the participants in the
room by their speech, so that if a participant uses language the host
could kick him from the room.
Is there a way to do it?

thanks.
Eyal Mahalal

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] create table in mysql using asterisk

[Eyal]

Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600"
("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40),
"number_in_line" int(2), PRIMARY KEY("id")")
and this is the warning that i get in the cli:
app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL server version for the right syntax to
use near '"conference_600" ("id" int(11) NOT NULL auto_increment,
"channel_id" varchar(40)' at line 1

What is the problem do you think?
Do I in the direction or have a completely different way to do this?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] create table in mysql using asterisk

[Eyal]

Thanks
But that's not the problem, I also tried without the quotes and still
the error appears only this time it is like this
app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
failed. Error: You have an error in your SQL syntax; check the manual
that corresponds to your MySQL server version for the right syntax to
use near '' at line 1

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James
Sharp
Sent: Monday, January 09, 2012 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] create table in mysql using asterisk

On 01/09/2012 02:44 AM, Eyal wrote:
> Hi,
> I try to create a new table using MYSQL command in asterisk.
> This is what i write:
> *Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600"
> ("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40),
> "number_in_line" int(2), PRIMARY KEY("id")")*
> and this is the warning that i get in the cli:
> *app_addon_sql_mysql.c:383 aMYSQL_query: aMYSQL_query: mysql_query
> failed. Error: You have an error in your SQL syntax; check the manual
> that corresponds to your MySQL server version for the right syntax to
> use near '"conference_600" ("id" int(11) NOT NULL auto_increment,
> "channel_id" varchar(40)' at line 1
> *
> What is the problem do you think?
> Do I in the direction or have a completely different way to do this?

That's a MySQL syntax error, not an Asterisk error.  However, the 
solution is to not put quotes around your table and field names.  That 
will make MySQL happy.

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Change the caller's phone number

[Eyal]

Hi,
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew the call came from
our system.
But I have a problem with that, not all phone number change some of the
anonymous calls stay anonymous and the recipient See in the caller ID
display unlisted number.
I use this commend:
Set(CALLERID(all)="722979797" <722979797>)

Is anyone having a similar problem or know what the problem?

Thanks.


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Change the caller's phone number

[Eyal]

Thanks but it did not help resolve the situation.


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj
Khasib
Sent: Thursday, January 19, 2012 4:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Change the caller's phone number

or this 

Set(${CALLERID(all)}="722979797" <722979797>)

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Thursday, January 19, 2012 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Change the caller's phone number

try the following 
Set(${CALLERID}="722979797" <722979797>)

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal
[e...@mcr-m.com]
Sent: Thursday, January 19, 2012 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Change the caller's phone number

Hi,
I have a system that receives calls from clients and directs them to an
external phone,
before I pass on the client I change the client's phone number to a
number that I choose, so that The call recipient knew the call came from
our system.
But I have a problem with that, not all phone number change some of the
anonymous calls stay anonymous and the recipient See in the caller ID
display unlisted number.
I use this commend:
Set(CALLERID(all)="722979797" <722979797>)

Is anyone having a similar problem or know what the problem?

Thanks.


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] play sound file

[Eyal]

Hi,

How can I play a sound file from the middle and end it after a certain
number of seconds?

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] play sound file

[Eyal]

Thanks

 

But this is not what I am looking for, in this way I can start the sound
file from some point in the file but the callers must hear the file
until the end.

I need something that allows me to start from some place in the file and
end it in some other place in the file (say song from time 01:32 until
01:57),

Or

Like the controlplayback doing fast-forward but without having to click
any key by caller.

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir
Iqbal
Sent: Thursday, January 26, 2012 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] play sound file

 

check this
http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback


Nasir Iqbal

ICTBroadcast

SMS, Fax and Voice broadcasting solution

http://www.ictbroadcast.com/





On Wed, Jan 25, 2012 at 8:29 PM, Eyal  wrote:

Hi,

How can I play a sound file from the middle and end it after a certain
number of seconds?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] play sound file

[Eyal]

Thanks

Any more ideas?

