Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

[Floimair Florian]

For anyone interested I documented my findings as a comment to the original 
wiki article here:

https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime?refresh=1501592246325&refresh=1501592309852&refresh=1501592385401&focusedCommentId=37455051#comment-37455051



With best regards

Florian Floimair


Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian
Gesendet: Donnerstag, 13. Juli 2017 11:52
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Yeah, I will do that.

I have some more things I need to clarify, so I will just gather all the 
information and sum it up beforehand.



With best regards

Florian Floimair

Von: 
asterisk-users-boun...@lists.digium.com<https://linkprotect.cudasvc.com/url?a=https://asterisk-users-boun...@lists.digium.com&c=E,1,0XqWlrx_jzejrGNUHtTyaJK4qUM_42T294RO02AeZx1yC7S4VAhPghx0V3eABtc6iCyi2Hw94tXfogxQe7-_d4nHtaVKyDRhuV9ENoTvLdcyt5Oj&typo=1>
 
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 Im Auftrag von Marcelo Terres
Gesendet: Mittwoch, 12. Juli 2017 22:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 
https://linkprotect.cudasvc.com/url?a=https://%26lt;asterisk-users@lists.digium.com&c=E,1,Dm7JnZlS9SB0f1GpjBHWKwoW8bbEMCNe9IfvHNMjrm3moyvQ-k1T4SHRcAh35JxByJanXoMJU-ydogOJWdWr9lWzPV6UzSVJqyTbgkkfRkQs0CZdzykzYw,,&typo=1>>
Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Please open a Ticket (https://issues.asterisk.org), to let them know that they 
need to update the documentation in Wiki and also handle this situation when 
using Alembic in Debian 9 (could happens in other Distros too).

Marcelo H. Terres mailto:mhter...@gmail.com>>
IM: 
mhter...@jabber.mundoopensource.com.br<mailto:mhter...@jabber.mundoopensource.com.br>
https://www.mundoopensource.com.br<https://linkprotect.cudasvc.com/url?a=https://www.mundoopensource.com.br&c=E,1,cC98P-aUo0sGW0nm4x61IEJa3J0afB1zbPdps3H6n3PIgLTX7AqK3Tu23_zNx520Yyc8Yzs47UV9i25C0XsWCC24S8fQAXstahwvJHpIxRcmrMu0_FU,&typo=1>
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 12 July 2017 at 13:11, Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/<https://linkprotect.cudasvc.com/url?a=https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/&c=E,1,vcstDda1EXiLuEfYH_bWSEQiKUNSlxsJREZBgYMwpFTlFa1RWIuFe-eoxVvQIGuxWq4LaHyhrIGhulqolz16NPWIGtPYLbSURFxq9b0Tl5B4r3vd56NqBNAzGA,,&typo=1>).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)



With best regards

Florian Floimair


-Ursprüngliche Nachricht-
Von: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ&typo=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance

Re: [asterisk-users] Asterisk realtime in combination with ARI - error while trying to prepare SQL statement for writing into database

[Joshua Colp]

On Thu, Jul 13, 2017, at 10:01 AM, Floimair Florian wrote:
> Hey guys!
> 
> I successfully got Asterisk realtime (14.6.0) with MariaDB (MySQL fork)
> running on Debian 9.
> 
> I will document the steps to do so shortly (the main difference is
> default encoding and the odbc connector & its configuration).
> 
> What I’m trying to do now is to use ARI to create PJSIP endpoints as
> outlined in this wiki article:
> https://wiki.asterisk.org/wiki/display/AST/ARI+Push+Configuration
> 
> So far so good. Excecuting the examples using curl PUT aor and auth was
> working fine, when I ran the line for the endpoint object I got an error
> in the Asterisk CLI and log stating:



> 
> Carefully comparing the attributes from the JSON output of the curl
> statement (which was successfully completed) and the columns in the
> database I found the culprit:
> 
> The database is missing the following attribute: „dtls_fingerprint“
> I prepared my database using the alembic scripts from the 14.6.0 source.
> After manually adding the column to the database and restarting Asterisk
> the statement was successfully executed and everything works fine now.
> 
> In addition to the missing attribute there are few columns in the
> database table that do not match to a JSON attribute in the pjsip
> endpoint object.
> These are:
> 
> -  contact_deny
> 
> -  contact_permit
> 
> -  contact_user
> 
> -  deny
> 
> -  disallow
> 
> -  external_media_address
> 
> -  permit
> 
> -  redirect_method
> 
> 
> So far to my findings.
> 
> Should I file a bug-report for this?

Yes, an issue should be reported for this.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

-- 
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
  https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

[Floimair Florian]

Yeah, I will do that.

I have some more things I need to clarify, so I will just gather all the 
information and sum it up beforehand.



With best regards

Florian Floimair


Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Marcelo Terres
Gesendet: Mittwoch, 12. Juli 2017 22:55
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: Re: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Please open a Ticket (https://issues.asterisk.org), to let them know that they 
need to update the documentation in Wiki and also handle this situation when 
using Alembic in Debian 9 (could happens in other Distros too).

Marcelo H. Terres mailto:mhter...@gmail.com>>
IM: 
mhter...@jabber.mundoopensource.com.br<mailto:mhter...@jabber.mundoopensource.com.br>
https://www.mundoopensource.com.br<https://linkprotect.cudasvc.com/url?a=https://www.mundoopensource.com.br&c=E,1,cC98P-aUo0sGW0nm4x61IEJa3J0afB1zbPdps3H6n3PIgLTX7AqK3Tu23_zNx520Yyc8Yzs47UV9i25C0XsWCC24S8fQAXstahwvJHpIxRcmrMu0_FU,&typo=1>
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 12 July 2017 at 13:11, Floimair Florian 
mailto:f.floim...@commend.com>> wrote:
Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/<https://linkprotect.cudasvc.com/url?a=https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/&c=E,1,vcstDda1EXiLuEfYH_bWSEQiKUNSlxsJREZBgYMwpFTlFa1RWIuFe-eoxVvQIGuxWq4LaHyhrIGhulqolz16NPWIGtPYLbSURFxq9b0Tl5B4r3vd56NqBNAzGA,,&typo=1>).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)



With best regards

Florian Floimair


-Ursprüngliche Nachricht-
Von: 
asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>
 
[mailto:asterisk-users-boun...@lists.digium.com<mailto:asterisk-users-boun...@lists.digium.com>]
 Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 
mailto:asterisk-users@lists.digium.com>>
Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ&typo=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance



With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
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Security and Communication by Commend

FN 178618z | LG Salzburg

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Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

[Marcelo Terres]

Please open a Ticket (https://issues.asterisk.org), to let them know that
they need to update the documentation in Wiki and also handle this
situation when using Alembic in Debian 9 (could happens in other Distros
too).

Marcelo H. Terres 
IM: mhter...@jabber.mundoopensource.com.br
https://www.mundoopensource.com.br
https://twitter.com/mhterres
https://linkedin.com/in/marceloterres

On 12 July 2017 at 13:11, Floimair Florian  wrote:

> Nevermind guys!
>
> I just found out the solution myself:
>
> MariaDB in Debian uses utf8mb4 as default character set (see here:
> https://mariadb.com/kb/en/mariadb/differences-in-
> mariadb-in-debian-and-ubuntu/).
>
> I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf
> in the following files:
>
> 50-client.cnf (1 line)
> 50-mysql-clients.cnf (1 line)
> 50-server.cnf (2 lines)
>
>
>
> With best regards
>
> Florian Floimair
>
>
> -Ursprüngliche Nachricht-
> Von: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] Im Auftrag von Floimair Florian
> Gesendet: Mittwoch, 12. Juli 2017 13:50
> An: Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> Betreff: [asterisk-users] Asterisk realtime - Error with index length in
> alembic script
>
> Hi!
>
> I just tried setting up Asterisk realtime database following the wiki
> article https://wiki.asterisk.org/wiki/display/AST/Setting+up+
> PJSIP+Realtime on a Debian 9 machine (which switched from MyQSL to
> MariaDB).
>
> One has to install mariadb-plugin-connect, python-mysqldb and alembic
> packages (alembic does not work when installed via pip).
> Additionally - since MariaDB by default does not have a root user password
> set and running mysql -u root requires sudo as well - you need to execute
> the following:
> sudo mysql_secure_installation
> sudo mysql_upgrade -p --force
>
> So far so good. I run into problems when running alembic when I get to the
> following change:
> https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_
> increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjs
> CUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_
> tDVCtQerGg6RZ&typo=1
> mariadb fails this operation with error "Specified key was too long; max
> key length is 767 bytes" when it tries to increase some fields to
> varchar(255).
>
> Any idea how to solve this? Do I maybe have to switch to a different
> encoding for this to work?
>
> Thanks in advance
>
>
>
> With best regards
>
> Florian Floimair
>
> COMMEND INTERNATIONAL GMBH
> A-5020 Salzburg, Saalachstraße 51
> https://linkprotect.cudasvc.com/url?a=http://www.commend.com&c=E,1,
> AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_
> t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,&typo=1
>
> Security and Communication by Commend
>
> FN 178618z | LG Salzburg
>
> -Ursprüngliche Nachricht-
> Von: https://linkprotect.cudasvc.com/url?a=https://asterisk-
> users-boun...@lists.digium.com&c=E,1,7s7_D_Myc9BrXsqexg-
> b_jeGW99IlnqrhZCMhGKzBBE0m7-4lzl4Pqf0FBhPDU7YvysBh3XyuK7jq
> olYZryc5Pv214OOwiAf7rFVSlR6XZKzTS_0oyqQLA,,&typo=1
> https://linkprotect.cudasvc.com/url?a=https://[mailto:
> asterisk-users-boun...@lists.digium.com&c=E,1,
> b02t9WMuMstwiWAHz0XrrZjHTVSQwnEy5yxXJi5pqNE6eqJ_ZzijQ4_
> PsoLa3tnaco3BYXQ5Ck2OHfmk_Dm4EHbE77z220o2c-VzuvBbEcq7PCY,&typo=1] Im
> Auftrag von Thomas
> Gesendet: Montag, 10. Juli 2017 14:07
> An: https://linkprotect.cudasvc.com/url?a=https://asterisk-
> us...@lists.digium.com&c=E,1,LICqKTGOt1JJCqd7cLtDeAYTRlaeW-
> 0IaAjeofhcEGlqHiUa9FX1v_0Z61fjn6Cglc1LwJESdZ5CsnB1ZeUM
> Pn7gV2z5agkz3kh8onHV0Oxnmn9c4DjH6CBU&typo=1
> Betreff: [asterisk-users] ConfBridge increase talking volume as standard
>
> Hello,
>
> is it possible to increase talking volume for caller in ConfBridge as
> standard without need to press buttons after joining an conference room.
>
> best regards
> Thomas
>
> --
> _
> -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.
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> fLD5Wkjc1g637rszrIIFlYV5gEq-t1OY5td0MjI,&typo=1 --
>
> Check out the new Asterisk community forum at: https://community.asterisk.
> org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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> qWyv

Re: [asterisk-users] Asterisk realtime - Error with index length in alembic script

[Floimair Florian]

Nevermind guys!

I just found out the solution myself:

MariaDB in Debian uses utf8mb4 as default character set (see here: 
https://mariadb.com/kb/en/mariadb/differences-in-mariadb-in-debian-and-ubuntu/).

I needed to uncomment the lines with utf8mb4 in /etc/mysql/maria.db.conf in the 
following files:

50-client.cnf (1 line)
50-mysql-clients.cnf (1 line)
50-server.cnf (2 lines)

 
 
With best regards

Florian Floimair 


-Ursprüngliche Nachricht-
Von: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Floimair Florian
Gesendet: Mittwoch, 12. Juli 2017 13:50
An: Asterisk Users Mailing List - Non-Commercial Discussion 

Betreff: [asterisk-users] Asterisk realtime - Error with index length in 
alembic script

Hi!

I just tried setting up Asterisk realtime database following the wiki article 
https://wiki.asterisk.org/wiki/display/AST/Setting+up+PJSIP+Realtime on a 
Debian 9 machine (which switched from MyQSL to MariaDB).

One has to install mariadb-plugin-connect, python-mysqldb and alembic packages 
(alembic does not work when installed via pip).
Additionally - since MariaDB by default does not have a root user password set 
and running mysql -u root requires sudo as well - you need to execute the 
following:
sudo mysql_secure_installation
sudo mysql_upgrade -p --force

So far so good. I run into problems when running alembic when I get to the 
following change:
https://linkprotect.cudasvc.com/url?a=https://e96a0b8071c_increase_pjsip_column_size.py&c=E,1,dGJHzJtuX7eYDELI39tEC4ecYafZjsCUjWDL5p09DOWe28cNAbd_GFmJLD2jBZfffS-vYPvUH1CUUjR7gX1rtdvm5NFCTV_tDVCtQerGg6RZ&typo=1
mariadb fails this operation with error "Specified key was too long; max key 
length is 767 bytes" when it tries to increase some fields to varchar(255).

Any idea how to solve this? Do I maybe have to switch to a different encoding 
for this to work?

Thanks in advance

 
 
With best regards

Florian Floimair

COMMEND INTERNATIONAL GMBH
A-5020 Salzburg, Saalachstraße 51
https://linkprotect.cudasvc.com/url?a=http://www.commend.com&c=E,1,AMmXtJHsI4hmbBwyuM3M1xbRoGx-jiTaCiP7NxokUuMnadAIjb8xkI9P1ki_t2xzN78KaSN07DIpAamHT5VUTD5fbWuRqBVMk8ZAvCXz4t9wk89RgGU28EY,&typo=1

Security and Communication by Commend

FN 178618z | LG Salzburg

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Von: 
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 Im Auftrag von Thomas
Gesendet: Montag, 10. Juli 2017 14:07
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Betreff: [asterisk-users] ConfBridge increase talking volume as standard

Hello,

is it possible to increase talking volume for caller in ConfBridge as standard 
without need to press buttons after joining an conference room.

best regards
Thomas 

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Re: [asterisk-users] Asterisk Realtime RTUPDATE issue

[Joshua Colp]

Ahmed Munir wrote:

Hi,

I'm currently using Asterisk 11.7.0.The issue currently I'm facing in
Asterisk realtime sip_buddies table i.e. if I try to unregister the
extension, ipaddr, port, regseconds, fullcontact, useragent and lastms
remain still populated with data unless do the sip reload. This issue
also obser

In sip.conf the parameter I've enabled/uncommented  for realtime are
only 'rtcachefriends=yes' and rest of the realtime parameters are
commented (set as default).

Please advise, what I'm may missed out.


How exactly are you trying to unregister? A REGISTER with 0 Expires? 
What does the console show?


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Re: [asterisk-users] Asterisk realtime peer registration

[Ishfaq Malik]

On 10 June 2014 05:27,  wrote:

> Hello there
>
> I'd like to use sip users and peers realtime.
> I think I done all I need to get asterisk works fine in realtime:
>
>
> res_odbc.conf configuration.
>
> extconfig.conf
> sippeers => odbc,asterisk,sipclient
> sipusers => odbc,asterisk,sipclient
>
> sip.conf
> [general]
> rtcachefriends=yes
>
> The sipclient table as suggest in this article: SIP Realtime, MySQL table
> structure (https://wiki.asterisk.org/wiki/display/ ... +structure
> 
> )
>
> The user registered on asterisk works fine, but not the peer.
> I'd like to use my voipdiscount account as a peer to do external call.
>
> Name/username Host Dyn Forcerport ACL Port Status Realtime
> 2000/2000 xxx.xxx.xxx.xxx D N 65476 OK (117 ms) Cached RT
>
>
>
> Mysql entry on sipclient table is below:
>
> "3" " "sip.voipdiscount.com" "5060" \N "XX" \N \N \N \N "
> sip.voipdiscount.com" "peer" "default" \N \N "XXX" \N "" \N
> "rfc2833" "yes" "no" \N \N \N \N \N "port,invite" \N \N \N \N \N \N
> "01234556678" \N \N \N \N \N \N \N \N \N \N \N \N \N \N \N "
> sip.voipdiscount.com" "X" "yes" \N \N \N \N \N \N \N \N \N \N \N
> \N \N \N \N \N \N "XX" \N "voipdiscount_out" \N \N \N \N \N \N \N
> \N \N \N \N \N \N \N \N
>
> I enabled also sip debug, but I don't see any attempt towards
> sip.voipaccount.com
> What am I doing wrong?
> Someone can help me?
>
> Thanks in advance
> Pietro
>
>
>
>
> Try changing the type from peer to friend.

Regards

Ish


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Re: [asterisk-users] Asterisk Realtime Static Voicemail

[Leandro Dardini]

2013/11/11 John T. Bittner 

>  Guys,
>
>
>
> I need you help on this one.
>
>
>
> Don’t know when this broke but we have a custom gui that runs on top of
> Asterisk running a real-time static for configurations.
>
> Nothing has changed with the database other than upgrades of Asterisk 10.
>
>
>
> Customer complained that there password was not changing when they called
> into voicemail and changed it.
>
> Database is running standard ast_config with the following fields.
>
>
>
> ++--+--+-+-++
>
> | Field  | Type | Null | Key | Default | Extra  |
>
> ++--+--+-+-++
>
> | id | int(11)  | NO   | PRI | NULL| auto_increment |
>
> | cat_metric | int(11)  | NO   | | 0   ||
>
> | var_metric | int(11)  | NO   | | 0   ||
>
> | commented  | int(11)  | NO   | | 0   ||
>
> | filename   | varchar(128) | NO   | | ||
>
> | category   | varchar(128) | NO   | | default ||
>
> | var_name   | varchar(128) | NO   | | ||
>
> | var_val| varchar(255) | NO   | | ||
>
> ++--+--+-+-++
>
> 8 rows in set (0.00 sec)
>
>
>
> Did some tests and asterisk does change the password but in the
> /etc/asterisk/voicemail.conf file.
>
> Rename the file to see if it will then try the database. It recreates the
> file and changes the password.
>
> The issue is when it reads the password it looks at ast_config so it never
> really changes.
>
> Ran debug and no errors, I don’t even see it trying to update mysql
>
>
>
> Any idea what this could be.  The file below is an exact match of what’s
> in ast_config.
>
>
>
> /etc/asterisk/voicemail.conf
>
> ;! Automatically generated configuration file
>
> ;! Filename: voicemail.conf (/etc/asterisk/voicemail.conf)
>
> ;! Generator: AppVoicemail
>
> ;! Creation Date: Mon Nov 11 01:12:51 2013
>
> ;!
>
> [default]
>
> 9105 = 1234,Genee Jacobs,,,tz=|attach=|saycid=|hidefromdir=
>
> 201 = ,Anne Long,,,tz=|attach=|saycid=|hidefromdir=|delete=
>
> [zonemessages]
>
> pacific = US/Pacific|'vm-received' Q 'digits/at' IMp
>
> eastern = America/New_York|'vm-received' Q 'digits/at' IMp
>
> central = America/Chicago|'vm-received' Q 'digits/at' IMp
>
> central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
>
> military = Zulu|'vm-received' q 'digits/at' H N 'hours'
> 'phonetic/z_p'
>
> gmt = Europe/London|'vm-received' q 'digits/at' H N 'hours'
>
> cet = Europe/Zurich|'vm-received' q 'digits/at' H N 'hours'
>
> hkg = Asia/Hong_Kong|'vm-received' q 'digits/at' H N 'hours'
>
> [general]
>
> format = wav49|gsm|wav
>
> serveremail = nwvoicem...@randrealty.com
>
> attach = yes
>
> emaildateformat = %A, %B %d, %Y at %r
>
> maxlogins = 3
>
> sendvoicemail = yes
>
> operator = yes
>
> pagerdateformat = %A, %B %d, %Y at %r
>
> externnotify = /usr/local/sigman/scripts/voicemailapp
>
>
>
>
>
> John Bittner
>
> CTO
>
>  380 US Highway 46, Suite 500
>
> Totowa, NJ 07512
>
> Phone: 201.806.2602 x2405
>
> Fax:   201.806.2604
>
> Cell:   973.390.1090
>
> www.xaccel.net
>
>
>
>
>
>
> *CONFIDENTIALITY NOTICE: This e-mail message, including any attachments,
> is for the sole use of the intended recipient(s) and may contain
> confidential and privileged information which should not be shared or
> forwarded. Any unauthorized review, use, disclosure or distribution is
> prohibited. If you are not the intended recipient, please contact the
> sender by reply e-mail and destroy all copies of the e-mail.*
>
>
>
> --
>
>
Do you have compiled asterisk by yourself? In the Voicemail Build Option,
what option have you selected? I think you need to select "ODBC Storage"
and then configure ODBC on the system to connect to your database.