 

 

אייל מהלל

מתכנת IVR

משרד: 03-6034293 שלוחה 220

נייד: 054-4793007

פקס: 03-6006081

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Wilfer
Sent: Thursday, January 26, 2012 11:24 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] play sound file

 

2012-01-26 10:11, Eyal skrev: 

Thanks

 

But this is not what I am looking for, in this way I can start the sound file 
from some point in the file but the callers must hear the file until the end.

I need something that allows me to start from some place in the file and end it 
in some other place in the file (say song from time 01:32 until 01:57),

Or

Like the controlplayback doing fast-forward but without having to click any key 
by caller.


You can do that by combining ControlPlayback and use TIMEOUT function.

If you don't want the user to be able to use any keys you can use all keys as 
stop-keys in ControlPlayback and have some logic restart the playback at 
position ${CPLAYBACKOFFSET}. 

For more details do:

core show application ControlPlayback 
core show function TIMEOUT 

/Johan






 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Iqbal
Sent: Thursday, January 26, 2012 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] play sound file

 

check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback


Nasir Iqbal

ICTBroadcast

SMS, Fax and Voice broadcasting solution

http://www.ictbroadcast.com/






On Wed, Jan 25, 2012 at 8:29 PM, Eyal  wrote:

Hi,

How can I play a sound file from the middle and end it after a certain number 
of seconds?


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 






--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello
 
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users






-- 
Johan Wilfer email: jo...@jttech.se
JT Tech | Developer  webb: http://jttech.se
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Is there a default dial plan that is not in extention.conf?

[Eyal Goltzman]

Hi,

 

I have a trivial peace of dialplan for exten 100. I try to change it to _1XX
and the asterisk act according to a different (Default??) dial plan and not
the one I want? Is that possible? Where is the other dialplan sits? In my
extention.conf I can't see something that look like what asterisk is
dialing.

How can I trace\debug my dialplan?

 

Thanks,

 

Eyal

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Is there a default dial plan that is not in extention.conf?

[Eyal Goltzman]

Thank you all,

This is what I see after CLI> dialplan show 1...@default :

  '100' =>  hint: SIP/100&IAX2/100
[pbx_config]
1. Dial(${HINT})
[pbx_config]
  '_1XX' => 1. Playback(digits/4)
[pbx_config]

>From where come the 2 first lines?? I only have the third one as the only
one under my default context at extention.conf.

And what is [pbx_config]?

Thanks

Eyal

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Friday, June 25, 2010 4:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is there a default dial plan that is not in
extention.conf?

On Fri, Jun 25, 2010 at 02:25:38PM +0300, Eyal Goltzman wrote:
> Hi,
> 
>  
> 
> I have a trivial peace of dialplan for exten 100. I try to change it to
_1XX
> and the asterisk act according to a different (Default??) dial plan and
not
> the one I want? Is that possible? Where is the other dialplan sits? In my
> extention.conf I can't see something that look like what asterisk is
> dialing.
> 
> How can I trace\debug my dialplan?

To see where it comes from, run in the Asterisk CLI:

  dialplan show 

or:

  dialplan show @

Here is a partial output from 'dialplan show' here, that shows all of
them (but is normally overly long)

[ Context 'app_queue_gosub_virtual_context' created by 'app_queue' ]
  's' =>1. NoOp() [app_queue]

[ Context 'parkedcalls' created by 'features' ]
  '700' =>  1. Park() [features]

[ Context 'app_dial_gosub_virtual_context' created by 'app_dial' ]
  's' =>1. NoOp() [app_dial]

[ Context 'from-pstn' created by 'pbx_config' ]
  '_X.' =>  1. Answer()   [pbx_config]
2. Playback(demo-instruct)[pbx_config]
3. Hangup()   [pbx_config]

[ Context 'ael-dundi-e164' created by 'pbx_ael' ]
  's' =>1. MSet(LOCAL(exten)=${ARG1}) [pbx_ael]
2. Goto(${exten},1)   [pbx_ael]
3. Return()   [pbx_ael]


'pbx_config' is dialplan that was generated from your extensions.conf. 
'pbx_ael' is dialplan that was generated from extensions.ael.
Various other modules include their own minor dialplan snippets.


'dialplan show @' also resolves various 'include=>'
directives.

If you had:

[local]
include => phones
exten => 120,1,Dial(SIP/trunk/123456)

[phones]
exten => 100,1,Dial(SIP/phone1)

the 'dialplan show local' would show the equivalent of

  include => phones
  exten => 120,1,Dial(SIP/trunk/123456)

whereas 'dialplan show 1...@local would show the actual (equivalent of)

  exten => 100,1,Dial(SIP/phone1)

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.830 / Virus Database: 271.1.1/2961 - Release Date: 06/24/10
21:35:00


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] unwanted entries created in dialplan from users.conf, how can I get rid of it?

[Eyal Goltzman]

Hello,

When I call "dialplan reload" I can see the following lines:

  == Parsing '/etc/asterisk/extensions.conf':   == Found
-- Registered extension context 'default' (0x8a72410) in local table
0x8a679d0; registrar: pbx_config
-- Added extension '_1XX' priority 1 to default (0x8a72410)
.
.
.
  == Parsing '/etc/asterisk/users.conf':   == Found
-- Added extension '100' priority -1 to default (0xb731b4e8)
-- Added extension '100' priority 1 to default (0xb731b4e8)
-- Added extension '101' priority -1 to default (0xb731b4e8)
-- Added extension '101' priority 1 to default (0xb731b4e8)

That result in a dialplan that look like that:
[ Context 'default' created by 'pbx_config' ]
'100' =>  hint: SIP/100&IAX2/100  [pbx_config]
  1. Dial(${HINT})[pbx_config]
'_1XX' => 1. Playback(digits/4)   [pbx_config]

I don't want those 2 first line to be there only the _1XX. How do I get rid
of it? Why it is added to the dial plan automatically?

extention.conf look like this:
[default]
exten => _1XX,1,Playback(digits/4)

users.cong look like this:
[100]
username = 100
transfer = yes
mailbox = 100
call-limit = 100
type = peer
fullname = Polycom
registersip = no
host = dynamic
callgroup = 1
type = peer
context = default
cid_number = 100
hasvoicemail = no
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = 100
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
macaddress = 
linenumber = 1
LINEKEYS = 1
disallow = all
allow = ulaw,gsm


Thanks,

Eyal


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Where can I find a minimal set of empty configuration files (SIP only)?

[Eyal Goltzman]

Hello,

After installing and learning Asterisk I found myself with a need for a
minimal set of empty configuration files with only the "must have" stuff in
order to setup a SIP only machine, is there a place to find it?

Thanks,

Eyal


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Using AMI Originate to call 2 outside numbers and connect them

[eyal goltzman]

Hello,

Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
How?

Thanks
Eyal
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Using AMI Originate to call 2 outside numbers and connect them

[Eyal Goltzman]

Thanks Tzafrir,

If I use the other channel as the extension parameter it will do the work
for me also?

Eyal

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Saturday, July 03, 2010 2:15 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using AMI Originate to call 2 outside numbers
and connect them

On Sat, Jul 03, 2010 at 01:33:25AM +0300, eyal goltzman wrote:
> Hello,
> 
> Can I use AMI Originate to call 2 outside numbers (SIP) and connect them?
> How?