Leandro
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Re: [asterisk-users] Asterisk Realtime Extension... strange behaviour

[Frank]

Remove the line _X. , and try 3 digits other than 110 112 , let us know 
if it works.


On 2/12/13 5:55 AM, Yves A. wrote:

Hi,

I encountered a strange behaviour using realtime extensions... (on
Asterisk 11.2)

when I use the following static dialplan, everything works as expected..:

[from-sip]
exten =>  110,1,Dial(DAHDI/g0/${EXTEN})
exten =>  112,1,Dial(DAHDI/g0/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})
exten => _X.,1,Dial(DAHDI/g0/${EXTEN})

will say... if a sip phone calls "110" or "112" the call is routed into
PSTN (german emergency call)
if a sip phone calls any three digit number, the call should be routet
to the corresponding SIP user
and if a sip phone calls any other number the call should be routed into
PSTN... thats ok and works as expected.

when I change to realtime:
[from-sip]
switch => Realtime

and put the diaplan into the database
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_XXX""1""Dial""SIP/${EXTEN}"
"4""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"

only the emergency calls work and any other call goes to DAHDI... I cant
reach any other SIP phone.
Even when swapping the content of the rows 3 and 4 in the database to
idcontextextenpriorityappappdata
"1""from-sip""110""1""Dial""DAHDI/g0/${EXTEN}"
"2""from-sip""112""1""Dial""DAHDI/g0/${EXTEN}"
"3""from-sip""_X.""1""Dial""DAHDI/g0/${EXTEN}"
"4""from-sip""_XXX""1""Dial""SIP/${EXTEN}"

makes no difference...
I thought, using realtime extensions would read the dialplan from top to
bottom, ordered by "id"... but it
seems to be ignored somehow and the extension "_X." catches the calls
before the extensionpattern "_XXX" is reached.

I _could_ avoid this be prefixing "external" numbers with a leading 0
for example... but I dont want to... as I said.. using
static extension via extensions.conf the dialplan works as expected...

Am I missing something?

regards,
yves



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Re: [asterisk-users] asterisk realtime database structure

[Daniel-Constantin Mierla]

On 8/4/12 10:38 AM, virendra bhati wrote:

best link for asterisk realtime is below one

http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example


On Fri, Aug 3, 2012 at 1:51 PM, Leandro Dardini > wrote:


If you check the contrib/realtime/mysql directory in the source
tree, you'll find scripts for almost all the tables.


Thank you all for the hints!

Cheers,
Daniel

--
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Re: [asterisk-users] asterisk realtime database structure

[virendra bhati]

best link for asterisk realtime is below one

http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example


On Fri, Aug 3, 2012 at 1:51 PM, Leandro Dardini  wrote:

> If you check the contrib/realtime/mysql directory in the source tree,
> you'll find scripts for almost all the tables.
>
> Leandro
>
>
>
>>
>> 2012/8/3 Daniel-Constantin Mierla 
>>
>>> Hello,
>>>
>>> I was wondering if there is a tool that can create the realtime database
>>> structure for latest Asterisk version or a web resource/file containing the
>>> sql scripts. Hope I haven't missed obvious things, I had no luck searching
>>> on the web, in the wiki I found few pages with bits of sql or table
>>> structures, like:
>>>
>>> https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,**
>>> +MySQL+table+structure
>>> https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storage
>>>
>>> I have several table structures from the Asterisk 1.6, I dug for them in
>>> the code or found on the web when I wrote the tutorial about integration
>>> with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.*
>>> *1.x-asterisk-1.6.2-astdb),
>>> but hopefully now it is an easy way to get the db structure.
>>>
>>> Thanks,
>>> Daniel
>>>
>>> --
>>> Daniel-Constantin Mierla - http://www.asipto.com
>>> http://twitter.com/#!/miconda - 
>>> http://www.linkedin.com/in/**miconda
>>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>>> http://asipto.com/u/katu
>>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>>> http://asipto.com/u/kpw
>>>
>>>
>>> --
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>>>   
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>>>
>>
>>
>
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Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[C. Savinovich]

Basic?... no man, I am kid!Christian SavinovichVoIP & Telephony Consultant646-982-3572 


 Original Message 
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as
extension's next priority
From: "Raj Mathur (राज  माथुर)" <r...@linux-delhi.org>
Date: Fri, August 03, 2012 2:21 pm
To: asterisk-users@lists.digium.com

On Friday 03 Aug 2012, C. Savinovich wrote:
>You don't use 'n's in your dialplan?, you number it yourself?
> man,  what if you have a 300 line dialplan and then you decide to
> insert a new line in the middle?

If you ever used BASIC you'd remember the trick is to increment line 
numbers (priorities) by 10.  I presume a dialplan would work fine even 
if the priorities aren't sequential, as long as they're increasing 
monotonically.

Could someone confirm?

Having said that, I use n with abandon.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Eric Wieling]

Using "n" with labels is what most people do.  A dialplan isn't javascript, you 
don't need two hundred 3 line functions.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro Dardini
Sent: Friday, August 03, 2012 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; 
r...@linux-delhi.org
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as 
extension's next priority

No, numbers have to be in sequence. 

Leandro

I am typing from my mobile phone...

Il giorno 03/ago/2012 20:28, "Raj Mathur (राज माथुर)"  ha 
scritto:


On Friday 03 Aug 2012, C. Savinovich wrote:
>You don't use 'n's in your dialplan?, you number it yourself?
> man,  what if you have a 300 line dialplan and then you decide to
> insert a new line in the middle?

If you ever used BASIC you'd remember the trick is to increment line
numbers (priorities) by 10.  I presume a dialplan would work fine even
if the priorities aren't sequential, as long as they're increasing
monotonically.

Could someone confirm?

Having said that, I use n with abandon.

Regards,

-- Raj
--
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Leandro Dardini]

No, numbers have to be in sequence.

Leandro

I am typing from my mobile phone...
Il giorno 03/ago/2012 20:28, "Raj Mathur (राज माथुर)" 
ha scritto:

> On Friday 03 Aug 2012, C. Savinovich wrote:
> >You don't use 'n's in your dialplan?, you number it yourself?
> > man,  what if you have a 300 line dialplan and then you decide to
> > insert a new line in the middle?
>
> If you ever used BASIC you'd remember the trick is to increment line
> numbers (priorities) by 10.  I presume a dialplan would work fine even
> if the priorities aren't sequential, as long as they're increasing
> monotonically.
>
> Could someone confirm?
>
> Having said that, I use n with abandon.
>
> Regards,
>
> -- Raj
> --
> Raj Mathur  || r...@kandalaya.org   || GPG:
> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> It is the mind that moves   || http://schizoid.in   || D17F
>
> --
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Raj Mathur (राज माथुर)]

On Friday 03 Aug 2012, C. Savinovich wrote:
>You don't use 'n's in your dialplan?, you number it yourself?
> man,  what if you have a 300 line dialplan and then you decide to
> insert a new line in the middle?

If you ever used BASIC you'd remember the trick is to increment line 
numbers (priorities) by 10.  I presume a dialplan would work fine even 
if the priorities aren't sequential, as long as they're increasing 
monotonically.

Could someone confirm?

Having said that, I use n with abandon.

Regards,

-- Raj
-- 
Raj Mathur  || r...@kandalaya.org   || GPG:
http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves   || http://schizoid.in   || D17F

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Danny Nicholas]

The "structured" way of thinking (that cursed philosophy where you write 100
lines of code to avoid a goto) says you should have your contexts small
enough to not need "n"s.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Friday, August 03, 2012 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as
extension's next priority

 

On Fri, Aug 3, 2012 at 9:35 AM, C. Savinovich 
wrote:

 

AJ,

 

   You don't use 'n's in your dialplan?, you number it yourself? man,  what
if you have a 300 line dialplan and then you decide to insert a new line in
the middle?

 

Some might say that you should never do that.  I mean, not in one context
anyway, where the line numbers matter.

 

-- 

Carlos Alvarez

TelEvolve

602-889-3003

 

 

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Carlos Alvarez]

On Fri, Aug 3, 2012 at 9:35 AM, C. Savinovich
wrote:

>
> AJ,
>
>You don't use 'n's in your dialplan?, you number it yourself? man,
> what if you have a 300 line dialplan and then you decide to insert a new
> line in the middle?
>

Some might say that you should never do that.  I mean, not in one context
anyway, where the line numbers matter.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[C. Savinovich]

AJ,   You don't use 'n's in your dialplan?, you number it yourself? man,  what if you have a 300 line dialplan and then you decide to insert a new line in the middle?Christian SavinovichVoIP & Telephony Consultant646-982-3572 


 Original Message ----
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as
extension's next priority
From: A J Stiles <asterisk_l...@earthshod.co.uk>
Date: Fri, August 03, 2012 11:45 am
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>

On Friday 03 August 2012, C. Savinovich wrote:
> >>>Not to bash on the developer who did this I get that we don't always
> >>>think out side the box all the time
> 
> You can bash others all you want for not thinking outside the box, but
> where is your effort to think outside the box yourself?.  All you have to
> do, (that's what I did, and took me like 4 hours) is write a program that
> parses through your dialplan code and translates the n's into actual
> numbers, including the translation of the gotos into line numbers.  Sucks?
> yes.  Is the realtime limitation going to stop me from doing what I want?
> no way.

That is the sort of thing that might actually be worth submitting upstream.  
There must be loads of dialplans out there that use "same", "n" and labels all 
over the place.  The only reason mine don't, is because I've been using 
Asterisk since before these features were introduced and I got used to the old 
ways.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[A J Stiles]

On Friday 03 August 2012, C. Savinovich wrote:
> >>>Not to bash on the developer who did this I get that we don't always
> >>>think out side the box all the time
> 
> You can bash others all you want for not thinking outside the box, but
> where is your effort to think outside the box yourself?.  All you have to
> do, (that's what I did, and took me like 4 hours) is write a program that
> parses through your dialplan code and translates the n's into actual
> numbers, including the translation of the gotos into line numbers.  Sucks?
> yes.  Is the realtime limitation going to stop me from doing what I want?
> no way.

That is the sort of thing that might actually be worth submitting upstream.  
There must be loads of dialplans out there that use "same", "n" and labels all 
over the place.  The only reason mine don't, is because I've been using 
Asterisk since before these features were introduced and I got used to the old 
ways.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[C. Savinovich]

>>>Not to bash on the developer who did this I get that we don't always think out side the box all the timeYou can bash others all you want for not thinking outside the box, but where is your effort to think outside the box yourself?.  All you have to do, (that's what I did, and took me like 4 hours) is write a program that parses through your dialplan code and translates the n's into actual numbers, including the translation of the gotos into line numbers.  Sucks? yes.  Is the realtime limitation going to stop me from doing what I want? no way.Christian SavinovichVoIP & Telephony Consultant646-982-3572 


 Original Message ----
Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as
extension's next priority
From: Leandro Dardini <ldard...@gmail.com>
Date: Fri, August 03, 2012 10:11 am
To: brya...@zktech.com,  Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com>

I am kissing every inch of land where each one of the asterisk's developer is putting his feet. In the last 10 years I have worked thanks to the availability of the asterisk code. Most of my income was possible just thanks to asterisk, so I am pretty biased when trying to evaluate if the asterisk code is good or not. You can understand if I "love" the way asterisk has been coded. Nevertheless things can be better and they can be better thanks to you. Asterisk is open source and Mark is a very kind person when you submit patches, so put your ideas in new code and send to him. If you don't know how to code, hire some developer and have him to code your view of a better RT code. If it will be accepted by the core developer, all us will be happy. if it will not accepted, you'll be happy with you own personal branch. I run for a small period of time my personal asterisk tree because the italian telephony system is flawed and clients want services not suitable for the general asterisk audience, so there is nothing to worry to have your personal asterisk code. LeandroPSI think your idea of extension RT can be accomplished with some triggers and replacing the extension table with a view on your own n-enabled extension table 2012/8/3 Bryant Zimmerman <brya...@zktech.com> Leandro  I have to disagree reasonable designers would have done a better job with this one. But we developers are not always so reasonable. The issue is many developers when pushing to put features in they don't put on their designers hat and think out side the box first.Heaven knows I have been guilty of this one over the years and had to go back and refactor.  It is not so reasonable to think that this limitation has to exist developers have been putting order by fields in db driven systems for years. What of the guy who want's to use n(target) or 4(target) (I know this may have not been an option when RT was first done now it is) so they can add specialized jumping code. If I had been designing the Realtime (today) I would have added a field for the priority and made it a full alpha / numeric and added an order by field.  As it sits now how do you do n, i, h  or tags ect It kinda sucks and limits the Realtime. Not to bash on the developer who did this I get that we don't always think out side the box all the time nor was some of this ability available when the RT was written. but know it does so what do we do. Unfortunately I am not a ansi C guy or I could probably fix it .   Thanks  Bryant Zimmerman (ZK Tech Inc.)   From: "Leandro Dardini" <ldard...@gmail.com> Sent: Friday, August 03, 2012 2:18 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority   It is reasonable 'n' is not usable as priority number.  How can asterisk know the second priority if all other priority have 'n' as priority number? In a relational database there is no 'sequential read'. In other words, you need to assign the priority to all entries.  Leandro Il giorno 03/ago/2012 06:27, "virendra bhati" <virbh...@gmail.com> ha scritto:  Hi Team,  I want to used 'n' as priority in asterisk realtime but asterisk don't support n as next priority  I am using Asterisk 1.4.41   --   Thanks and regards   Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbh...@gmail.com Skype id:- virbhati2 New Delhi(India) -- _ -- 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-

Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Leandro Dardini]

I am kissing every inch of land where each one of the asterisk's developer
is putting his feet. In the last 10 years I have worked thanks to the
availability of the asterisk code. Most of my income was possible just
thanks to asterisk, so I am pretty biased when trying to evaluate if the
asterisk code is good or not. You can understand if I "love" the way
asterisk has been coded.
Nevertheless things can be better and they can be better thanks to you.
Asterisk is open source and Mark is a very kind person when you submit
patches, so put your ideas in new code and send to him. If you don't know
how to code, hire some developer and have him to code your view of a better
RT code. If it will be accepted by the core developer, all us will be
happy. if it will not accepted, you'll be happy with you own personal
branch. I run for a small period of time my personal asterisk tree because
the italian telephony system is flawed and clients want services not
suitable for the general asterisk audience, so there is nothing to worry to
have your personal asterisk code.

Leandro

PS
I think your idea of extension RT can be accomplished with some triggers
and replacing the extension table with a view on your own n-enabled
extension table

2012/8/3 Bryant Zimmerman 

> Leandro
>
> I have to disagree reasonable designers would have done a better job with
> this one. But we developers are not always so reasonable. The issue is
> many developers when pushing to put features in they don't put on their
> designers hat and think out side the box first.Heaven knows I have been
> guilty of this one over the years and had to go back and refactor.  It is
> not so reasonable to think that this limitation has to exist developers
> have been putting order by fields in db driven systems for years. What of
> the guy who want's to use n(target) or 4(target) (I know this may have not
> been an option when RT was first done now it is) so they can
> add specialized jumping code. If I had been designing the Realtime (today)
> I would have added a field for the priority and made it a full alpha /
> numeric and added an order by field.  As it sits now how do you do n, i, h
> or tags ect It kinda sucks and limits the Realtime. Not to bash on the
> developer who did this I get that we don't always think out side the box
> all the time nor was some of this ability available when the RT was
> written. but know it does so what do we do. Unfortunately I am not a ansi C
> guy or I could probably fix it .
>
> Thanks
>
> Bryant Zimmerman (ZK Tech Inc.)
>
> --
> *From*: "Leandro Dardini" 
> *Sent*: Friday, August 03, 2012 2:18 AM
> *To*: "Asterisk Users Mailing List - Non-Commercial Discussion" <
> asterisk-users@lists.digium.com>
> *Subject*: Re: [asterisk-users] Asterisk realtime don't support 'n' as
> extension's next priority
>
>  It is reasonable 'n' is not usable as priority number.  How can asterisk
> know the second priority if all other priority have 'n' as priority number?
> In a relational database there is no 'sequential read'.
>
> In other words, you need to assign the priority to all entries.
>
> Leandro
> Il giorno 03/ago/2012 06:27, "virendra bhati"  ha
> scritto:
>
>> Hi Team,
>>
>> I want to used *'n*' as priority in asterisk realtime but asterisk don't
>> support n as next priority
>>
>>
>> I am using Asterisk 1.4.41
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9718300881
>> Asterisk Developer
>> E-mail-: virbh...@gmail.com
>> Skype id:- virbhati2
>> New Delhi(India)
>>
>>
>> --
>> _
>> -- 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
>>
>
>
> --
> _
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>http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Bryant Zimmerman]

Leandro

I have to disagree reasonable designers would have done a better job with 
this one. But we developers are not always so reasonable. The issue is many 
developers when pushing to put features in they don't put on their 
designers hat and think out side the box first.Heaven knows I have been 
guilty of this one over the years and had to go back and refactor.  It is 
not so reasonable to think that this limitation has to exist developers 
have been putting order by fields in db driven systems for years. What of 
the guy who want's to use n(target) or 4(target) (I know this may have not 
been an option when RT was first done now it is) so they can add 
specialized jumping code. If I had been designing the Realtime (today) I 
would have added a field for the priority and made it a full alpha / 
numeric and added an order by field.  As it sits now how do you do n, i, h  
or tags ect It kinda sucks and limits the Realtime. Not to bash on the 
developer who did this I get that we don't always think out side the box 
all the time nor was some of this ability available when the RT was 
written. but know it does so what do we do. Unfortunately I am not a ansi C 
guy or I could probably fix it . 

Thanks

Bryant Zimmerman (ZK Tech Inc.)


 From: "Leandro Dardini" 
Sent: Friday, August 03, 2012 2:18 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: Re: [asterisk-users] Asterisk realtime don't support 'n' as 
extension's next priority

It is reasonable 'n' is not usable as priority number.  How can asterisk 
know the second priority if all other priority have 'n' as priority number? 
In a relational database there is no 'sequential read'. 

In other words, you need to assign the priority to all entries.  

Leandro Il giorno 03/ago/2012 06:27, "virendra bhati"  
ha scritto:  Hi Team,

I want to used 'n' as priority in asterisk realtime but asterisk don't 
support n as next priority

I am using Asterisk 1.4.41  
-- 

Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[A J Stiles]

On Friday 03 August 2012, virendra bhati wrote:
> Hi Team,
> 
> I want to used *'n*' as priority in asterisk realtime but asterisk don't
> support n as next priority
> 
> I am using Asterisk 1.4.41

Well, what you are wanting would be mathematically impossible!

A simple text file is read sequentially, so you can infer priorities from the 
position of lines within the file.

A database is truly random-access, and so doesn't have any such natural 
ordering.  You have to indicate the sequence explicitly within a field in each 
record, and refer to this with an ORDER BY clause.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[virendra bhati]

It means ... Asterisk don't make any IVR at realtime. It just fire
Mysql/Odbc query and get *app and appdata.*



On Fri, Aug 3, 2012 at 11:50 AM, Leandro Dardini  wrote:

> It is reasonable 'n' is not usable as priority number.  How can asterisk
> know the second priority if all other priority have 'n' as priority number?
> In a relational database there is no 'sequential read'.
>
> In other words, you need to assign the priority to all entries.
>
> Leandro
> Il giorno 03/ago/2012 06:27, "virendra bhati"  ha
> scritto:
>
>> Hi Team,
>>
>> I want to used *'n*' as priority in asterisk realtime but asterisk don't
>> support n as next priority
>>
>> I am using Asterisk 1.4.41
>>
>> --
>>
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9718300881
>> Asterisk Developer
>> E-mail-: virbh...@gmail.com
>> Skype id:- virbhati2
>> New Delhi(India)
>>
>>
>> --
>> _
>> -- 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
>



-- 

Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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Re: [asterisk-users] asterisk realtime database structure

[Leandro Dardini]

If you check the contrib/realtime/mysql directory in the source tree,
you'll find scripts for almost all the tables.

Leandro


>
> 2012/8/3 Daniel-Constantin Mierla 
>
>> Hello,
>>
>> I was wondering if there is a tool that can create the realtime database
>> structure for latest Asterisk version or a web resource/file containing the
>> sql scripts. Hope I haven't missed obvious things, I had no luck searching
>> on the web, in the wiki I found few pages with bits of sql or table
>> structures, like:
>>
>> https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,**
>> +MySQL+table+structure
>> https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storage
>>
>> I have several table structures from the Asterisk 1.6, I dug for them in
>> the code or found on the web when I wrote the tutorial about integration
>> with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.**
>> 1.x-asterisk-1.6.2-astdb),
>> but hopefully now it is an easy way to get the db structure.
>>
>> Thanks,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - 
>> http://www.linkedin.com/in/**miconda
>> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
>> http://asipto.com/u/katu
>> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
>> http://asipto.com/u/kpw
>>
>>
>> --
>> __**__**_
>> -- 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
>>
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>> To UNSUBSCRIBE or update options visit:
>>   
>> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>>
>
>
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Re: [asterisk-users] asterisk realtime database structure

[Leandro Dardini]

If you check the contrib/realtime direco

2012/8/3 Daniel-Constantin Mierla 

> Hello,
>
> I was wondering if there is a tool that can create the realtime database
> structure for latest Asterisk version or a web resource/file containing the
> sql scripts. Hope I haven't missed obvious things, I had no luck searching
> on the web, in the wiki I found few pages with bits of sql or table
> structures, like:
>
> https://wiki.asterisk.org/**wiki/display/AST/SIP+Realtime,**
> +MySQL+table+structure
> https://wiki.asterisk.org/**wiki/display/AST/ODBC+**Voicemail+Storage
>
> I have several table structures from the Asterisk 1.6, I dug for them in
> the code or found on the web when I wrote the tutorial about integration
> with Kamailio 3.1 (http://kb.asipto.com/**asterisk:realtime:kamailio-3.**
> 1.x-asterisk-1.6.2-astdb),
> but hopefully now it is an easy way to get the db structure.
>
> Thanks,
> Daniel
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - 
> http://www.linkedin.com/in/**miconda
> Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
> http://asipto.com/u/katu
> Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
> http://asipto.com/u/kpw
>
>
> --
> __**__**_
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Re: [asterisk-users] Asterisk realtime don't support 'n' as extension's next priority

[Leandro Dardini]

It is reasonable 'n' is not usable as priority number.  How can asterisk
know the second priority if all other priority have 'n' as priority number?
In a relational database there is no 'sequential read'.

In other words, you need to assign the priority to all entries.

Leandro
Il giorno 03/ago/2012 06:27, "virendra bhati"  ha
scritto:

> Hi Team,
>
> I want to used *'n*' as priority in asterisk realtime but asterisk don't
> support n as next priority
>
> I am using Asterisk 1.4.41
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9718300881
> Asterisk Developer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> New Delhi(India)
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[virendra bhati]

strange last night my serve had this issue but when next morning i check
with register 1000 sip account no issue has come

thanks for your reply

On Fri, Jul 27, 2012 at 1:30 PM, Ishfaq Malik  wrote:

> Can you please show the database entry for that peer then?
>
> On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote:
> > My sip.conf don't have any entry related to sip pees. I have
> > everything into database.
> >
> > for more details please check below url, which have good example of
> > asterisk realtime
> >
> > http://bahjons.com/stuff/asterisk-realtime-installation-guide
> >
> > On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz 
> > wrote:
> > can you post your sip.conf for  Exten. 1000?
> > it does not seem like you have
> > [1000]
> >
> > mailbox=1000@default
> >
> >
> > Thanks,
> > -motty
> >
> >
> >
> > __
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> > virendra bhati
> > Sent: Thursday, July 26, 2012 10:35 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [asterisk-users] Asterisk Realtime issue after
> > registering withx-lite
> >
> >
> >
> > Hi All,
> >
> > I have an small issue, which is not creating any problem on
> > working syatem but not sure about the problem that is why
> > eager to know about it. I had installed Asterisk realtime with
> > Asterisk 1.4.41. Every thing is working good but getting
> > warning at Asterisk CLI.
> >
> > [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via:
> > '[' is not a valid host
> > [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via:
> > '[' is not a valid host
> > [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897
> > handle_request_subscribe: Received SIP subscribe for peer
> > without mailbox: 1000
> > Really destroying SIP dialog
> > '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.'
> > Method: SUBSCRIBE
> > [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via:
> > '[' is not a valid host
> > [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via:
> > '[' is not a valid host
> > [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897
> > handle_request_subscribe: Received SIP subscribe for peer
> > without mailbox: 1000
> > Really destroying SIP dialog
> > '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.'
> > Method: SUBSCRIBE
> >
> >
> > If anyone have any suggestion please reply to me.
> >
> > --
> >
> > Thanks and regards
> >
> >  Virendra Bhati
> > +91-9718300881
> > Asterisk Developer
> > E-mail-: virbh...@gmail.com
> > Skype id:- virbhati2
> > New Delhi(India)
> >
> >
> >
> >
> > --
> >
> _
> > -- 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
> >
> >
> >
> > --
> >
> > Thanks and regards
> >
> >  Virendra Bhati
> > +91-9718300881
> > Asterisk Developer
> > E-mail-: virbh...@gmail.com
> > Skype id:- virbhati2
> > New Delhi(India)
> >
> >
> >
> > --
> > _
> > -- 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
>
> --
> Ishfaq Malik 
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
> NORTH, MANCHESTER
> SCIENCE PARK, MANCHESTER, M156SE
> COMPANY REG NO. 04920552
>
>
> --
> _
> -- 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
>



-- 

Thanks and regards

 Virendra B

Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[Ishfaq Malik]

Can you please show the database entry for that peer then?

On Thu, 2012-07-26 at 23:20 +0530, virendra bhati wrote:
> My sip.conf don't have any entry related to sip pees. I have
> everything into database.
> 
> for more details please check below url, which have good example of
> asterisk realtime
> 
> http://bahjons.com/stuff/asterisk-realtime-installation-guide
> 
> On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz 
> wrote:
> can you post your sip.conf for  Exten. 1000?
> it does not seem like you have 
> [1000]
>  
> mailbox=1000@default
>  
>  
> Thanks, 
> -motty
> 
> 
> 
> __
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> virendra bhati
> Sent: Thursday, July 26, 2012 10:35 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Asterisk Realtime issue after
> registering withx-lite
> 
> 
> 
> Hi All,
> 
> I have an small issue, which is not creating any problem on
> working syatem but not sure about the problem that is why
> eager to know about it. I had installed Asterisk realtime with
> Asterisk 1.4.41. Every thing is working good but getting
> warning at Asterisk CLI.
> 
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via:
> '[' is not a valid host
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via:
> '[' is not a valid host
> [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer
> without mailbox: 1000
> Really destroying SIP dialog
> '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.'
> Method: SUBSCRIBE
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via:
> '[' is not a valid host
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via:
> '[' is not a valid host
> [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer
> without mailbox: 1000
> Really destroying SIP dialog
> '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.'
> Method: SUBSCRIBE
> 
> 
> If anyone have any suggestion please reply to me. 
> 
> -- 
> 
> Thanks and regards
> 
>  Virendra Bhati
> +91-9718300881
> Asterisk Developer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> New Delhi(India)
> 
> 
> 
> 
> --
> _
> -- 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
> 
> 
> 
> -- 
> 
> Thanks and regards
> 
>  Virendra Bhati
> +91-9718300881
> Asterisk Developer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> New Delhi(India)
> 
> 
> 
> --
> _
> -- 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

-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[virendra bhati]

My sip.conf don't have any entry related to sip pees. I have everything
into database.

for more details please check below url, which have good example of
asterisk realtime

http://bahjons.com/stuff/asterisk-realtime-installation-guide

On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz  wrote:

> **
> can you post your sip.conf for  Exten. 1000?
> it does not seem like you have
> [1000]
>
> mailbox=1000@default
>
> **
> *Thanks, *
> *-motty*
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
> *Sent:* Thursday, July 26, 2012 10:35 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Asterisk Realtime issue after registering
> withx-lite
>
>  Hi All,
>
> I have an small issue, which is not creating any problem on working syatem
> but not sure about the problem that is why eager to know about it. I had
> installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
> good but getting warning at Asterisk CLI.
>
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer without mailbox:
> 1000
> Really destroying SIP dialog
> '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
> SUBSCRIBE
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer without mailbox:
> 1000
> Really destroying SIP dialog
> '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
> SUBSCRIBE
>
>
> If anyone have any suggestion please reply to me.
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9718300881
> Asterisk Developer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> New Delhi(India)
>
>
> --
> _
> -- 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
>



-- 

Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
--
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Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[motty.cruz]

can you post your sip.conf for  Exten. 1000?
it does not seem like you have 
[1000]
 
mailbox=1000@default
 
 
Thanks, 
-motty

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, July 26, 2012 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Realtime issue after registering
withx-lite


Hi All,

I have an small issue, which is not creating any problem on working syatem
but not sure about the problem that is why eager to know about it. I had
installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
good but getting warning at Asterisk CLI.

[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE


If anyone have any suggestion please reply to me. 

-- 


Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)


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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Matt Riddell]

On 29/09/11 12:53 AM, Nick Khamis wrote:

Hello David,

I have this discussion also on the -dev mailing list. and suggested
that we use a database hook to trigger the originate process (pleasee
see "Outbound Call Implementation"). However, compiling it directly
into asterisk as a realtime moodule insted of using AMI etc...


I'm pretty sure nobody recommended you compile it directly into Asterisk :-)

Asterisk Realtime does not support polling a database.  It is for 
loading configuration from.  Basically it is a way to replace 
(supplement) the config files.


Originating a call is not a config file.

Call origination happens via the following means:

1. Asterisk Manager
2. Call files
3. VoIP/Analogue Phone/Line (includes GoogleTalk etc)

What you're wanting to do (initiate a call from a database record) is 
something that you would be better doing externally and using an 
existing trigger for the actual call (i.e. one of the above).


It's like saying you want to write an application to be built into 
Asterisk to initiate a call and play a message.  You'd instead use one 
of the above, then the Playback or Background application to play the 
message.


--
Cheers,

Matt Riddell
___

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello David,

I have this discussion also on the -dev mailing list. and suggested
that we use a database hook to trigger the originate process (pleasee
see "Outbound Call Implementation"). However, compiling it directly
into asterisk as a realtime moodule insted of using AMI etc...

Cheers,

Nick.



On Wed, Sep 28, 2011 at 1:23 AM, Sam Govind  wrote:
> Correct me if I'm wrong or don't know anything other than AMI Originate
> Event or a call file to kick start a call from asterisk ! So making a new or
> modifying asterisk call-file cron job/poller seems like a nice idea but why
> put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
> The simple idea is create a MySQL trigger for your Table insertion, the data
> in the table at insertion time becomes parameters for a simple script that
> triggers an AMI event (or call file) whichever is easier for you.
>
> On Tue, Sep 27, 2011 at 6:54 PM, Nick Khamis  wrote:
>>
>> Hello David,
>>
>> At first I assumed asterisk used call files out of the box for
>> normal-initiated/instantiated calls however,
>> this is incorrect. I think call files was the easy approach for client
>> just to place a file with call details
>> in some location. I am trying to do the same with a db record. My
>> first question is, how does asterisk
>> initiate calls, i.e. what part of the source code is responsible for
>> that. Are there any threads involved etc.
>>
>> Cheers,
>>
>> Nick.
>>
>> On Tue, Sep 27, 2011 at 9:35 AM, David Moring 
>> wrote:
>> > Hi Nick,
>> >
>> > Understand your reasoning - though as Matt points out sql db isn't in
>> > the
>> > core so compiling it there would preclude seemless upgrades.  Also, I
>> > personally would be concerned putting the calls right into the call-file
>> > thread might create an issue if you hung on a db query or insert.
>> >  Finally
>> > (and I'd love to hear the answer not knowing), but I believe
>> > "normally-initiated/instantiated" calls are handled with direct calls
>> > via
>> > either SIP requests and/or AMI - thus even using the proposed method, I
>> > *think* the db/file-drop method is going to create some overhead that
>> > might
>> > not scale well...
>> >
>> > Best,
>> >
>> > David
>> >
>> > -Original Message-
>> > From: Nick Khamis 
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > 
>> > Date: Mon, 26 Sep 2011 18:49:07 -0400
>> > Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
>> >
>> > Hello David,
>> >
>> > Thank you so much for your response. I am sure it can be easily done
>> > using AGI. The reason I am leaning more
>> > towards storing the call information in a database record, is because
>> > our existing client applications can be easily
>> > modified to write to MySQL. The asterisk cron/thread that would
>> > querying the DB should be no different than existing implementation
>> > used process the call files?
>> > For those of you that may be interested in what we are doing. We are
>> > developing an application that will apply NLP
>> > services on text generated using the speech to text module, and
>> > generate the response that will then be forwarded to
>> > the text to speech.
>> >
>> > Cheers,
>> >
>> > Nick
>> > .
>> >
>> >
>> >
>> > --
>> > _
>> > -- 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
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Correct me if I'm wrong or don't know anything other than AMI Originate
Event or a call file to kick start a call from asterisk ! So making a new or
modifying asterisk call-file cron job/poller seems like a nice idea but why
put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
The simple idea is create a MySQL trigger for your Table insertion, the data
in the table at insertion time becomes parameters for a simple script that
triggers an AMI event (or call file) whichever is easier for you.

On Tue, Sep 27, 2011 at 6:54 PM, Nick Khamis  wrote:

> Hello David,
>
> At first I assumed asterisk used call files out of the box for
> normal-initiated/instantiated calls however,
> this is incorrect. I think call files was the easy approach for client
> just to place a file with call details
> in some location. I am trying to do the same with a db record. My
> first question is, how does asterisk
> initiate calls, i.e. what part of the source code is responsible for
> that. Are there any threads involved etc.
>
> Cheers,
>
> Nick.
>
> On Tue, Sep 27, 2011 at 9:35 AM, David Moring 
> wrote:
> > Hi Nick,
> >
> > Understand your reasoning - though as Matt points out sql db isn't in the
> > core so compiling it there would preclude seemless upgrades.  Also, I
> > personally would be concerned putting the calls right into the call-file
> > thread might create an issue if you hung on a db query or insert.
>  Finally
> > (and I'd love to hear the answer not knowing), but I believe
> > "normally-initiated/instantiated" calls are handled with direct calls via
> > either SIP requests and/or AMI - thus even using the proposed method, I
> > *think* the db/file-drop method is going to create some overhead that
> might
> > not scale well...
> >
> > Best,
> >
> > David
> >
> > -Original Message-----
> > From: Nick Khamis 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > Date: Mon, 26 Sep 2011 18:49:07 -0400
> > Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
> >
> > Hello David,
> >
> > Thank you so much for your response. I am sure it can be easily done
> > using AGI. The reason I am leaning more
> > towards storing the call information in a database record, is because
> > our existing client applications can be easily
> > modified to write to MySQL. The asterisk cron/thread that would
> > querying the DB should be no different than existing implementation
> > used process the call files?
> > For those of you that may be interested in what we are doing. We are
> > developing an application that will apply NLP
> > services on text generated using the speech to text module, and
> > generate the response that will then be forwarded to
> > the text to speech.
> >
> > Cheers,
> >
> > Nick
> > .
> >
> >
> >
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> >
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello David,

At first I assumed asterisk used call files out of the box for
normal-initiated/instantiated calls however,
this is incorrect. I think call files was the easy approach for client
just to place a file with call details
in some location. I am trying to do the same with a db record. My
first question is, how does asterisk
initiate calls, i.e. what part of the source code is responsible for
that. Are there any threads involved etc.

Cheers,

Nick.

On Tue, Sep 27, 2011 at 9:35 AM, David Moring  wrote:
> Hi Nick,
>
> Understand your reasoning - though as Matt points out sql db isn't in the
> core so compiling it there would preclude seemless upgrades.  Also, I
> personally would be concerned putting the calls right into the call-file
> thread might create an issue if you hung on a db query or insert.  Finally
> (and I'd love to hear the answer not knowing), but I believe
> "normally-initiated/instantiated" calls are handled with direct calls via
> either SIP requests and/or AMI - thus even using the proposed method, I
> *think* the db/file-drop method is going to create some overhead that might
> not scale well...
>
> Best,
>
> David
>
> -Original Message-
> From: Nick Khamis 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 26 Sep 2011 18:49:07 -0400
> Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
>
> Hello David,
>
> Thank you so much for your response. I am sure it can be easily done
> using AGI. The reason I am leaning more
> towards storing the call information in a database record, is because
> our existing client applications can be easily
> modified to write to MySQL. The asterisk cron/thread that would
> querying the DB should be no different than existing implementation
> used process the call files?
> For those of you that may be interested in what we are doing. We are
> developing an application that will apply NLP
> services on text generated using the speech to text module, and
> generate the response that will then be forwarded to
> the text to speech.
>
> Cheers,
>
> Nick
> .
>
>
>
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> To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello Matt,

Thanks again for your response. This all makes perfect sense. Much
like sip friends, moh, extensions etc..., have ability to process both
existing config files and realtime configurations, the dial app should
be able to do the same, leaving the current functionality untouched. I
have no plans to commit the changes however, would be more than happy
to submit it for review and possible inclusion in the source tree.
AGI is the obvious choice for most, but I am a little interested in
the internals of asterisk. Honestly speaking I have not looked into
the source code yet, and was just posting to the group for some
insight in the implementation in charge of the dial app.

Nick.

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello David,

Thank you so much for your response. I am sure it can be easily done
using AGI. The reason I am leaning more
towards storing the call information in a database record, is because
our existing client applications can be easily
modified to write to MySQL. The asterisk cron/thread that would
querying the DB should be no different than existing implementation
used process the call files?
For those of you that may be interested in what we are doing. We are
developing an application that will apply NLP
services on text generated using the speech to text module, and
generate the response that will then be forwarded to
the text to speech.

Cheers,

Nick.

On Mon, Sep 26, 2011 at 6:29 PM, David Moring  wrote:
> Nick,
>
> I just created a .NET dialer for a client not too long ago and it didn't
> take more than a few hours - it doesn't use a db (thinking that's an awful
> lot of empty polling waiting for a call when the call will be records in the
> call records anyway), but simply connects/logs in and kicks it off through
> the AMI... not difficult really so I'd just study up on AMI first since
> that's the "actuator"... My .02 anyway
>
> DMoring
>
> -Original Message-
> From: Nick Khamis 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 26 Sep 2011 18:04:07 -0400
> Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
>
> Hello Matt,
>
> Thank you so much for your response. The first two are not a problem.
> I would rather write the app in C and compile it directly into
> asterisk. Are there
> any direction on how to work with the dial app? Any types of threads *
> may use etc...
>
> Thanks in Advance.
>
> Nick Khamis
> Ph.D Computer Science
> McGill University
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Matt Riddell]

On 27/09/11 11:03 AM, Danny Nicholas wrote:

Will you be recording this presentation for those of us who can't get to
Astricon?