Originate one channel to the application Dial to dial to the other
channel?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

No virus found in this incoming message.
Checked by AVG - www.avg.com 
Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/02/10
21:35:00


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP Trunk configuration problem - fromdomain

[Eyal Goltzman]

Hello,

I'm trying to register to my provider sip trunk, I got from him an host IP
(a.b.c.d) to connect to and my provider recognize me based on the fixed IP
(x.y.z.w) he gave me (no need for username and password)

In the sip.conf I add:

[mytrunk]
type=friend
insecure=no
host=a.b.c.d
fromdomain=x.y.z.w
qualify=3600
nat=no ; change to yes if you are behind NAT
bindport=5060
bindaddr=0.0.0.0
context=default
disallow=all
allow=ulaw
allow=alaw

Now, my asterisk resides in my internal network (10.100.101.107) and in the
SIP requests that sent to the provider I can see (via a sniffer) that the
"From" and "Contact" fields have - sip:aster...@10.100.101.107 and not the
x.y.z.w I expected to see as a result of the fromdomain=x.y.z.w.

Any idea?

Thanks,

Eyal


-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to change the IP in the SIP contact header

[Eyal Goltzman]

Hello,

 

I'm trying to use a SIP trunk service and the provider ask me to have the IP
address of the contact header as my public IP and not as my private one, how
can I do it?

 

See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w
is my public address:

 

sipINVITE sip:144@ a.b.c.d SIP/2.0

Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport

Max-Forwards: 70

From: "Polycom" ;tag=as7435100b

To: 

Contact: 

Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Mon, 05 Jul 2010 15:49:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 292

 

v=0

o=root 1812163927 1812163927 IN IP4 10.100.101.107

s=Asterisk PBX 1.6.1.20

c=IN IP4 10.100.101.107

t=0 0

m=audio 18848 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

 

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to change the IP in the SIP contact header

[Eyal Goltzman]

Yes, I tried and it did not solve the problem, 

 

Thanks

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie A.
Stapleton
Sent: Monday, July 05, 2010 9:05 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to change the IP in the SIP contact header

 

Have you tried setting

 

externip=

 

In the [general] of your sip.conf?

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eyal Goltzman
Sent: Monday, July 05, 2010 1:58 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] How to change the IP in the SIP contact header

 

Hello,

 

I'm trying to use a SIP trunk service and the provider ask me to have the IP
address of the contact header as my public IP and not as my private one, how
can I do it?

 

See attached the SIP invite where a.b.c.d is the SIP server IP and x.y.z.w
is my public address:

 

sipINVITE sip:144@ a.b.c.d SIP/2.0

Via: SIP/2.0/UDP 10.100.101.107:5060;branch=z9hG4bK76d52819;rport

Max-Forwards: 70

From: "Polycom" ;tag=as7435100b

To: 

Contact: 

Call-ID: 08116cf06661dc091de10c1b3315d...@84.94.96.110

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Date: Mon, 05 Jul 2010 15:49:31 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 292

 

v=0

o=root 1812163927 1812163927 IN IP4 10.100.101.107

s=Asterisk PBX 1.6.1.20

c=IN IP4 10.100.101.107

t=0 0

m=audio 18848 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

 

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2978 - Release Date: 07/05/10
09:36:00

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How can get user inputs from called party after dial?

[eyal goltzman]

Hi,
I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.

How do I do it?

If I use Dial it will not return until the end of the call, isn't it?

Thanks,

Eyal
-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How can get user inputs from called party after dial?

[Eyal Goltzman]

Thanks, but I'm missing something here, the dial command is where? 

 

I need to do something like:

Dial(1234)

Read(1 digit)

DoSomthing(based on digit from 1234)

 

And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.

 

Eyal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?

 

You need read():

http://www.voip-info.org/wiki/view/Asterisk+cmd+Read

 

It's as easy as:

 

exten => s,n,Read(variable,,11)

exten => s,n,NoOp(${variable})

 

Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.

 

-Bruce

On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman  wrote:

Hi,

I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.

 

How do I do it?

 

If I use Dial it will not return until the end of the call, isn't it?

 

Thanks,

 

Eyal

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How can get user inputs from called party after dial?

[Eyal Goltzman]

Thank you Bruce,

 

I think we are not on the same page.

 

I have an AMI script that issue an originate command, after one channel is
connected I'm in my dialplan at  extensions_custom.conf (I use FreePBX).