Dunno whether they'll be recording - I haven't done an Astricon 
presentation for a few years now :-)


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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Matt Riddell]

On 27/09/11 11:04 AM, Nick Khamis wrote:

Hello Matt,

Thank you so much for your response. The first two are not a problem.
I would rather write the app in C and compile it directly into
asterisk. Are there
any direction on how to work with the dial app? Any types of threads *
may use etc...


I wouldn't compile it directly into Asterisk for a few reasons:

1. It's not likely to ever get accepted into the mainstream codebase as 
it would require a dependency on MySQL.  This means you would 
continually need to adapt your code as Asterisk changes.


2. If you were to write something that could be accepted into the 
mainstream code base, it would need to support multiple database systems 
and would likely need to be integrated with Asterisk realtime.


3. If there was a problem in the code it would bring down Asterisk, 
whereas if it was external it would just bring down the app.


4. If you ever wanted to support a cluster of Asterisk machines it would 
likely require significant changes.


If the app were written in C but external to Asterisk you would overcome 
all of these.  The only exception being that you would want to check the 
Asterisk Manager version when connecting.


You could then make the app GPL or whatever if you wanted to distribute 
it and then other people could help maintain it.


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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello Matt,

Thank you so much for your response. The first two are not a problem.
I would rather write the app in C and compile it directly into
asterisk. Are there
any direction on how to work with the dial app? Any types of threads *
may use etc...

Thanks in Advance.

Nick Khamis
Ph.D Computer Science
McGill University

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Danny Nicholas]

Will you be recording this presentation for those of us who can't get to
Astricon?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, September 26, 2011 5:01 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App

On 27/09/11 10:51 AM, Danny Nicholas wrote:
> Matt - how dare you tell a man asking for a fish to learn how!

:-)  It would have taken way too much time to explain all the steps.

Although, having said that I am doing a tutorial at Astricon on how to use
the Asterisk Manager, so if he's at Astricon I could teach him one of the
steps :-)

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Matt Riddell]

On 27/09/11 10:51 AM, Danny Nicholas wrote:

Matt - how dare you tell a man asking for a fish to learn how!


:-)  It would have taken way too much time to explain all the steps.

Although, having said that I am doing a tutorial at Astricon on how to 
use the Asterisk Manager, so if he's at Astricon I could teach him one 
of the steps :-)


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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Danny Nicholas]

Matt - how dare you tell a man asking for a fish to learn how!

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Monday, September 26, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App

On 27/09/11 2:57 AM, Nick Khamis wrote:
> That would be amazing! And would allow for more possibilities. A call 
> is made with the simple insertion of new call data. Some directions on 
> this please?

Just write a program that polls the database, if it sees a record it runs an
Originate command via the Asterisk Manager.

Things you would need to learn:

1. The language you use for the program (i.e. C/PHP/Java etc) 2. How to
create/use MySQL databases 3. How the Asterisk Manager works

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Matt Riddell]

On 27/09/11 2:57 AM, Nick Khamis wrote:

That would be amazing! And would allow for more possibilities. A call
is made with the simple insertion of new call data. Some directions on
this please?


Just write a program that polls the database, if it sees a record it 
runs an Originate command via the Asterisk Manager.


Things you would need to learn:

1. The language you use for the program (i.e. C/PHP/Java etc)
2. How to create/use MySQL databases
3. How the Asterisk Manager works

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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

Hello Ronald,

Thank you so much for your response. I forgot to mentions that I also
have sip_extensions working realtime, and storing dialplans, as well
as sip_friends, moh, voice mail etc... I was referring to the "call
data", that I think is stored in a call file? The actual mechanism
that fires asterisk to make a call? Moving that cron to process call
data stored in a record in a table called for example sip_calls? Sam,
had a suggestion of using AGI, but I just though of the idea of
modifying the asterisk cron that is in charge of making calls upon
creation of the call file, to an entry in a table? I hope I am
explaining this correctly.

Thanks in Advance,

Nick.

On Mon, Sep 26, 2011 at 2:05 AM, Ronald Cepres  wrote:
> Hi Nick,
> You mean if it is possible for Asterisk to use realtime dialplan? If it is,
> AFAIK it is possible using a table format for realtime extensions.
> Regards,
> Ronald
>
> On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind  wrote:
>>
>> Hmmm..interesting..I haven't came across anything like this so far..How
>> about making a new table for the insertion of a new call data..and trigger
>> some script to activate AMI/Call file according to new call data.
>>
>> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>>
>> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:
>>>
>>> Hello Everyone,
>>>
>>> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
>>> wondering if it Is possible to have Asterisk make a calls based on a
>>> record inserted in a table realtime? If I have to develop something using
>>> AGI
>>> or AMI, I can do this  with a little direction?
>>>
>>> Thanks in Advance,
>>>
>>> Nick
>>>
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>>
>>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Nick Khamis]

That would be amazing! And would allow for more possibilities. A call
is made with the simple insertion of new call data. Some directions on
this please?

Cheers,

Nick.

On Mon, Sep 26, 2011 at 1:33 AM, Sam Govind  wrote:
> Hmmm..interesting..I haven't came across anything like this so far..How
> about making a new table for the insertion of a new call data..and trigger
> some script to activate AMI/Call file according to new call data.
> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>
> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:
>>
>> Hello Everyone,
>>
>> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
>> wondering if it Is possible to have Asterisk make a calls based on a
>> record inserted in a table realtime? If I have to develop something using
>> AGI
>> or AMI, I can do this  with a little direction?
>>
>> Thanks in Advance,
>>
>> Nick
>>
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Bruce Ferrell]

One thing I do is to use the mysql command to run simple queries and the
Goto a context with the information... simple and clean

On 09/25/2011 10:33 PM, Sam Govind wrote:
> Hmmm..interesting..I haven't came across anything like this so
> far..How about making a new table for the insertion of a new call
> data..and trigger some script to activate AMI/Call file according to
> new call data.
>
> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>
> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  > wrote:
>
> Hello Everyone,
>
> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
> wondering if it Is possible to have Asterisk make a calls based on a
> record inserted in a table realtime? If I have to develop
> something using AGI
> or AMI, I can do this  with a little direction?
>
> Thanks in Advance,
>
> Nick
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Ronald Cepres]

Hi Nick,

You mean if it is possible for Asterisk to use realtime dialplan? If it is,
AFAIK it is possible using a table format for realtime extensions.

Regards,
Ronald

On Mon, Sep 26, 2011 at 1:33 PM, Sam Govind  wrote:

> Hmmm..interesting..I haven't came across anything like this so far..How
> about making a new table for the insertion of a new call data..and trigger
> some script to activate AMI/Call file according to new call data.
>
>
> http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10
>
>
> On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:
>
>> Hello Everyone,
>>
>> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
>> wondering if it Is possible to have Asterisk make a calls based on a
>> record inserted in a table realtime? If I have to develop something using
>> AGI
>> or AMI, I can do this  with a little direction?
>>
>> Thanks in Advance,
>>
>> Nick
>>
>> --
>> _
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>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Hmmm..interesting..I haven't came across anything like this so far..How
about making a new table for the insertion of a new call data..and trigger
some script to activate AMI/Call file according to new call data.

http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10

On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:

> Hello Everyone,
>
> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
> wondering if it Is possible to have Asterisk make a calls based on a
> record inserted in a table realtime? If I have to develop something using
> AGI
> or AMI, I can do this  with a little direction?
>
> Thanks in Advance,
>
> Nick
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] asterisk realtime & calling sip users

[Sherwood McGowan]

No problem mate, I was kinda wondering. That error message seemed familiar

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Re: [asterisk-users] asterisk realtime & calling sip users

[Nick Ustinov]

after some deep tracing it turned out to be a faulty router problem

thanks.



On Sun, Dec 26, 2010 at 9:38 AM, Sherwood McGowan
 wrote:
> On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov  wrote:
>> Hello
>>
>> We have recently upgraded to Realtime engine (sip buddies and
>> extensions) and now have problems with calling local SIP users.
>> I have rtcachefriends=yes but tried with 'no' and it's even worse.
>> (asterisk 1.8.1.1 + realtime mysql)
>>
>> Here's an example:
>>
>> User 1000 registers successfully and can then be called with
>> Dial(SIP/1000,30) successfully
>>
>> After some time when I try to call this user the asterisk just keeps
>> hanging until timeout occurs:
>>
>> -- Calling 1000
>>
>> and the debug says:
>>
>> [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: ** SIP timers:
>> Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id
>> #1213))
>> [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: Trying to put 'INVITE
>> sip:' onto UDP socket destined for 78.84.202.65:48406
>> [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: ** SIP timers:
>> Rescheduling retransmission 5 to 8000 ms (t1 500 ms (Retrans id
>> #1213))
>> [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: Trying to put 'INVITE
>> sip:' onto UDP socket destined for 78.84.202.65:48406
>>
>> however if i do 'sip show peers' it shows the peer normally:
>>
>> 1000/nlcyhguv 78.84.202.65 D N 34817 Unmonitored Cached RT
>>
>>
>> User 1000 has nat=yes and is behind NAT.
>> Before we moved to Realtime it all used to work well.
>>
>>
>> Any advice would be appreciated.
>>
>> Thanks in advance,
>> Nick
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>
> This is really odd, could you do use a favor and show the full output
> of "sip show peer 1000 load"? I want to see it to better help you.
>
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Re: [asterisk-users] asterisk realtime & calling sip users

[Sherwood McGowan]

On Sat, Dec 25, 2010 at 11:28 AM, Nick Ustinov  wrote:
> Hello
>
> We have recently upgraded to Realtime engine (sip buddies and
> extensions) and now have problems with calling local SIP users.
> I have rtcachefriends=yes but tried with 'no' and it's even worse.
> (asterisk 1.8.1.1 + realtime mysql)
>
> Here's an example:
>
> User 1000 registers successfully and can then be called with
> Dial(SIP/1000,30) successfully
>
> After some time when I try to call this user the asterisk just keeps
> hanging until timeout occurs:
>
> -- Calling 1000
>
> and the debug says:
>
> [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: ** SIP timers:
> Rescheduling retransmission 4 to 4000 ms (t1 500 ms (Retrans id
> #1213))
> [2010-12-24 12:30:11] DEBUG[12870] chan_sip.c: Trying to put 'INVITE
> sip:' onto UDP socket destined for 78.84.202.65:48406
> [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: ** SIP timers:
> Rescheduling retransmission 5 to 8000 ms (t1 500 ms (Retrans id
> #1213))
> [2010-12-24 12:30:15] DEBUG[12870] chan_sip.c: Trying to put 'INVITE
> sip:' onto UDP socket destined for 78.84.202.65:48406
>
> however if i do 'sip show peers' it shows the peer normally:
>
> 1000/nlcyhguv 78.84.202.65 D N 34817 Unmonitored Cached RT
>
>
> User 1000 has nat=yes and is behind NAT.
> Before we moved to Realtime it all used to work well.
>
>
> Any advice would be appreciated.
>
> Thanks in advance,
> Nick
>
> --
> _
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>

This is really odd, could you do use a favor and show the full output
of "sip show peer 1000 load"? I want to see it to better help you.

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Re: [asterisk-users] Asterisk Realtime Billing Question???

[Sherwood McGowan]

Tarek,

Ouch, I'm quite sorry. I couldn't sleep when I tried to around 4:30AM
after working on a project all night. Unfortunately, I'm not quite
sure what your question was...

:( Maybe when I wake up a bit more


On Thu, Oct 21, 2010 at 5:38 AM, Tarek Sawah  wrote:
>
> actually my mail was not meant to be disrespectful. it was an inquiry. i have 
> a billing system and had a few of those thoughts regarding real time billing. 
> my issue was explaining to a customer that his call disconnected an hour 
> earlier because someone else used his account.. I'm doing retail not 
> wholesale, you may understand my question more clearly now?
>
> Regards
>
>
> Tarek Sawah
>
> Information Technology Adviser
>
> Integrated Digital Systems
>
> CCNP, MCSE, RHCE, TELECOM
>
> USA: +1 386 492 9993
>
>
>
>
>
>
>
> 
>> From: sherwood.mcgo...@gmail.com
>> Date: Thu, 21 Oct 2010 05:18:17 -0500
>> To: asterisk-users@lists.digium.com
>> Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
>>
>> Tarek,
>>
>> I'm not sure why it would be our problem is someone came into your
>> office and started making long distance calls over a trunk I was
>> providing your company I'm pretty sure that if I had tried that
>> with some of my carriers in the past they would have laughed until
>> they cried...
>>
>> Oh, and also, since this was a wholesale carrier, the customers were
>> in control of their own freeze amount. It was there to allow THEM to
>> control their account better. I'd be willing to bet that my clients
>> would have been happy to just keep billing them for every minute they
>> used.
>>
>> Lastly, I would like to just say, I'm not the guy who requested the
>> feature, I'm the guy who figured out how to make it happen, and making
>> it happen back in early 2006, when the MySQL addon was just BARELY
>> stable...
>>
>> It's ok, I don't need respect, I have the knowledge that I'm the mick,
>> and I'm awesome :P
>>
>> Cheers :D
>>
>> On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah  wrote:
>> > If you look at it the way you want it.. you usually tell your customer the
>> > available funds and minutes in their account right?
>> > How will you explain "politely" that you have dropped their calls for lack
>> > of balance because someone else used their account?
>> > If you don't tell them their balance and call duration before call .. then
>> > that won't be a problem.
>> > Now you can do some kind of script to do the math and disconnect calls when
>> > balance is over.
>> >
>> > From: asterisk-users-boun...@lists.digium.com
>> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
>> > INDRODIYA
>> > Sent: Thursday, October 21, 2010 9:31 AM
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
>> >
>> > Hi Sherwood ,
>> >
>> > well , i think you did not understand my question , i want real time 
>> > billing
>> > like as i mentioned that if i want to dial 5 number with different call 
>> > rate
>> > how can i access same
>> > balance into those 5 people, if all are connected how can i periodically
>> > update billing , as you suggested it will assign total balance to those 5
>> > people but actually we can not do like this as total balance of user $100 ,
>> > as per your suggestion it will give $100 for those 5 people which is
>> > practically wrong i think.
>> >
>> > give your thougts.
>> >
>> > regards
>> > dhaval
>> > On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
>> >  wrote:
>> > On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
>> >  wrote:
>> > Hello All,
>> >
>> > after so long time i posted a new question regarding billing, hope  anyone
>> > have some solution.
>> >
>> > I have situation in that i want to do billing of more than 1 call in real
>> > time below are scenario and explanation.
>> >
>> >
>> > Scenario:
>> >  A customer called my DID number and after that from here i dial few number
>> > let say 5 number. once number are placed into DIAL
>> > i will put this customer into conference [MEETME] , once a Members are
>> > picked up call they will also patched into conferen

Re: [asterisk-users] Asterisk Realtime Billing Question???

[Jeff LaCoursiere]

[snipped very confusing top and bottom posting mix]

On Thu, 21 Oct 2010, Sherwood McGowan wrote:

> Dhaval,
> 
> You're right, I forgot one thing. The "frozen" table's id column should not
> be an autoincrement, it should be set by the insert statement, using the
> original method I decsribed for creating a unique integer from the callerid
> number and the current EPOCH. That way, you can be sure that multiple
> concurrent calls that have frozen funds will only retrieve the record they
> created. (Oh and, once you thaw the frozen funds, delete the appropriate
> record in the frozen table)
> 
> I'm not sure why you think this will only work for a single call at a time.
> Each time a call occurs that is related to an account will cause more money
> to be "frozen" from that account, thereby causing future calls to have less
> available balance and therefore less time for a call limit. This works for
> ANY number of concurrent calls on an account, and every one of those calls
> freezes funds based on the rate at which THAT call's amount to freeze was
> calculated against.
> 
> EACH call determines IT'S rate, which is then used to determine the amount
> to freeze from the account ON THAT CALL. Additionally, since the rate is
> specific to each call, the limiting of the length of THAT call, your issue
> of limiting is also a non-issue.
>

I also have worked on the logic for this scenario, and I gave up.  Our calling 
card system now locks a balance and forces the account to one simultaneous call 
at a time.  We report the maximum length of a call to the customer just 
before the ringing starts, and as someone else stated - to cut it off 
prematurely is very confusing to the customer (and one of the number one 
complaints against calling cards - if you sell in Florida it could 
actually get you in serious trouble).

The problem with each call freezing a portion of the balance is that no one 
call has access to the whole balance, and that was determined (in our case) to 
be unacceptable, and is definitely unacceptable to the calling card 
customer.

But I don't think we are talking about calling cards.  I am guessing that 
Dhaval is trying to create a termination company, and has customers that 
maintain a balance with him that want to be able to place multiple 
simultaneous calls.  A common problem.  We often end up with negative 
balances with our upstreams for this very reason - we may be near the 
bottom of our balance and several calls in progress terminate and bring us 
below zero.  I am sure this is what he is trying to avoid, as the industry 
is full of people that will simply walk away from a negative balance.

Dhaval - your wish, I think, is to manage exactly in real time to decrease the 
balance as the calls progress.  In that way all calls in progress would be 
cutoff simultaneously as the balance hit zero.  That kind of scenario would be 
very complicated with asterisk.  Some external program would have to keep track 
of the balance and the calls currently in progress, and cut them off at the 
appropriate time.  I would be very interested if anyone has attempted 
this.  I envision something that EVERY SECOND deducts from a balance for 
every call in progress, at the current rate for each call.  Not impossible 
for sure...

Cheers,

j

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Re: [asterisk-users] Asterisk Realtime Billing Question???

[Tarek Sawah]

actually my mail was not meant to be disrespectful. it was an inquiry. i have a 
billing system and had a few of those thoughts regarding real time billing. my 
issue was explaining to a customer that his call disconnected an hour earlier 
because someone else used his account.. I'm doing retail not wholesale, you may 
understand my question more clearly now?

Regards


Tarek Sawah

Information Technology Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993








> From: sherwood.mcgo...@gmail.com
> Date: Thu, 21 Oct 2010 05:18:17 -0500
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
>
> Tarek,
>
> I'm not sure why it would be our problem is someone came into your
> office and started making long distance calls over a trunk I was
> providing your company I'm pretty sure that if I had tried that
> with some of my carriers in the past they would have laughed until
> they cried...
>
> Oh, and also, since this was a wholesale carrier, the customers were
> in control of their own freeze amount. It was there to allow THEM to
> control their account better. I'd be willing to bet that my clients
> would have been happy to just keep billing them for every minute they
> used.
>
> Lastly, I would like to just say, I'm not the guy who requested the
> feature, I'm the guy who figured out how to make it happen, and making
> it happen back in early 2006, when the MySQL addon was just BARELY
> stable...
>
> It's ok, I don't need respect, I have the knowledge that I'm the mick,
> and I'm awesome :P
>
> Cheers :D
>
> On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah  wrote:
> > If you look at it the way you want it.. you usually tell your customer the
> > available funds and minutes in their account right?
> > How will you explain "politely" that you have dropped their calls for lack
> > of balance because someone else used their account?
> > If you don't tell them their balance and call duration before call .. then
> > that won't be a problem.
> > Now you can do some kind of script to do the math and disconnect calls when
> > balance is over.
> >
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
> > INDRODIYA
> > Sent: Thursday, October 21, 2010 9:31 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
> >
> > Hi Sherwood ,
> >
> > well , i think you did not understand my question , i want real time billing
> > like as i mentioned that if i want to dial 5 number with different call rate
> > how can i access same
> > balance into those 5 people, if all are connected how can i periodically
> > update billing , as you suggested it will assign total balance to those 5
> > people but actually we can not do like this as total balance of user $100 ,
> > as per your suggestion it will give $100 for those 5 people which is
> > practically wrong i think.
> >
> > give your thougts.
> >
> > regards
> > dhaval
> > On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
> >  wrote:
> > On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
> >  wrote:
> > Hello All,
> >
> > after so long time i posted a new question regarding billing, hope  anyone
> > have some solution.
> >
> > I have situation in that i want to do billing of more than 1 call in real
> > time below are scenario and explanation.
> >
> >
> > Scenario:
> >  A customer called my DID number and after that from here i dial few number
> > let say 5 number. once number are placed into DIAL
> > i will put this customer into conference [MEETME] , once a Members are
> > picked up call they will also patched into conference and
> > talking is started, every thing working fine with DIAL-PLAN and DB look up.
> >
> > Now, i want to do billing on customer dialed my DID, and from that actually
> > it DIALED 5 numbers, how can i DO real time billing
> > into this situation, like numbers can be different It can be ISD,STD,Local
> > and also free .
> >
> > if customer having initial balance of $100 then how can i check balance
> > every time.in a situation once balance is nil then i want to disconnect
> > calls . is any one facing this type of situation.
> >
> > give me some  idea ,
> >
> > regards
> > Dhaval
> >
> > --
> > __

Re: [asterisk-users] Asterisk Realtime Billing Question???

[Sherwood McGowan]

Tarek,

I'm not sure why it would be our problem is someone came into your
office and started making long distance calls over a trunk I was
providing your company I'm pretty sure that if I had tried that
with some of my carriers in the past they would have laughed until
they cried...

Oh, and also, since this was a wholesale carrier, the customers were
in control of their own freeze amount. It was there to allow THEM to
control their account better. I'd be willing to bet that my clients
would have been happy to just keep billing them for every minute they
used.

Lastly, I would like to just say, I'm not the guy who requested the
feature, I'm the guy who figured out how to make it happen, and making
it happen back in early 2006, when the MySQL addon was just BARELY
stable...

It's ok, I don't need respect, I have the knowledge that I'm the mick,
and I'm awesome :P

Cheers :D

On Thu, Oct 21, 2010 at 4:37 AM, Tarek Sawah  wrote:
> If you look at it the way you want it.. you usually tell your customer the
> available funds and minutes in their account right?
> How will you explain "politely" that you have dropped their calls for lack
> of balance because someone else used their account?
> If you don't tell them their balance and call duration before call .. then
> that won't be a problem.
> Now you can do some kind of script to do the math and disconnect calls when
> balance is over.
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
> INDRODIYA
> Sent: Thursday, October 21, 2010 9:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???
>
> Hi Sherwood ,
>
> well , i think you did not understand my question , i want real time billing
> like as i mentioned that if i want to dial 5 number with different call rate
> how can i access same
> balance into those 5 people, if all are connected how can i periodically
> update billing , as you suggested it will assign total balance to those 5
> people but actually we can not do like this as total balance of user $100 ,
> as per your suggestion it will give $100 for those 5 people which is
> practically wrong i think.
>
> give your thougts.
>
> regards
> dhaval
> On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
>  wrote:
> On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
>  wrote:
> Hello All,
>
> after so long time i posted a new question regarding billing, hope  anyone
> have some solution.
>
> I have situation in that i want to do billing of more than 1 call in real
> time below are scenario and explanation.
>
>
> Scenario:
>  A customer called my DID number and after that from here i dial few number
> let say 5 number. once number are placed into DIAL
> i will put this customer into conference [MEETME] , once a Members are
> picked up call they will also patched into conference and
> talking is started, every thing working fine with DIAL-PLAN and DB look up.
>
> Now, i want to do billing on customer dialed my DID, and from that actually
> it DIALED 5 numbers, how can i DO real time billing
> into this situation, like numbers can be different It can be ISD,STD,Local
> and also free .
>
> if customer having initial balance of $100 then how can i check balance
> every time.in a situation once balance is nil then i want to disconnect
> calls . is any one facing this type of situation.
>
> give me some  idea ,
>
> regards
> Dhaval
>
> --
> _
> -- 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
>
>
> Dhaval,
> This sounds very much like a system I'm working on for a client right now.
> I'm not permitted to disclose much about it due to the NDA i signed, but
> I'll risk giving you a point in the right direction.
>
> First, you should create a table in your database that has a column called
> callid, and other columns that you will have to decide upon. This table will
> be called something like 'call_references'. Oh, and you'll want to define
> callid as the primary key for records in that table, but DO NOT make it an
> autoincrement, you're going to populate it with a value that is described in
> the next step.
>
> Second, at the beginning of the original call you mentioned, define a
> variable that will be u

Re: [asterisk-users] Asterisk Realtime Billing Question???

[Tarek Sawah]

If you look at it the way you want it.. you usually tell your customer the
available funds and minutes in their account right?
How will you explain "politely" that you have dropped their calls for lack
of balance because someone else used their account?
If you don't tell them their balance and call duration before call .. then
that won't be a problem. 
Now you can do some kind of script to do the math and disconnect calls when
balance is over.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, October 21, 2010 9:31 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk Realtime Billing Question???

Hi Sherwood ,

well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same 
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think. 

give your thougts.

regards
dhaval
On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan
 wrote:
On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA
 wrote:
Hello All,

after so long time i posted a new question regarding billing, hope  anyone
have some solution.

I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation. 


Scenario:
 A customer called my DID number and after that from here i dial few number
let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and 
talking is started, every thing working fine with DIAL-PLAN and DB look up. 

Now, i want to do billing on customer dialed my DID, and from that actually
it DIALED 5 numbers, how can i DO real time billing 
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .

if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect 
calls . is any one facing this type of situation.

give me some  idea ,

regards
Dhaval

--
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Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction. 

First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like 'call_references'. Oh, and you'll want to define
callid as the primary key for records in that table, but DO NOT make it an
autoincrement, you're going to populate it with a value that is described in
the next step.

Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}. 

Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value). 

Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value. 

Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls. 

I wish I could tell you more, but I can't violate my client's Non-Disclosure
Agreement.

Hope this helps you out!

Sherwood McGowan


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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              http://www.asterisk.org/hello

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  http://lists.digium.com/mailman/listinfo/asterisk-users



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Re: [asterisk-users] Asterisk Realtime Billing Question???

[Sherwood McGowan]

On Thu, Oct 21, 2010 at 3:23 AM, DHAVAL INDRODIYA
wrote:

> thanks mate,
>
> for useful and good information provided by you, i am not asking you that
> please write down your all LOGIC and explain everything to me, as per your
> explanation i can see it will deduct amount for only 1 call but what
> actually i am searching for is if user made 5 concurrent calls and i have to
> limit
> all calls and each destination number having different rate may be some of
> them ISD and some of them local. that will create more problem to me, i
> think there is some solutions for this . could you suggest any reference for
> the same, it will be more helpful to me.
>
> thanks in advance,
> regards
> Dhaval
>
>  *snip*
>>>
>>>

>  Dhaval,
 This sounds very much like a system I'm working on for a client right
 now. I'm not permitted to disclose much about it due to the NDA i signed,
 but I'll risk giving you a point in the right direction.

 First, you should create a table in your database that has a column
 called callid, and other columns that you will have to decide upon. This
 table will be called something like '*call_references*'. Oh, and you'll
 want to define callid as the primary key for records in that table, but DO
 NOT make it an autoincrement, you're going to populate it with a value that
 is described in the next step.

 Second, at the beginning of the original call you mentioned, define a
 variable that will be unique to that call. I personally have done this by
 stripping all non-digits from the caller's callerid (using
 Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
 to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.

 Next (this is where I have to start being a bit vague), you're going to
 perform an INSERT query, creating a new call_references record (using that
 variable I just showed you how to construct as callid's value).

 Now, when you defined that variable, you should have preceded the
 variable name with two underscores ( __ ), which will tell Asterisk that
 channels spawned by the current channel will inherit that variable and it's
 value.

 Voila, you now have a method for storing realtime data such as billing
 information between MULTIPLE calls.

 I wish I could tell you more, but I can't violate my client's
 Non-Disclosure Agreement.

 Hope this helps you out!

 Sherwood McGowan


>> Well, you got the right guy, I've written several different RT billing
>> setups for clients ranging from small residential ITSPs all the way up to a
>> wholesale carrier in Austria. . .
>>
>> What you'd have to do is have a column called "freeze" in your table that
>> you keep customer accounts and billing info in (mainly, the balance). Then,
>> you'd need a 'frozen' table, with three columns: id, accountid (or some
>> other name/reference that references the customer in question), and amount.
>>
>> Now, the "freeze" column in the account table defines how many minutes
>> worth of funds (at the rate the call is being charged to the customer)
>> you're going to make unavailable to the customer until the call is
>> completed. You multiply the value from "freeze" against the rate the call is
>> going to be charged at, resulting in "amount_to_freeze". Subtract that
>> number from the customer's current balance, and then create a record in the
>> "frozen" table with that customer's accountid and put the value of
>> amount_to_freeze into the "amount" column.
>>
>> Finally,when the customer's call(s) completes, calculate the total charge
>> for the call, check to see if it's more than `frozen`.`amount`, and if it
>> is, subtract `frozen`.`amount` from the total charge, and then subtract the
>> remaining amount from the customer's balance. If the total is *not* more
>> than than `frozen`.`amount`, you'll subtract total from `frozen`.`amount`,
>> and then ADDING the remaining amount to the customer's balance. (Being the
>> doofus I am, I called that procedure "thawing", LOL)
>>
>> In addition to the freezing of funds, you'll need to perform some magic
>> and limit the length of the customer's calls based on the balance of the
>> account just before freezing funds. This will need to be in conjunction with
>> having a maximum number of concurrent calls the customer can have, and
>> taking that into account when limiting each call.
>>
>> It sounds complicated but I wrote this type of system several times, the
>> first couple were native to Asterisk using AELv2 (no AGI calls, more secure,
>> less resources hogged, etc), and then I wrote the last one using MySQL
>> stored procedures to perform just about ALL of the calculations and logic.
>> Basically at the beginning of a call, Asterisk would execute a stored
>> procedure called something like freeze_and_limit and passing two arguments,
>> the accountid and the rate per minute their call is

Re: [asterisk-users] Asterisk Realtime Billing Question???

[DHAVAL INDRODIYA]

thanks mate,

for useful and good information provided by you, i am not asking you that
please write down your all LOGIC and explain everything to me, as per your
explanation i can see it will deduct amount for only 1 call but what
actually i am searching for is if user made 5 concurrent calls and i have to
limit
all calls and each destination number having different rate may be some of
them ISD and some of them local. that will create more problem to me, i
think there is some solutions for this . could you suggest any reference for
the same, it will be more helpful to me.

thanks in advance,
regards
Dhaval

On Thu, Oct 21, 2010 at 12:49 PM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:

>
>
> On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA <
> dhaval.it01...@gmail.com> wrote:
>
>> Hi Sherwood ,
>>
>> well , i think you did not understand my question , i want real time
>> billing
>> like as i mentioned that if i want to dial 5 number with different call
>> rate how can i access same
>> balance into those 5 people, if all are connected how can i periodically
>> update billing , as you suggested it will assign total balance to those 5
>> people but actually we can not do like this as total balance of user $100 ,
>> as per your suggestion it will give $100 for those 5 people which is
>> practically wrong i think.
>>
>> give your thougts.
>>
>> regards
>> dhaval
>>
>>
>> On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan <
>> sherwood.mcgo...@gmail.com> wrote:
>>
>>> On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA <
>>> dhaval.it01...@gmail.com> wrote:
>>>
 Hello All,

 after so long time i posted a new question regarding billing, hope
 anyone have some solution.

 I have situation in that i want to do billing of more than 1 call in
 real time below are scenario and explanation.


 Scenario:
  A customer called my DID number and after that from here i dial few
 number let say 5 number. once number are placed into DIAL
 i will put this customer into conference [MEETME] , once a Members are
 picked up call they will also patched into conference and
 talking is started, every thing working fine with DIAL-PLAN and DB look
 up.

 Now, i want to do billing on customer dialed my DID, and from that
 actually it DIALED 5 numbers, how can i DO real time billing
 into this situation, like numbers can be different It can be
 ISD,STD,Local and also free .

 if customer having initial balance of $100 then how can i check balance
 every time.in a situation once balance is nil then i want to disconnect

 calls . is any one facing this type of situation.

 give me some  idea ,

 regards
 Dhaval


 --
 _
 -- 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

>>>
>>>
>>> Dhaval,
>>> This sounds very much like a system I'm working on for a client right
>>> now. I'm not permitted to disclose much about it due to the NDA i signed,
>>> but I'll risk giving you a point in the right direction.
>>>
>>> First, you should create a table in your database that has a column
>>> called callid, and other columns that you will have to decide upon. This
>>> table will be called something like '*call_references*'. Oh, and you'll
>>> want to define callid as the primary key for records in that table, but DO
>>> NOT make it an autoincrement, you're going to populate it with a value that
>>> is described in the next step.
>>>
>>> Second, at the beginning of the original call you mentioned, define a
>>> variable that will be unique to that call. I personally have done this by
>>> stripping all non-digits from the caller's callerid (using
>>> Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
>>> to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
>>>
>>> Next (this is where I have to start being a bit vague), you're going to
>>> perform an INSERT query, creating a new call_references record (using that
>>> variable I just showed you how to construct as callid's value).
>>>
>>> Now, when you defined that variable, you should have preceded the
>>> variable name with two underscores ( __ ), which will tell Asterisk that
>>> channels spawned by the current channel will inherit that variable and it's
>>> value.
>>>
>>> Voila, you now have a method for storing realtime data such as billing
>>> information between MULTIPLE calls.
>>>
>>> I wish I could tell you more, but I can't violate my client's
>>> Non-Disclosure Agreement.
>>>
>>> Hope this helps you out!
>>>
>>> Sherwood McGowan
>>>
>>>
>>> --
>>> _

Re: [asterisk-users] Asterisk Realtime Billing Question???

[Sherwood McGowan]

On Thu, Oct 21, 2010 at 1:30 AM, DHAVAL INDRODIYA
wrote:

> Hi Sherwood ,
>
> well , i think you did not understand my question , i want real time
> billing
> like as i mentioned that if i want to dial 5 number with different call
> rate how can i access same
> balance into those 5 people, if all are connected how can i periodically
> update billing , as you suggested it will assign total balance to those 5
> people but actually we can not do like this as total balance of user $100 ,
> as per your suggestion it will give $100 for those 5 people which is
> practically wrong i think.
>
> give your thougts.
>
> regards
> dhaval
>
>
> On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan <
> sherwood.mcgo...@gmail.com> wrote:
>
>> On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA <
>> dhaval.it01...@gmail.com> wrote:
>>
>>> Hello All,
>>>
>>> after so long time i posted a new question regarding billing, hope
>>> anyone have some solution.
>>>
>>> I have situation in that i want to do billing of more than 1 call in real
>>> time below are scenario and explanation.
>>>
>>>
>>> Scenario:
>>>  A customer called my DID number and after that from here i dial few
>>> number let say 5 number. once number are placed into DIAL
>>> i will put this customer into conference [MEETME] , once a Members are
>>> picked up call they will also patched into conference and
>>> talking is started, every thing working fine with DIAL-PLAN and DB look
>>> up.
>>>
>>> Now, i want to do billing on customer dialed my DID, and from that
>>> actually it DIALED 5 numbers, how can i DO real time billing
>>> into this situation, like numbers can be different It can be
>>> ISD,STD,Local and also free .
>>>
>>> if customer having initial balance of $100 then how can i check balance
>>> every time.in a situation once balance is nil then i want to disconnect
>>> calls . is any one facing this type of situation.
>>>
>>> give me some  idea ,
>>>
>>> regards
>>> Dhaval
>>>
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>   http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> Dhaval,
>> This sounds very much like a system I'm working on for a client right now.
>> I'm not permitted to disclose much about it due to the NDA i signed, but
>> I'll risk giving you a point in the right direction.
>>
>> First, you should create a table in your database that has a column called
>> callid, and other columns that you will have to decide upon. This table will
>> be called something like '*call_references*'. Oh, and you'll want to
>> define callid as the primary key for records in that table, but DO NOT make
>> it an autoincrement, you're going to populate it with a value that is
>> described in the next step.
>>
>> Second, at the beginning of the original call you mentioned, define a
>> variable that will be unique to that call. I personally have done this by
>> stripping all non-digits from the caller's callerid (using
>> Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
>> to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
>>
>> Next (this is where I have to start being a bit vague), you're going to
>> perform an INSERT query, creating a new call_references record (using that
>> variable I just showed you how to construct as callid's value).
>>
>> Now, when you defined that variable, you should have preceded the variable
>> name with two underscores ( __ ), which will tell Asterisk that channels
>> spawned by the current channel will inherit that variable and it's value.
>>
>> Voila, you now have a method for storing realtime data such as billing
>> information between MULTIPLE calls.
>>
>> I wish I could tell you more, but I can't violate my client's
>> Non-Disclosure Agreement.
>>
>> Hope this helps you out!
>>
>> Sherwood McGowan
>>
>>
>> --
>> _
>> -- 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
>

Well, you got the right guy, I've written several different

Re: [asterisk-users] Asterisk Realtime Billing Question???

[DHAVAL INDRODIYA]

Hi Sherwood ,

well , i think you did not understand my question , i want real time billing
like as i mentioned that if i want to dial 5 number with different call rate
how can i access same
balance into those 5 people, if all are connected how can i periodically
update billing , as you suggested it will assign total balance to those 5
people but actually we can not do like this as total balance of user $100 ,
as per your suggestion it will give $100 for those 5 people which is
practically wrong i think.

give your thougts.

regards
dhaval

On Thu, Oct 21, 2010 at 11:44 AM, Sherwood McGowan <
sherwood.mcgo...@gmail.com> wrote:

> On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA <
> dhaval.it01...@gmail.com> wrote:
>
>> Hello All,
>>
>> after so long time i posted a new question regarding billing, hope  anyone
>> have some solution.
>>
>> I have situation in that i want to do billing of more than 1 call in real
>> time below are scenario and explanation.
>>
>>
>> Scenario:
>>  A customer called my DID number and after that from here i dial few
>> number let say 5 number. once number are placed into DIAL
>> i will put this customer into conference [MEETME] , once a Members are
>> picked up call they will also patched into conference and
>> talking is started, every thing working fine with DIAL-PLAN and DB look
>> up.
>>
>> Now, i want to do billing on customer dialed my DID, and from that
>> actually it DIALED 5 numbers, how can i DO real time billing
>> into this situation, like numbers can be different It can be ISD,STD,Local
>> and also free .
>>
>> if customer having initial balance of $100 then how can i check balance
>> every time.in a situation once balance is nil then i want to disconnect
>> calls . is any one facing this type of situation.
>>
>> give me some  idea ,
>>
>> regards
>> Dhaval
>>
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> Dhaval,
> This sounds very much like a system I'm working on for a client right now.
> I'm not permitted to disclose much about it due to the NDA i signed, but
> I'll risk giving you a point in the right direction.
>
> First, you should create a table in your database that has a column called
> callid, and other columns that you will have to decide upon. This table will
> be called something like '*call_references*'. Oh, and you'll want to
> define callid as the primary key for records in that table, but DO NOT make
> it an autoincrement, you're going to populate it with a value that is
> described in the next step.
>
> Second, at the beginning of the original call you mentioned, define a
> variable that will be unique to that call. I personally have done this by
> stripping all non-digits from the caller's callerid (using
> Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
> to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.
>
> Next (this is where I have to start being a bit vague), you're going to
> perform an INSERT query, creating a new call_references record (using that
> variable I just showed you how to construct as callid's value).
>
> Now, when you defined that variable, you should have preceded the variable
> name with two underscores ( __ ), which will tell Asterisk that channels
> spawned by the current channel will inherit that variable and it's value.
>
> Voila, you now have a method for storing realtime data such as billing
> information between MULTIPLE calls.
>
> I wish I could tell you more, but I can't violate my client's
> Non-Disclosure Agreement.
>
> Hope this helps you out!
>
> Sherwood McGowan
>
>
> --
> _
> -- 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] Asterisk Realtime Billing Question???