 

Now I'm issuing a Dial command to the another party that when he pick up the
phone I play for him a message (using the A option in the Dial command) and
then want to wait for his input, this is the case.

 

Eyal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?

 

You need to do some reading :-)

 

I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this: 

 

[first-Dialplan]

exten => s,1,Answer

exten => s,n,Playback(Welcome)

exten => s,n,Read(numb,,10)

exten => s,n,NoOp(${numb})

 

And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a "asterisk -r" and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}

 

If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.

 

-Bruce

On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman  wrote:

Thanks, but I'm missing something here, the dial command is where? 

 

I need to do something like:

Dial(1234)

Read(1 digit)

DoSomthing(based on digit from 1234)

 

And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.

 

Eyal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?

 

You need read():

http://www.voip-info.org/wiki/view/Asterisk+cmd+Read

 

It's as easy as:

 

exten => s,n,Read(variable,,11)

exten => s,n,NoOp(${variable})

 

Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.

 

-Bruce

On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman  wrote:

Hi,

I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.

 

How do I do it?

 

If I use Dial it will not return until the end of the call, isn't it?

 

Thanks,

 

Eyal

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How can get user inputs from called party after dial?

[Eyal Goltzman]

Thank you Bruce, In the below example you sent the dialplan will stop after
Dial. 

 

I found the solution to my problem in the M option of the Dial command that
let you run a macro BEFORE the parties are connected and continue the
dialplan based on the MACRO_RESULT.

 

Thanks for your help,

Eyal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?

 

For dial you do this:

 

[first-Dialplan]

exten => s,1,Answer

exten => s,n,Dial(SIP/provider/111222)

exten => s,n,Playback(Welcome)

exten => s,n,Read(numb,,10)

exten => s,n,NoOp(${numb})

 

-Bruce

 

On Sat, Jul 10, 2010 at 2:51 PM, bruce bruce  wrote:

You need to do some reading :-)

 

I will give you a quick teach here. At the end of file
/etc/asterisk/extensions_custom.conf (if you are running FreePBX) OR in
/etc/asterisk/extensions.conf (if you are running vanilla Asterisk) add
this: 

 

[first-Dialplan]

exten => s,1,Answer

exten => s,n,Playback(Welcome)

exten => s,n,Read(numb,,10)

exten => s,n,NoOp(${numb})

 

And send your inbound route to context first-Dialplan so that it's triggered
when a call comes in. Then on terminal do a "asterisk -r" and you
will see the NoOp show the DTMF number entered. From there on you can do
anything you want with the variable ${numb}

 

If any part of above is unclear to you, you must consult your friend,
google, for examples of Asterisk dialplan.

 

-Bruce

 

On Sat, Jul 10, 2010 at 2:38 PM, Eyal Goltzman  wrote:

Thanks, but I'm missing something here, the dial command is where? 

 

I need to do something like:

Dial(1234)

Read(1 digit)

DoSomthing(based on digit from 1234)

 

And as far as I understand the Dial start the call and only come back (ig
you use the g option) after call finished.

 

Eyal

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Sent: Saturday, July 10, 2010 9:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How can get user inputs from called party
after dial?

 

You need read():

http://www.voip-info.org/wiki/view/Asterisk+cmd+Read

 

It's as easy as:

 

exten => s,n,Read(variable,,11)

exten => s,n,NoOp(${variable})

 

Above will take up to 11 digits input by user and will display it back in
NoOP on Asterisk CLI.

 

-Bruce

On Sat, Jul 10, 2010 at 2:16 PM, eyal goltzman  wrote:

Hi,

I want to dial a party, play him a message and wait for his input, i.e. DTMF
digits and use them to control the rest of the dial plan.

 

How do I do it?

 

If I use Dial it will not return until the end of the call, isn't it?

 

Thanks,

 

Eyal

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

 

No virus found in this incoming message.
Checked by AVG - www.avg.com
Version: 9.0.830 / Virus Database: 271.1.1/2991 - Release Date: 07/10/10
09:36:00

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users