[Sherwood McGowan]

On Thu, Oct 21, 2010 at 12:24 AM, DHAVAL INDRODIYA  wrote:

> Hello All,
>
> after so long time i posted a new question regarding billing, hope  anyone
> have some solution.
>
> I have situation in that i want to do billing of more than 1 call in real
> time below are scenario and explanation.
>
>
> Scenario:
>  A customer called my DID number and after that from here i dial few number
> let say 5 number. once number are placed into DIAL
> i will put this customer into conference [MEETME] , once a Members are
> picked up call they will also patched into conference and
> talking is started, every thing working fine with DIAL-PLAN and DB look up.
>
>
> Now, i want to do billing on customer dialed my DID, and from that actually
> it DIALED 5 numbers, how can i DO real time billing
> into this situation, like numbers can be different It can be ISD,STD,Local
> and also free .
>
> if customer having initial balance of $100 then how can i check balance
> every time.in a situation once balance is nil then i want to disconnect
> calls . is any one facing this type of situation.
>
> give me some  idea ,
>
> regards
> Dhaval
>
>
> --
> _
> -- 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
>


Dhaval,
This sounds very much like a system I'm working on for a client right now.
I'm not permitted to disclose much about it due to the NDA i signed, but
I'll risk giving you a point in the right direction.

First, you should create a table in your database that has a column called
callid, and other columns that you will have to decide upon. This table will
be called something like '*call_references*'. Oh, and you'll want to define
callid as the primary key for records in that table, but DO NOT make it an
autoincrement, you're going to populate it with a value that is described in
the next step.

Second, at the beginning of the original call you mentioned, define a
variable that will be unique to that call. I personally have done this by
stripping all non-digits from the caller's callerid (using
Set(newcid=${FILTER(0123456789,${CALLERID(number)})} ), and then adding the
to ${EPOCH}. I did it this way: ${MATH(${newcid}+${EPOCH})}.

Next (this is where I have to start being a bit vague), you're going to
perform an INSERT query, creating a new call_references record (using that
variable I just showed you how to construct as callid's value).

Now, when you defined that variable, you should have preceded the variable
name with two underscores ( __ ), which will tell Asterisk that channels
spawned by the current channel will inherit that variable and it's value.

Voila, you now have a method for storing realtime data such as billing
information between MULTIPLE calls.

I wish I could tell you more, but I can't violate my client's Non-Disclosure
Agreement.

Hope this helps you out!

Sherwood McGowan
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Re: [asterisk-users] Asterisk Realtime Billing Question???

[Zeeshan Zakaria]

Billing is a complex matter, however the CDR contains all the necessary
information. You need to carefully see how calls are recorded in the CDR
when meetme finishes, and how all the legs were connected. Usually
accountcode, dstchannel and channel columns have the required info which
tells how the calls were connected. Once you know how the calls were
connected, get each calls duration and destination and multiply them with
the destination calling cost. You can program this logic in PHP and use as a
AGI script in the hangup context of your dialplan. Its not quick and easy
task to do, but there is no quick and easy way for this purpose. Similarly
to be able to not proceed with the call if account balance is zero, you need
to check account balance before calling the meetme application.

Zeeshan A Zakaria

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On 2010-10-21 1:32 AM, "DHAVAL INDRODIYA"  wrote:

Hello All,

after so long time i posted a new question regarding billing, hope  anyone
have some solution.

I have situation in that i want to do billing of more than 1 call in real
time below are scenario and explanation.


Scenario:
 A customer called my DID number and after that from here i dial few number
let say 5 number. once number are placed into DIAL
i will put this customer into conference [MEETME] , once a Members are
picked up call they will also patched into conference and
talking is started, every thing working fine with DIAL-PLAN and DB look up.

Now, i want to do billing on customer dialed my DID, and from that actually
it DIALED 5 numbers, how can i DO real time billing
into this situation, like numbers can be different It can be ISD,STD,Local
and also free .

if customer having initial balance of $100 then how can i check balance
every time.in a situation once balance is nil then i want to disconnect
calls . is any one facing this type of situation.

give me some  idea ,

regards
Dhaval


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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Phibee Network Operation Center]

  Hi

thanks for your answer, i put that don't work, but it's a error, that work.

But Asterisk crash when i use my second extensions table, i don't know why
(limitation of number of line ?)

I don't have the answer actually ;=)

bye
Jerome


Le 01/10/2010 11:07, Захаров Антон a écrit :
>[ivr_holiday]
> switch =>  Realtime/ivr_holid...@extensions
>
> where 'ivr_holidays'  is context and 'extensions' is table
>
> On 01.10.2010 12:52, Phibee Network Operation Center wrote:
>> Hi
>>
>> i am not a expert on Asterisk and search a lot of small information :
>>
>>I use Asterisk 1.6.1.4 with MySQL.
>>
>> That's work and in my extension.conf, i have:
>>[as5300-incoming]
>>switch =>   Realtime
>>
>> and in extconfig.conf
>>extensions =>   mysql,general,VOIP_Extensions
>> A lot of Extension are into the table VOIP_Extensions.
>>
>> I am search to know if i can add a :
>>[beta-incoming]
>>switch =>   Realtime
>>
>>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
>>
>>
>> Anyone know if it's possible ? (use two table for extension)
>>
>> Thanks
>> Jerome SCHEVINGT
>>
>


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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Phibee Network Operation Center]

After test, that's don't work :=<




Le 01/10/2010 11:07, Захаров Антон a écrit :
>[ivr_holiday]
> switch =>  Realtime/ivr_holid...@extensions
>
> where 'ivr_holidays'  is context and 'extensions' is table
>
> On 01.10.2010 12:52, Phibee Network Operation Center wrote:
>> Hi
>>
>> i am not a expert on Asterisk and search a lot of small information :
>>
>>I use Asterisk 1.6.1.4 with MySQL.
>>
>> That's work and in my extension.conf, i have:
>>[as5300-incoming]
>>switch =>   Realtime
>>
>> and in extconfig.conf
>>extensions =>   mysql,general,VOIP_Extensions
>> A lot of Extension are into the table VOIP_Extensions.
>>
>> I am search to know if i can add a :
>>[beta-incoming]
>>switch =>   Realtime
>>
>>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
>>
>>
>> Anyone know if it's possible ? (use two table for extension)
>>
>> Thanks
>> Jerome SCHEVINGT
>>
>


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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Phibee Network Operation Center]

  Le 01/10/2010 11:10, Steve Howes a écrit :
> On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
>> That's work and in my extension.conf, i have:
>>  [as5300-incoming]
>>  switch =>  Realtime
>>
>> and in extconfig.conf
>>  extensions =>  mysql,general,VOIP_Extensions
>> A lot of Extension are into the table VOIP_Extensions.
>>
>> I am search to know if i can add a :
>>  [beta-incoming]
>>  switch =>  Realtime
>>
>>  but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
>>
>>
>> Anyone know if it's possible ? (use two table for extension)
> Dont think you can use two tables.. But you're using two contexts there 
> right? Just have your 'beta' stuff in the same table, but different context.
>
> S

Yes but i want two table, not two context in one table.

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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Phibee Network Operation Center]

Thanks, it's limited the number of table ?


Le 01/10/2010 11:07, Захаров Антон a écrit :
>[ivr_holiday]
> switch =>  Realtime/ivr_holid...@extensions
>
> where 'ivr_holidays'  is context and 'extensions' is table
>
> On 01.10.2010 12:52, Phibee Network Operation Center wrote:
>> Hi
>>
>> i am not a expert on Asterisk and search a lot of small information :
>>
>>I use Asterisk 1.6.1.4 with MySQL.
>>
>> That's work and in my extension.conf, i have:
>>[as5300-incoming]
>>switch =>   Realtime
>>
>> and in extconfig.conf
>>extensions =>   mysql,general,VOIP_Extensions
>> A lot of Extension are into the table VOIP_Extensions.
>>
>> I am search to know if i can add a :
>>[beta-incoming]
>>switch =>   Realtime
>>
>>but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
>>
>>
>> Anyone know if it's possible ? (use two table for extension)
>>
>> Thanks
>> Jerome SCHEVINGT
>>
>


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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Steve Howes]

On 1 Oct 2010, at 09:52, Phibee Network Operation Center wrote:
> That's work and in my extension.conf, i have:
> [as5300-incoming]
> switch => Realtime
> 
> and in extconfig.conf
> extensions => mysql,general,VOIP_Extensions
> A lot of Extension are into the table VOIP_Extensions.
> 
> I am search to know if i can add a :
> [beta-incoming]
> switch => Realtime
> 
> but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
> 
> 
> Anyone know if it's possible ? (use two table for extension)

Dont think you can use two tables.. But you're using two contexts there right? 
Just have your 'beta' stuff in the same table, but different context.

S
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Re: [asterisk-users] Asterisk/Realtime and MySQL

[Захаров Антон]

  [ivr_holiday]
switch => Realtime/ivr_holid...@extensions

where 'ivr_holidays'  is context and 'extensions' is table

On 01.10.2010 12:52, Phibee Network Operation Center wrote:
>Hi
>
> i am not a expert on Asterisk and search a lot of small information :
>
>   I use Asterisk 1.6.1.4 with MySQL.
>
> That's work and in my extension.conf, i have:
>   [as5300-incoming]
>   switch =>  Realtime
>
> and in extconfig.conf
>   extensions =>  mysql,general,VOIP_Extensions
> A lot of Extension are into the table VOIP_Extensions.
>
> I am search to know if i can add a :
>   [beta-incoming]
>   switch =>  Realtime
>
>   but not use the table "VOIP_Extensions" but "VOIP_Extensions_Beta"
>
>
> Anyone know if it's possible ? (use two table for extension)
>
> Thanks
> Jerome SCHEVINGT
>


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Re: [asterisk-users] asterisk realtime SIP configuration

[Jonathan Thurman]

On Wed, Jul 21, 2010 at 3:09 AM, Murali Vasu  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

If you want to see the peers on the CLI, then you have to enable
caching of the peers.  Add this to your sip.conf file:

[general]
rtcachefriends=yes


-Jonathan


>  i can able to connect to the data source through "odbc show" in the
> CLI, Any hep in this regard is highly appreciated. Following is the
> configuration and specification.
>
>  Server Specification:
>
>     1) asterisk-1.6.2.6
>     2) CentOS- 5.2 (64-bit)
>     3) Postgresql- 8.1
>
>  Configuration:
>
>  odbc.ini
>
>  [PostgreSQL]
> Description = Test to Postgres
> Driver  = PostgreSQL
> Trace   = Yes
> TraceFile   = /tmp/sql.log
> Database    = bedrock
> Servername  = localhost
> UserName    =
> Password    =
> Port    = 5432
> Protocol    = 6.4
> ReadOnly    = No
> RowVersioning   = No
> ShowSystemTables    = No
> ShowOidColumn   = No
> FakeOidIndex    = No
> ConnSettings    =
>
>  odbcinst.ini
>
> [PostgreSQL]
> Description = ODBC for PostgreSQL
> Driver  = /usr/lib64/libodbcpsql.so
> Setup   = /usr/lib64/libodbcpsqlS.so
> FileUsage   = 1
>
>     res_odbc.conf
>
> [postgres]
> enabled => yes
> dsn => PostgreSQL
> username =>postgres
> password =>postgres
> pre-connect => yes
>
>
>     Database table in postgres "sip" :
>
>  Column |  Type  |    Modifiers
> ++--
>  id | integer    | not null default
> nextval('sip_id_seq'::regclass)
>  name   | character varying(80)  | not null
>  accountcode    | character varying(20)  |
>  amaflags   | character varying(7)   |
>  callgroup  | character varying(10)  |
>  callerid   | character varying(80)  |
>  directmedia    | character varying(3)   | default 'yes'::character varying
>  context    | character varying(80)  | default 'default'::character
> varying
>  defaultip  | character varying(15)  |
>  dtmfmode   | character varying(7)   |
>  fromuser   | character varying(80)  |
>  fromdomain | character varying(80)  |
>  host   | character varying(31)  | not null default
> 'dynamic'::character varying
>  insecure   | character varying(4)   |
>  language   | character varying(2)   |
>  mailbox    | character varying(50)  |
>  md5secret  | character varying(80)  |
>  nat    | character varying(5)   | not null default 'no'::character
> varying
>  permit | character varying(95)  |
>  deny   | character varying(95)  |
>  mask   | character varying(95)  |
>  pickupgroup    | character varying(10)  |
>  port   | character varying(5)   |
>  qualify    | character varying(3)   |
>  restrictcid    | character varying(1)   |
>  rtptimeout | character varying(3)   |
>  rtpholdtimeout | character varying(3)   |
>  secret | character varying(80)  |
>  type   | character varying  | not null default
> 'friend'::character varying
>  username   | character varying(80)  |
>  disallow   | character varying(100) | default 'all'::character varying
>  allow  | character varying(100) | default 'alaw,ulaw'::character
> varying
>  musiconhold    | character varying(100) |
>  regseconds | integer    | not null default 0
>  ipaddr | character varying(15)  |
>  regexten   | character varying(80)  |
>  cancallforward | character varying(3)   | default 'yes'::character varying
>  lastms | character varying(80)  |
>  useragent  | character varying(100) |
>  defaultuser    | character varying(100) |
>  fullcontact    | character varying(100) |
>  regserver  | character varying(100) |
> Indexes:
>     "sip_conf_pkey" PRIMARY KEY, btree (id)
>     "name" UNIQUE, btree (name)
>
>     extconfig.conf
>
> sipusers => odbc,postgres,sip
> sippeers => odbc,postgres,sip
>
>
> Thanks & Regards
>
> Murali Vasu
>
>
>
>
>
>
>
>
> --
> Smile is the only priceless gift you can give without a price.
>
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Re: [asterisk-users] Asterisk Realtime Extensions => for all context ?

[Samuel Nair]

Are the user1, user2... SIP clients..?

sam!!

Phibee Network Operation Center wrote:
> Hi
>
> I Use Asterisk 1.6.1 with Realtime and a mySQL database,
>
> Actually, my extensions.conf are:
>
> ===
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [globals]
> CONSOLE=Console/dsp ; Console interface for demo
>
> [as5300-incoming]
> switch => Realtime
>
> [as5300-outgoing]
> switch => Realtime
>
> [user1]
> switch => Realtime
>
> [user2]
> switch => Realtime
>
> [user3]
> switch => Realtime
> ===
>
>
> and into my table, i put the context ...
>
> anyone know if i can use a generic:
>
> [general]
> static=yes
> writeprotect=no
> autofallthrough=yes
> clearglobalvars=no
> priorityjumping=no
>
> [globals]
> CONSOLE=Console/dsp ; Console interface for demo
> switch => Realtime
>
> and use directly the context of my database ? without put userX into my 
> extensions.conf
>
> thanks
> Jerome
>
>
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Re: [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?

[Phibee Network Operation Center]

Hi

ok i have understand ;=)

bye


Phibee Network Operation Center a écrit :
> Hi
>
> actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
>
> I read on the wiki:
>
> ===
> Database Config
> put the following in res_mysql.conf
>
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = myuser
> dbpass = mypass
> dbport = 3306
>
> Values in sip.conf or iax.conf like in older versions of * are no longer 
> used.
>
>
> Database Table
> Lets create the table we need:
>
> NOTE: You can use any table name you wish, just make sure the table name 
> matches what you have the family name bound to.
>
> ===
>
>
> But i don't see where i put the Table Name ? (if i don't want use 
> sip_buddies)
>
> and he have a sample of Table Structure, can i add a new champs for my 
> personnal
> software without problems ?
>
> Thanks
> jerome
>
>
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Re: [asterisk-users] Asterisk, Realtime and specify MySQL Table Name ?

[Samuel Nair]

You can set it up in extconfig.conf:

iaxusers => mysql,asterisk,iaxusers
iaxpeers => mysql,asterisk,iaxusers
sipusers => mysql,asterisk,sipusers
sippeers => mysql,asterisk,sipusers
voicemail => mysql,asterisk,voicemail
extensions => mysql,asterisk,extensions
queues => mysql,asterisk,queues
queue_members => mysql,asterisk,queue_members
meetme => mysql,asterisk,meetme

sam!!

Phibee Network Operation Center wrote:
> Hi
>
> actually, i test a new Asterisk Server and i want add Mysql Realtime SIP.
>
> I read on the wiki:
>
> ===
> Database Config
> put the following in res_mysql.conf
>
> [general]
> dbhost = 127.0.0.1
> dbname = asterisk
> dbuser = myuser
> dbpass = mypass
> dbport = 3306
>
> Values in sip.conf or iax.conf like in older versions of * are no longer 
> used.
>
>
> Database Table
> Lets create the table we need:
>
> NOTE: You can use any table name you wish, just make sure the table name 
> matches what you have the family name bound to.
>
> ===
>
>
> But i don't see where i put the Table Name ? (if i don't want use 
> sip_buddies)
>
> and he have a sample of Table Structure, can i add a new champs for my 
> personnal
> software without problems ?
>
> Thanks
> jerome
>
>
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Re: [asterisk-users] Asterisk realtime

[michel freiha]

Dear Sir,

The issue has been solved

rtcachefriends=no
\and everything will work

Thanks

On Mon, Mar 2, 2009 at 10:31 PM, michel freiha  wrote:

> Hi all,
>
> I'm using asterisk in real time mode...All extensions are defined in table
> sip_buddies...Everything looks fine and asterisk is reading extensions info
> from the sip_buddies table...The problem occurs as soon as any information
> on an extension is changed from sip_buddies table...Which mean, if I change
> the secret field in sip_buddies table then i should reload asterisk to read
> again the new secret otherwise the old secret will stay used by asterisk
>
> Any suggestions will be appreciated
>
> Regards
>
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Re: [asterisk-users] Asterisk realtime

[Carlos Chavez]

On Mon, 2009-03-02 at 22:31 +0200, michel freiha wrote:
> Hi all,
> 
> I'm using asterisk in real time mode...All extensions are defined in
> table sip_buddies...Everything looks fine and asterisk is reading
> extensions info from the sip_buddies table...The problem occurs as
> soon as any information on an extension is changed from sip_buddies
> table...Which mean, if I change the secret field in sip_buddies table
> then i should reload asterisk to read again the new secret otherwise
> the old secret will stay used by asterisk
> 
> Any suggestions will be appreciated
> 

This usually happens when you are using realtime caching.  In that case
you simple need to do a "sip prune realtime peer [peername]" or "sip
prune realtime all" so the database configuration is read again.


> 
-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Asterisk Realtime Configuration

[Pedram M]

Matt,

Thanks I don't know what I was thinking...

Modifying the extensions.conf to extensions did the trick.

Regards,
Pedram

On Wed, Nov 5, 2008 at 5:47 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:

> On 6/11/2008 2:39 p.m., Pedram M wrote:
> > Matt,
> >
> > Yep, I forgot to post that here is the extensions.conf:
> >
> >
> > [10]
> > switch =>  Realtime
>
> According to
>
> http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions
>
> this should be:
>
> switch => Realtime/[EMAIL PROTECTED]/options]
>
> ; If context is not given, current context is default
> ; If family is not given, family of 'extensions' is default
>
> Heh, just read back through your first post.
>
> Think your problem is the family in extconfig.conf
>
> Should be extensions not extensions.conf
>
> I.E. extensions => odbc,asterisk
>
> --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
>
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Re: [asterisk-users] Asterisk Realtime Configuration

[Matt Riddell]

On 6/11/2008 2:39 p.m., Pedram M wrote:
> Matt,
>
> Yep, I forgot to post that here is the extensions.conf:
>
>
> [10]
> switch =>  Realtime

According to

http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions

this should be:

switch => Realtime/[EMAIL PROTECTED]/options]

; If context is not given, current context is default
; If family is not given, family of 'extensions' is default

Heh, just read back through your first post.

Think your problem is the family in extconfig.conf

Should be extensions not extensions.conf

I.E. extensions => odbc,asterisk

-- 
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] Asterisk Realtime Configuration

[Pedram M]

Matt,

Yep, I forgot to post that here is the extensions.conf:


[10]
switch => Realtime


Regards,
Pedram

On Wed, Nov 5, 2008 at 5:25 PM, Matt Riddell <[EMAIL PROTECTED]> wrote:

> On 6/11/2008 2:06 p.m., Pedram M wrote:
> > Hi,
> >
> > Having some issues here with getting asterisk realtime for the dialplan
> > (extensions.conf) setup:
> >
> > mysql>  desc extensions_table;
> > +--+--+--+-+-++
> > | Field| Type | Null | Key | Default | Extra  |
> > +--+--+--+-+-++
> > | id   | int(11)  | NO   | MUL | NULL| auto_increment |
> > | context  | varchar(255) | NO   | PRI | ||
> > | exten| varchar(255) | NO   | PRI | ||
> > | priority | varchar(255) | NO   | PRI | 0   ||
> > | app  | varchar(255) | NO   | | ||
> > | appdata  | text | NO   | | ||
> > +--+--+--+-+-++
> >
> >
> > #
> > ### extconfig.conf file ###
> > #
> >
> > extensions.conf =>  mysql,attributed,extensions_table
> >
> >
> >
> > Asterisk debug shows:
> >
> >
> >  -- Attempting call on SIP/grnvoip/123804011818345 for [EMAIL 
> > PROTECTED]
> :1
> > (Retry 1)
> >
> >== Starting SIP/grnvoip-09592260 at 10,start,1 failed so falling back
> to
> > exten 's'
> >
> >== Starting SIP/grnvoip-09592260 at 10,s,1 still failed so falling
> back to
> > context 'default'
> >
> > [Nov  5 19:04:42] WARNING[29109]: pbx.c:2470 __ast_pbx_run: Channel
> > 'SIP/grnvoip-09592260' sent into invalid extension 's' in context
> 'default',
> > but no invalid handler
> >
> >
> > This is with a call file that looks like:
> >
> >
> > Channel: SIP/grnvoip/123804011818345XXX
> > MaxRetries: 0
> > RetryTime: 60
> > WaitTime: 30
> > Context: 10
> > Extension: start
> > Priority: 1
> >
> >
> > And in the database the context 10, extension start and priority 1 does
> > exist as shown below:
> >
> > mysql>  select context,exten,priority,app from extensions_table limit
> 0,3;
> > +-+---+--++
> > | context | exten | priority | app|
> > +-+---+--++
> > | 10  | start | 1| Set|
> > | 10  | start | 2| AMD|
> > | 10  | start | 3| WaitforSilence |
> > +-+---+--++
> >
> >
> >
> > Any ideas on where to begin w/ the debug would be very appreciated.
>
> Are you doing the switch from the dialplan in the [10] context?
>
> Have a look at the realtime page on the wiki (voip-info.org)
>
> --
> Kind Regards,
>
> Matt Riddell
> Director
> ___
>
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
>
> ___
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Re: [asterisk-users] Asterisk Realtime Configuration

[Matt Riddell]

On 6/11/2008 2:06 p.m., Pedram M wrote:
> Hi,
>
> Having some issues here with getting asterisk realtime for the dialplan
> (extensions.conf) setup:
>
> mysql>  desc extensions_table;
> +--+--+--+-+-++
> | Field| Type | Null | Key | Default | Extra  |
> +--+--+--+-+-++
> | id   | int(11)  | NO   | MUL | NULL| auto_increment |
> | context  | varchar(255) | NO   | PRI | ||
> | exten| varchar(255) | NO   | PRI | ||
> | priority | varchar(255) | NO   | PRI | 0   ||
> | app  | varchar(255) | NO   | | ||
> | appdata  | text | NO   | | ||
> +--+--+--+-+-++
>
>
> #
> ### extconfig.conf file ###
> #
>
> extensions.conf =>  mysql,attributed,extensions_table
>
>
>
> Asterisk debug shows:
>
>
>  -- Attempting call on SIP/grnvoip/123804011818345 for [EMAIL 
> PROTECTED]:1
> (Retry 1)
>
>== Starting SIP/grnvoip-09592260 at 10,start,1 failed so falling back to
> exten 's'
>
>== Starting SIP/grnvoip-09592260 at 10,s,1 still failed so falling back to
> context 'default'
>
> [Nov  5 19:04:42] WARNING[29109]: pbx.c:2470 __ast_pbx_run: Channel
> 'SIP/grnvoip-09592260' sent into invalid extension 's' in context 'default',
> but no invalid handler
>
>
> This is with a call file that looks like:
>
>
> Channel: SIP/grnvoip/123804011818345XXX
> MaxRetries: 0
> RetryTime: 60
> WaitTime: 30
> Context: 10
> Extension: start
> Priority: 1
>
>
> And in the database the context 10, extension start and priority 1 does
> exist as shown below:
>
> mysql>  select context,exten,priority,app from extensions_table limit 0,3;
> +-+---+--++
> | context | exten | priority | app|
> +-+---+--++
> | 10  | start | 1| Set|
> | 10  | start | 2| AMD|
> | 10  | start | 3| WaitforSilence |
> +-+---+--++
>
>
>
> Any ideas on where to begin w/ the debug would be very appreciated.

Are you doing the switch from the dialplan in the [10] context?

Have a look at the realtime page on the wiki (voip-info.org)

-- 
Kind Regards,

Matt Riddell
Director
___

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Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem

[Patrick Maartense]

Not really, I have all these files, I just mentioned the fields that are filled 
by my queries ( nex time I will post the complete create statements )

But I think (HOPE) I found the problem

I only had SIPPEERS defined in the config
NOT sipusers

Now I have not seen these problems anymore (at least for now)


Tnx anyway..
Reg
PM

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Montag, 08. September 2008 17:22
To: Asterisk Users
Subject: Re: [asterisk-users] Asterisk realtime MySQL clients from same IP 
problem

Patrick Maartense schrieb:
> Off course the columns all are fine ( right case)

Ok.

> There seems to be an irregularity between the sip peers table and the Sip 
> registry.

> But any idea on the backgrounds of this behaviour??

The SQL query tries to update the ipaddr, port and regseconds fields
which your table is lacking. Thus the query fails.

I'm not sure if that's the cause of your problem but there's no
point in spending time on trying to figure out why something
doesn't work if the setup is wrong in the first place.  :-)

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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No virus found in this incoming message.
Checked by AVG - http://www.avg.com 
Version: 8.0.169 / Virus Database: 270.6.18/1658 - Release Date: 07.09.2008 
15:30


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Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem

[Philipp Kempgen]

Patrick Maartense schrieb:
> Off course the columns all are fine ( right case)

Ok.

> There seems to be an irregularity between the sip peers table and the Sip 
> registry.

> But any idea on the backgrounds of this behaviour??

The SQL query tries to update the ipaddr, port and regseconds fields
which your table is lacking. Thus the query fails.

I'm not sure if that's the cause of your problem but there's no
point in spending time on trying to figure out why something
doesn't work if the setup is wrong in the first place.  :-)

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem

[Patrick Maartense]

Off course the columns all are fine ( right case) ( to much german language 
makes one write words with uppercase almost every word :(

There seems to be an irregularity between the sip peers table and the Sip 
registry.

For the Client. Well you know, one that has the installed base, also set the 
standards. (try converting CRLF to CR only on yours :) )

But any idea on the backgrounds of this behaviour??




-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp Kempgen
Sent: Montag, 08. September 2008 16:29
To: Asterisk Users
Subject: Re: [asterisk-users] Asterisk realtime MySQL clients from same IP 
problem

Patrick Maartense schrieb:

> Users are creeated in the sippers table with following Fields set
> 
> Name :  .unique 
> 
> Host : dynamic
> 
> Nat : yes
> 
> Type: friend
> 
> Callerid:  .unique value
> 
> Context: autocreate
> 
> Secret : xx
> 
> Disallow: all
> 
> Allow : all
> 
> Username : unique : same as Name

Use lowercase column names.

> [Sep  8 15:37:44] DEBUG[3448] res_config_mysql.c: MySQL RealTime: Update
> SQL: UPDATE sippeers SET ipaddr = '127.0.0.1', port = '5102', regseconds
> = '1220884664', username = '92543036' WHERE name = '92543036'

> Anyone has an idea what to do here ?

Where are the columns in following: ipaddr, port, regseconds?

Apart from that I'd appreciate if you could get a better email
client which does not insert so many useless blank lines. :-) SCNR.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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No virus found in this incoming message.
Checked by AVG - http://www.avg.com 
Version: 8.0.169 / Virus Database: 270.6.18/1658 - Release Date: 07.09.2008 
15:30


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Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem

[Jay R. Ashworth]

On Mon, Sep 08, 2008 at 04:28:52PM +0200, Philipp Kempgen wrote:
> Apart from that I'd appreciate if you could get a better email
> client which does not insert so many useless blank lines. :-) SCNR.

Likely, for you, like me, it's not that his email client is indersting
blank lines... it's that whatever you're using to render his HTML email
into text is doing it -- for me, it's lynx under Mutt.

He probably just needs to find the "send HTML email" knob and turn it
off.

*Break* it off, by preference, but what can you do.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth & Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

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 Those who count the vote decide everything.
   -- (Josef Stalin)

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Re: [asterisk-users] Asterisk realtime MySQL clients from same IP problem

[Philipp Kempgen]

Patrick Maartense schrieb:

> Users are creeated in the sippers table with following Fields set
> 
> Name :  .unique 
> 
> Host : dynamic
> 
> Nat : yes
> 
> Type: friend
> 
> Callerid:  .unique value
> 
> Context: autocreate
> 
> Secret : xx
> 
> Disallow: all
> 
> Allow : all
> 
> Username : unique : same as Name

Use lowercase column names.

> [Sep  8 15:37:44] DEBUG[3448] res_config_mysql.c: MySQL RealTime: Update
> SQL: UPDATE sippeers SET ipaddr = '127.0.0.1', port = '5102', regseconds
> = '1220884664', username = '92543036' WHERE name = '92543036'

> Anyone has an idea what to do here ?

Where are the columns in following: ipaddr, port, regseconds?

Apart from that I'd appreciate if you could get a better email
client which does not insert so many useless blank lines. :-) SCNR.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Wednesday 27 August 2008 09:29:55 Tilghman Lesher wrote:
> On Wednesday 27 August 2008 08:56:02 J.M. wrote:
> > On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher <
> >
> > [EMAIL PROTECTED]> wrote:
> > > Given that this is the case, we may want to do one of the following:
> > > a) document that qualify=yes is incompatible with realtime, unless
> > > rtcachefriends is turned on, b) automatically disallow qualify=yes if
> > > the peer is realtime and caching is not turned on, or c) automatically
> > > cache realtime peers whose qualify field is set to yes.
> >
> > What about an option d) If qualify=yes then Asterisk checks every 2
> > seconds, if qualify=no then Asterisk does not check, and if
> > qualify= then Asterisk checks every 
> > milliseconds.  That would seem to conform to how it behaves in sip.conf
> > (reference:
> > http://www.voip-info.org/wiki-Asterisk+config+sip.conf).
>
> Uh, no, you're misreading the documentation.  Qualify never checks every
> 2 seconds; the qualification time is the time in which a host must respond,
> not the frequency with which Asterisk checks the host.  The frequency is
> usually only once every 60 seconds.

I have implemented option B, with a clear warning in the logs not to do this.
Testing is requested.

http://bugs.digium.com/view.php?id=13383

-- 
Tilghman

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Wednesday 27 August 2008 08:56:02 J.M. wrote:
> On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher <
>
> [EMAIL PROTECTED]> wrote:
> > Given that this is the case, we may want to do one of the following:
> > a) document that qualify=yes is incompatible with realtime, unless
> > rtcachefriends is turned on, b) automatically disallow qualify=yes if the
> > peer is realtime and caching is not turned on, or c) automatically cache
> > realtime peers whose qualify field is set to yes.
>
> What about an option d) If qualify=yes then Asterisk checks every 2
> seconds, if qualify=no then Asterisk does not check, and if
> qualify= then Asterisk checks every 
> milliseconds.  That would seem to conform to how it behaves in sip.conf
> (reference:
> http://www.voip-info.org/wiki-Asterisk+config+sip.conf).

Uh, no, you're misreading the documentation.  Qualify never checks every
2 seconds; the qualification time is the time in which a host must respond,
not the frequency with which Asterisk checks the host.  The frequency is
usually only once every 60 seconds.

-- 
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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

On Mon, Aug 25, 2008 at 6:21 PM, Tilghman Lesher <
[EMAIL PROTECTED]> wrote:

> Given that this is the case, we may want to do one of the following:
> a) document that qualify=yes is incompatible with realtime, unless
> rtcachefriends is turned on, b) automatically disallow qualify=yes if the
> peer is realtime and caching is not turned on, or c) automatically cache
> realtime peers whose qualify field is set to yes.
>
>
What about an option d) If qualify=yes then Asterisk checks every 2 seconds,
if qualify=no then Asterisk does not check, and if qualify=
then Asterisk checks every  milliseconds.  That would seem to
conform to how it behaves in sip.conf (reference:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf).

jm
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Re: [asterisk-users] asterisk realtime

[Szasz Szabolcs]

Hi list! Thank for the help. Now, I can call the 8500 to listen to the 
inbound messages, change pin, but I have another problem. When I call a 
SIP extension configured in the MySQL database it says: "Call from '101' 
to extension '102' rejected because extension not found." My vmusers 
table: 
+--+-+-+-+--+--+--+---+-+
 
| uniqueid | customer_id | context | mailbox | password | fullname | 
email | pager | stamp | 
+--+-+-+-+--+--+--+---+-+
 
| 1 | 101 | default | 101 | 264241 | | [EMAIL PROTECTED] | NULL | 2008-08-12 
11:59:34 | | 2 | 102 | default | 102 | NULL | | [EMAIL PROTECTED] | NULL | 
2008-08-12 11:59:40 | 
+--+-+-+-+--+--+--+---+-+
 
sipusers table: 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
| name | username | type | secret | host | callerid | context | mailbox 
| nat | qualify | fromuser | authuser | fromdomain | insecure | 
canreinvite | disallow | allow | restrictcid | defaultip | ipaddr | port 
| regseconds | 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
| 101 | 101 | friend | NULL | home | NULL | default | 101 | yes | no | 
101 | NULL | home | NULL | no | NULL | NULL | NULL | home | home | 5060 
| NULL | | 102 | 102 | friend | NULL | home | NULL | default | 102 | yes 
| no | 102 | NULL | home | NULL | no | NULL | NULL | NULL | home | home 
| 5060 | NULL | 
+--+--+++--+--+-+-+-+-+--+--++--+-+--+---+-+---++--++
 
Can you see the problem? Please help. Szasz Szabolcs 
-- Message: 16 Date: Mon, 25 Aug 2008 
10:23:31 -0500 From: Tilghman Lesher <[EMAIL PROTECTED]> 
Subject: Re: [asterisk-users] asterisk realtime To: Asterisk Users 
Mailing List - Non-Commercial Discussion 
 Message-ID: 
<[EMAIL PROTECTED]> Content-Type: 
text/plain; charset="iso-8859-1" On Monday 25 August 2008 07:08:30 Szasz 
Szabolcs wrote:

> > Hi!
> > I am running CentOS 5 with Asterisk 1.4.21.2  I am trying to setup storage
> > of voicemail messages into MySQL. I installed unixODBC unixODBC-devel
> > libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured
> > and built Asterisk, using menuconfig to turn on ODBC voicemail storage. 
> > Here is the output of some config files:
> >
> > [MySQL]
> > Description = ODBC for MySQL
> > Driver  = /usr/lib/libmyodbc3.so
> > Setup   = /usr/lib/libodbcmyS.so
> > UsageCount  = 3
> >
> > [MySQL ODBC 3.51 Driver]
> > Description = ODBC 3.51 for MySQL
> > DRIVER  = /usr/lib/libmyodbc3.so
> > SETUP   = /usr/lib/libmyodbc3S.so
> > UsageCount  = 3
> >
> > [EMAIL PROTECTED] ~]# cat /usr/local/etc/odbc.ini
> > [astrealtime]
> > Description = MySQL Asterisk database
> > Trace   = Off
> > TraceFile   = stderr
> > Driver  = MySQL
> > SERVER  = localhost
> > USER= asterisk
> > PASSWORD= 123qwe
> > PORT= 3306
> > DATABASE= asterisk
> >
> > [EMAIL PROTECTED] ~]# cat /etc/asterisk/res_odbc.conf
> > ;;; odbc setup file
> >
> > ; ENV is a global set of environmental variables that will get set.
> > ; Note that all environmental variables can be seen by all connections,
> > ; so you can't have different values for different connections.
> > [ENV]
> > INFORMIXSERVER => my_special_database
> > INFORMIXDIR => /opt/informix
> >
> > ; All other sections are arbitrary names for database connections.
> >
> > [asterisk]
> > enabled => yes
> > dsn => astrealtime
> > username => asterisk
> > password => 123qwe
> > pre-connect => yes
> >
> >
> > ;[mysql2]
> > ;enabled => no
> > ;dsn => MySQL-asterisk
> > ;username => myuser
> > ;password => mypass
> > ;pre-connect => yes
> > ;
> > ; On some databases, the connection times out and a reconnection will be
> > 

Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Rob Hillis]

Tilghman Lesher wrote:
> Given that this is the case, we may want to do one of the following:
> a) document that qualify=yes is incompatible with realtime, unless
> rtcachefriends is turned on, b) automatically disallow qualify=yes if the
> peer is realtime and caching is not turned on, or c) automatically cache
> realtime peers whose qualify field is set to yes.
>
> I am open to discussion and suggestions.  Option a is certainly the least
> invasive, which doesn't change any behavior, so it's the default, but I think
> that enough people would consider this behavior a bug that we might change
> it.  The only question that I have is, change it to what?

My vote is option (c) with a note clearly documenting this.

The other option, of course is only to cache the required fields for 
qualify to be set to yes - IP address and port?


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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Monday 25 August 2008 17:07:18 J.M. wrote:
> > I have also reproduced the problem, albeit on the same machine, by
> > setting the "qualify" field to "yes".
>
> A more complete explanation of how I am able to reproduce the issue is:
>
> 1) Disconnect the client.
> 2) Change the client's record's "sip.qualify" field to "yes".
> 3) Connect the client.
>
> If I watch the MySQL query log I see at least a dozen of the aforementioned
> queries per second after the client has connected.
>
> To undo this:
>
> 1) Disconnect the client.
> 2) Change the client's record's "sip.qualify" field to "no".
> 3) Connect the client.
> 4) Restart the Asterisk service.
>
> Then the queries return to a normal volume.

Given that this is the case, we may want to do one of the following:
a) document that qualify=yes is incompatible with realtime, unless
rtcachefriends is turned on, b) automatically disallow qualify=yes if the
peer is realtime and caching is not turned on, or c) automatically cache
realtime peers whose qualify field is set to yes.

I am open to discussion and suggestions.  Option a is certainly the least
invasive, which doesn't change any behavior, so it's the default, but I think
that enough people would consider this behavior a bug that we might change
it.  The only question that I have is, change it to what?

-- 
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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

>
> I have also reproduced the problem, albeit on the same machine, by setting
> the "qualify" field to "yes".
>
>
A more complete explanation of how I am able to reproduce the issue is:

1) Disconnect the client.
2) Change the client's record's "sip.qualify" field to "yes".
3) Connect the client.

If I watch the MySQL query log I see at least a dozen of the aforementioned
queries per second after the client has connected.

To undo this:

1) Disconnect the client.
2) Change the client's record's "sip.qualify" field to "no".
3) Connect the client.
4) Restart the Asterisk service.

Then the queries return to a normal volume.

jm
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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

On Mon, Aug 25, 2008 at 4:05 PM, Al Baker <[EMAIL PROTECTED]> wrote:

>
> J.M. wrote:
> > On Thursday 21 August 2008 10:08:53 J.M. wrote:
> > > I am running Asterisk 1.4.21.2  with Realtime.
> >  I have a phone setup in the
> > > database and when I connect that phone to Asterisk there are
> > suddenly an
> > > endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> > > 'dynamic'" queries being run.  The only way to stop the flood of
> > queries
> > > coming from Asterisk to restart the Asterisk process.  Even
> > disconnecting
> > > the phone doesn't stop Asterisk from running the queries.
> > >
> > > Has anyone seen this before?  Why would Asterisk do that and
> > does anyone
> > > know the fix?
>
[...]

> >
> > Another way, which has worked so far for me, is to set the "qualify"
> > field in the "sip" table (or whatever you called the table that
> > corresponds to the sip.conf file) to "no".  I found this out from
> > reading the following URL:
> > http://www.asteriskguru.com/tutorials/peer_is_now_unreachable.html
> >
> Ok, but WHY is he getting an "ENDLESS" # of selects.
> Sure * needs to get the data, but unless he had an ENDLESS series of
> CALLS to/from that phone
> should * be making all those queries ??
>
> and
>  HOW is this going to scale up ?
>

Those are very good questions and I do not have the answers.  This page:
http://www.voip-info.org/wiki-Asterisk+config+sip.conf says that "qualify"
should default to "no".  It may be possible that if the "qualify" field in
the database is left blank or NULL then Realtime assumes it to be a 0.
Which would make Asterisk check the client every 0 milliseconds.

However, my "qualify" field was set to "yes".  I have also reproduced the
problem, albeit on the same machine, by setting the "qualify" field to
"yes".

jm
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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Al Baker]

J.M. wrote:
> On Fri, Aug 22, 2008 at 7:41 PM, Tilghman Lesher 
> <[EMAIL PROTECTED] 
> > wrote:
>
> On Thursday 21 August 2008 10:08:53 J.M. wrote:
> > I am running Asterisk 1.4.21.2  with Realtime.
>  I have a phone setup in the
> > database and when I connect that phone to Asterisk there are
> suddenly an
> > endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> > 'dynamic'" queries being run.  The only way to stop the flood of
> queries
> > coming from Asterisk to restart the Asterisk process.  Even
> disconnecting
> > the phone doesn't stop Asterisk from running the queries.
> >
> > Has anyone seen this before?  Why would Asterisk do that and
> does anyone
> > know the fix?
>
> Asterisk does that because realtime data is not cached by default,
> so for each
> access of the peer in question, Asterisk needs to reload the data
> on the peer
> from the database.  If you'd like, turn on rtcachefriends in
> sip.conf, which
> will cache the peer for the duration of the registration interval
> (or whatever
> you have rtexpire set to).  Also, to get correct behavior on
> reload, you'll
> need to have rtupdate turned on.  Some of the behavior isn't quite
> right in
> 1.4.21.2 , even, but it should be fixed once
> 1.4.22 is released.
>
> BTW, I would otherwise have responded sooner, but I am on vacation
> this week,
> and I am not responding to email as quickly as I would usually.
>
>
> Another way, which has worked so far for me, is to set the "qualify" 
> field in the "sip" table (or whatever you called the table that 
> corresponds to the sip.conf file) to "no".  I found this out from 
> reading the following URL: 
> http://www.asteriskguru.com/tutorials/peer_is_now_unreachable.html
>
> If this continues to work it has the advantages of putting as little 
> in the .conf files as is possible and keeping the real-time feel of 
> using a database without having to worry about whether the cache is 
> updated or not.
>
> jm
Ok, but WHY is he getting an "ENDLESS" # of selects.
Sure * needs to get the data, but unless he had an ENDLESS series of 
CALLS to/from that phone
should * be making all those queries ??

and
 HOW is this going to scale up ?

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[J . M .]

On Fri, Aug 22, 2008 at 7:41 PM, Tilghman Lesher <
[EMAIL PROTECTED]> wrote:

> On Thursday 21 August 2008 10:08:53 J.M. wrote:
> > I am running Asterisk 1.4.21.2 with Realtime.  I have a phone setup in
> the
> > database and when I connect that phone to Asterisk there are suddenly an
> > endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> > 'dynamic'" queries being run.  The only way to stop the flood of queries
> > coming from Asterisk to restart the Asterisk process.  Even disconnecting
> > the phone doesn't stop Asterisk from running the queries.
> >
> > Has anyone seen this before?  Why would Asterisk do that and does anyone
> > know the fix?
>
> Asterisk does that because realtime data is not cached by default, so for
> each
> access of the peer in question, Asterisk needs to reload the data on the
> peer
> from the database.  If you'd like, turn on rtcachefriends in sip.conf,
> which
> will cache the peer for the duration of the registration interval (or
> whatever
> you have rtexpire set to).  Also, to get correct behavior on reload, you'll
> need to have rtupdate turned on.  Some of the behavior isn't quite right in
> 1.4.21.2, even, but it should be fixed once 1.4.22 is released.
>
> BTW, I would otherwise have responded sooner, but I am on vacation this
> week,
> and I am not responding to email as quickly as I would usually.
>

Another way, which has worked so far for me, is to set the "qualify" field
in the "sip" table (or whatever you called the table that corresponds to the
sip.conf file) to "no".  I found this out from reading the following URL:
http://www.asteriskguru.com/tutorials/peer_is_now_unreachable.html

If this continues to work it has the advantages of putting as little in the
.conf files as is possible and keeping the real-time feel of using a
database without having to worry about whether the cache is updated or not.

jm
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Re: [asterisk-users] asterisk realtime

[Tilghman Lesher]

On Monday 25 August 2008 07:08:30 Szasz Szabolcs wrote:
> Hi!
> I am running CentOS 5 with Asterisk 1.4.21.2  I am trying to setup storage
> of voicemail messages into MySQL. I installed unixODBC unixODBC-devel
> libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured
> and built Asterisk, using menuconfig to turn on ODBC voicemail storage. 
> Here is the output of some config files:
>
> [MySQL]
> Description = ODBC for MySQL
> Driver  = /usr/lib/libmyodbc3.so
> Setup   = /usr/lib/libodbcmyS.so
> UsageCount  = 3
>
> [MySQL ODBC 3.51 Driver]
> Description = ODBC 3.51 for MySQL
> DRIVER  = /usr/lib/libmyodbc3.so
> SETUP   = /usr/lib/libmyodbc3S.so
> UsageCount  = 3
>
> [EMAIL PROTECTED] ~]# cat /usr/local/etc/odbc.ini
> [astrealtime]
> Description = MySQL Asterisk database
> Trace   = Off
> TraceFile   = stderr
> Driver  = MySQL
> SERVER  = localhost
> USER= asterisk
> PASSWORD= 123qwe
> PORT= 3306
> DATABASE= asterisk
>
> [EMAIL PROTECTED] ~]# cat /etc/asterisk/res_odbc.conf
> ;;; odbc setup file
>
> ; ENV is a global set of environmental variables that will get set.
> ; Note that all environmental variables can be seen by all connections,
> ; so you can't have different values for different connections.
> [ENV]
> INFORMIXSERVER => my_special_database
> INFORMIXDIR => /opt/informix
>
> ; All other sections are arbitrary names for database connections.
>
> [asterisk]
> enabled => yes
> dsn => astrealtime
> username => asterisk
> password => 123qwe
> pre-connect => yes
>
>
> ;[mysql2]
> ;enabled => no
> ;dsn => MySQL-asterisk
> ;username => myuser
> ;password => mypass
> ;pre-connect => yes
> ;
> ; On some databases, the connection times out and a reconnection will be
> ; necessary.  This setting configures the amount of time a connection
> ; may sit idle (in seconds) before a reconnection will be attempted.
> ;idlecheck => 3600
>
> ; Certain servers, such as MS SQL Server and Sybase use the TDS protocol,
> which ; limits the number of active queries per connection to 1.  By
> setting up pools ; of connections, Asterisk can be made to work with these
> servers. ;[sqlserver]
> ;enabled => no
> ;dsn => mickeysoft
> ;pooling => yes
> ;limit => 5
> ;username => oscar
> ;password => thegrouch
> ;pre-connect => yes
> ; Many databases have a default of '\' to escape special characters.  MS
> SQL ; Server does not.
> ;backslash_is_escape => no
>
>
> When I am testing the odbc-mysql connection it seems that's OK:
> [EMAIL PROTECTED] ~]# isql -v astrealtime
> +---+
>
> | Connected!|
> |
> | sql-statement |
> | help [tablename]  |
> | quit  |
>
> +---+
> SQL>
>
> But when I'm trying to access my voicemail from an extension I get these
> error messages is Asterisk CLI:
>
> [Aug 25 16:55:58] WARNING[5080]: res_odbc.c:463 ast_odbc_request_obj:
> Failed to connect to asterisk [Aug 25 16:55:58] ERROR[5080]:
> res_config_odbc.c:130 realtime_odbc: No database handle available with the
> name of 'asterisk' (check res_odbc.conf) [Aug 25 16:55:58] NOTICE[5080]:
> res_odbc.c:530 odbc_obj_connect: Connecting asterisk [Aug 25 16:55:58]
> WARNING[5080]: res_odbc.c:541 odbc_obj_connect: res_odbc: Error
> SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found,
> and no default driver specified
>
> Where is the problem? Please help!

Might be something as simple as having two different libraries installed, one
in /usr/local, and the other in /usr.  A simple way to fix this would be to:
ln -s /usr/local/etc/odbc.ini /etc/odbc.ini
ln -s /usr/local/etc/odbcinst.ini /etc/odbcinst.ini

If that works, you have library skew.  Remove either one of the library sets,
re-configure, and recompile Asterisk.

-- 
Tilghman

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Re: [asterisk-users] Asterisk Realtime pounds MySQL

[Tilghman Lesher]

On Thursday 21 August 2008 10:08:53 J.M. wrote:
> I am running Asterisk 1.4.21.2 with Realtime.  I have a phone setup in the
> database and when I connect that phone to Asterisk there are suddenly an
> endless number of "SELECT * FROM sip WHERE name = '1001' AND host =
> 'dynamic'" queries being run.  The only way to stop the flood of queries
> coming from Asterisk to restart the Asterisk process.  Even disconnecting
> the phone doesn't stop Asterisk from running the queries.
>
> Has anyone seen this before?  Why would Asterisk do that and does anyone
> know the fix?

Asterisk does that because realtime data is not cached by default, so for each
access of the peer in question, Asterisk needs to reload the data on the peer
from the database.  If you'd like, turn on rtcachefriends in sip.conf, which
will cache the peer for the duration of the registration interval (or whatever
you have rtexpire set to).  Also, to get correct behavior on reload, you'll
need to have rtupdate turned on.  Some of the behavior isn't quite right in
1.4.21.2, even, but it should be fixed once 1.4.22 is released.

BTW, I would otherwise have responded sooner, but I am on vacation this week,
and I am not responding to email as quickly as I would usually.

-- 
Tilghman

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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Saul Bejarano]

I took it out of the field name used by FreePBX on the database.
If you install FreePBX you can see it on the mysql table list.

Saul

Todd Fulton wrote:
> hi saul,
> 
> could you give me more info on the "VMX-CONTEXT" concept?  i tried to 
> google it, but could find nothing.
> 
> i am trying to do exactly what you state in terms of creating a virtual 
> slice of the box for each user.  thanks!
> 
> 
> todd
> 
> ---- Original Message ----
>     Subject: Re: [asterisk-users] asterisk realtime and creating "new"
> contexts
> From: Saul Bejarano <[EMAIL PROTECTED]>
> Date: Thu, August 14, 2008 8:01 pm
> To: asterisk-users@lists.digium.com
> 
> 
> The VMX-CONTEXT can be managed from database
> 
> mysql> select * from globals;
> But you will have to specify the same on each extension under sip
> once the extension under table sip is the one that calls the CONTEXT to
> route the call.
> 
> | PARKNOTIFY | SIP/200 |
> | RECORDEXTEN | "" |
> | RINGTIMER | 15 |
> | DIRECTORY | last |
> | AFTER_INCOMING | |
> | IN_OVERRIDE | forcereghours |
> | REGTIME | 7:55-17:05 |
> | REGDAYS | mon-fri |
> | DIRECTORY_OPTS | |
> | DIALOUTIDS | 1/2/3/4/5/6/ |
> | OUT_1 | ZAP/g0 |
> | VM_PREFIX | * |
> | VM_OPTS | |
> | VM_GAIN | |
> | VM_DDTYPE | u |
> | TIMEFORMAT | kM |
> | TONEZONE | us |
> | ALLOW_SIP_ANON | yes |
> | VMX_CONTEXT | from-internal
> 
> It will make it sort of complicated thought because it was build to be
> the generic element routing the calls out of the SIP Registrar, by
> having individual Context what you are trying to do is partition one
> Asterisk box to function as multiple offering invidual termination to
> each one of your users or complete management of a virtual slice of the
> box :)
> 
> Todd Fulton wrote:
>  > Hi,
>  >
>  > I'm trying to create a multi-tennant asterisk installation 
> where
>  > each of my customers has its own context. Well, I've got asterisk
>  > realtime working, and I can add/update extensions to existing
> contexts
>  > in extensions.conf without a problem. However, when I attempt to
> create
>  > database entries with a context that is NOT in extensions.conf, I
> get an
>  > error "invalid extension".
>  >
>  > I've found several posts around the net asking this question, but no
>  > answers. Has anyone out there dealt with this problem?
>  >
>  > Any help would be great!
>  >
>  >
>  > Todd
>  >
>  >
>  >
> 
>  >
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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Todd Fulton]

thanks!  this definitely helps.  now, i'm trying to think of a way to make this happen on multiple asterisk nodes at once.  i really wish realtime would simply read new contexts from the db.  i know that the dialplan is core to the system, but i think this core aspect should be ultimately configurable from a database using realtime.  sigh.todd 

 Original Message 
Subject: Re: [asterisk-users] asterisk realtime and creating "new"
contexts
From: Mike Clark <[EMAIL PROTECTED]>
Date: Fri, August 15, 2008 4:49 am
To: Asterisk Users Mailing List - Non-Commercial Discussion



Todd Fulton wrote:
> Hi,
>
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to 
> create database entries with a context that is NOT in extensions.conf, 
> I get an error "invalid extension".
>
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
>
> Any help would be great!
>
>
> Todd
>
Todd:

Unfortunately, new contexts don't seem to show up in "real time". I 
solved this in RAGUI by putting #exec statements in the extensions.cong 
file that scan the extensions table and generate the proper contexts. 
However, you still have to do a reload to get the contexts to be 
available in Asterisk.

Here is an example:

in extensions.conf

#exec /opt/pointcall/asterisk/scripts/load_extensions.rb

I used Ruby, but it could be Perl , PHP or whatever

load_extensions.rb

#!/usr/local/bin/ruby
#

require 'mysql'

hostname = "host"
username = "user"
password = "pass"
database = "rtdb"

my = Mysql.new(hostname, username, password, database)

res = my.query("SELECT DISTINCT context FROM extensions ORDER by context")
#
res.each do |row|
context = row[0]
print "\n"
print '[' + context + "]\n"
print "Switch => Realtime/" + context + "\n"
end


Thanks,

Mike

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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Mike Clark]

Todd Fulton wrote:
> Hi,
>
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to 
> create database entries with a context that is NOT in extensions.conf, 
> I get an error "invalid extension".
>
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
>
> Any help would be great!
>
>
> Todd
>
Todd:

Unfortunately, new contexts don't seem to show up in "real time". I 
solved this in RAGUI by putting #exec statements in the extensions.cong 
file that scan the extensions table and generate the proper contexts. 
However, you still have to do a reload to get the contexts to be 
available in Asterisk.

Here is an example:

in extensions.conf

#exec /opt/pointcall/asterisk/scripts/load_extensions.rb

I used Ruby, but it could be Perl , PHP or whatever

load_extensions.rb

#!/usr/local/bin/ruby
#

require 'mysql'

hostname = "host"
username = "user"
password = "pass"
database = "rtdb"

my = Mysql.new(hostname, username, password, database)

res = my.query("SELECT DISTINCT context FROM extensions ORDER by context")
#
res.each do |row|
context = row[0]
print "\n"
print '[' + context + "]\n"
print "Switch => Realtime/" + context + "\n"
end


Thanks,

Mike

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Re: [asterisk-users] asterisk realtime and creating "new" contexts

[Todd Fulton]

hi saul,could you give me more info on the "VMX-CONTEXT" concept?  i tried to google it, but could find nothing.i am trying to do exactly what you state in terms of creating a virtual slice of the box for each user.  thanks!todd

 Original Message 
Subject: Re: [asterisk-users] asterisk realtime and creating "new"
contexts
From: Saul Bejarano <[EMAIL PROTECTED]>
Date: Thu, August 14, 2008 8:01 pm
To: asterisk-users@lists.digium.com


The VMX-CONTEXT can be managed from database

mysql> select * from globals;
But you will have to specify the same on each extension under sip
once the extension under table sip is the one that calls the CONTEXT to 
route the call.

| PARKNOTIFY   | SIP/200 |
| RECORDEXTEN  | ""  |
| RINGTIMER| 15  |
| DIRECTORY| last|
| AFTER_INCOMING   | |
| IN_OVERRIDE  | forcereghours   |
| REGTIME  | 7:55-17:05  |
| REGDAYS  | mon-fri |
| DIRECTORY_OPTS   | |
| DIALOUTIDS   | 1/2/3/4/5/6/|
| OUT_1| ZAP/g0  |
| VM_PREFIX| *   |
| VM_OPTS  | |
| VM_GAIN  | |
| VM_DDTYPE| u   |
| TIMEFORMAT   | kM  |
| TONEZONE | us  |
| ALLOW_SIP_ANON   | yes |
| VMX_CONTEXT  | from-internal

It will make it sort of complicated thought because it was build to be 
the generic element routing the calls out of the SIP Registrar, by 
having individual Context what you are trying to do is partition one 
Asterisk box to function as multiple offering invidual termination to 
each one of your users or complete management of a virtual slice of the 
box :)

Todd Fulton wrote:
> Hi,
> 
> I'm trying to create a multi-tennant asterisk installation  where 
> each of my customers has its own context.  Well, I've got asterisk 
> realtime working, and I can add/update extensions to existing contexts 
> in extensions.conf without a problem.  However, when I attempt to create 
> database entries with a context that is NOT in extensions.conf, I get an 
> error "invalid extension".
> 
> I've found several posts around the net asking this question, but no 
> answers.  Has anyone out there dealt with this problem?
> 
> Any help would be great!
> 
> 
> Todd
> 
> 
> 
> 
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