​​http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER​​

 

 

 


valid for openser 1.1.x and asterisk 1.2.x


​​OpenSER​​ is a pure ​​VoIP​​ signaling server using Session Initiation Protocol - ​​SIP​​. It is flexible and highly configurable but cannot be used to provide media services as voicemail, anounncements or conferencing. For such services, ​​Asterisk​​ is the most suitable open source product. In this document we present how to configure ​​Asterisk​​ to use ​​OpenSER​​'s subscribers database to provide voicemail service. A basic configuration file for ​​OpenSER​​ is posted down the page, allowing to have a functional system by following the steps in this tutorial.


Requirements


 


UnixODBC Installation




Get the sources from ​​​http://www.unixodbc.org​​, compile and install them on your system


cd /usr/local/src
wget http://www.unixodbc.org/unixODBC-2.2.11.tar.gz
tar xvfz unixODBC-2.2.11.tar.gz
cd unixODBC-2.2.11
./configure –enable-gui=no
make
make install


NOTE: if you get error during compilation in 'sqp/lex.l', the line 240, related to 'YY_FLUSH_BUFFER', you can safely comment/remove that line.

NOTE: you must have /usr/local/lib in your /etc/ld.so.conf file or LD_LIBRARY_PATH environment variable.


MySQL Installation




You can install ​​MySQL​​ using the packaging system from you Linux distribution. The only requirements is to be ​​MySQL​​ 5.0+. For example, ​​http://dotdeb.org​​​ provides packages for Debian stable.

After installation, you can set the MySQL root password with a command like:


/usr/bin/mysqladmin -u root password 'your-new-password'




Asterisk Installation




Get Asterisk sources from ​​​http://www.asterisk.org​​. At this moment Asterisk 1.2.9.1 is the latest stable version.


cd /usr/local/src
wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.9.1.tar.gz
tar xvfz asterisk-1.2.9.1.tar.gz
cd asterisk-1.2.9.1


Edit 'apps/Makefile' and uncomment lines:


CFLAGS+=-DUSE_ODBC_STORAGE
CFLAGS+=-DEXTENDED_ODBC_STORAGE


Edit 'apps/app_voicemail.c' and change the size of memeber 'uniqueid' in 'struct ast_vm_user' to 64:


/* Structure for linked list of users */
struct ast_vm_user {
       char context[AST_MAX_CONTEXT];  /* Voicemail context */
       char mailbox[AST_MAX_EXTENSION];/* Mailbox id, unique within vm context
       char password[80];              /* Secret pin code, numbers only */
       char fullname[80];              /* Full name, for directory app */
       char email[80];                 /* E-mail address */
       char pager[80];                 /* E-mail address to pager (no attachme
       char serveremail[80];           /* From: Mail address */
       char mailcmd[160];              /* Configurable mail command */
       char language[MAX_LANGUAGE];    /* Config: Language setting */
       char zonetag[80];               /* Time zone */
       char callback[80];
       char dialout[80];
       char uniqueid[64];              /* Unique integer identifier */ 
       char exit[80];
       unsigned int flags;             /* VM_ flags */
       int saydurationm;
       int maxmsg;                     /* Maximum number of msgs per folder fo
       struct ast_vm_user *next;
};


Proceed with usual ​​Asterisk​​ installation:


make
make install




OpenSER Installation




You can download latest stable version via CVS snapshots. For branch 1.0.0 (latest release in this branch is 1.0.1) you can do:


cd /usr/local/src
wget http://www.openser-project.org/downloads/snapshots/openser-1.0.0/openser-1.0.0-cvs-latest.tgz
tar xvfz openser-1.0.0-cvs-latest.tgz
cd openser-1.0.0
make all include_modules="mysql"
make install include_modules="mysql"




UnixODBC MySQL Driver Installation




Simply install the package using the tools from your linux distribution. For example, for Debian:


apt-get install libmyodbc




Create OpenSER Database




To create the database needed by ​​OpenSER​​:


/usr/local/sbin/openser_mysql.sh create


This will create a database named 'openser' and will add a ​​MySQL​​ user 'openser' with full access to it. The default password is 'openserrw', do change it before (by editing usr/local/sbin/openser_mysql.sh) or immediately after you create the database.

Once you create the database, you need to add a new column to the 'subscriber' table to store the PIN for voicemail access:


ALTER TABLE subscriber ADD vmail_password varchar(32);




Create Asterisk Database




The database needed by ​​Asterisk​​ will contain two views ('vmusers' and 'sipusers') of tables from ​​OpenSER​​ database, therefore it is required to have ​​MySQL​​ 5.0+ since the views were introduced in this version. There is a real ​​MySQL​​ table ('voicemessages') which will store the voice messages.

Log in as root in ​​MySQL​​ server:


create database asterisk;

use asterisk;

CREATE TABLE `voicemessages` (
  `id` int(11) NOT NULL auto_increment,
  `msgnum` int(11) NOT NULL default '0',
  `dir` varchar(80) default '',
  `context` varchar(80) default '',
  `macrocontext` varchar(80) default '',
  `callerid` varchar(40) default '',
  `origtime` varchar(40) default '',
  `duration` varchar(20) default '',
  `mailboxuser` varchar(80) default '',
  `mailboxcontext` varchar(80) default '',
  `recording` longblob,
  PRIMARY KEY  (`id`),
  KEY `dir` (`dir`)
) ENGINE=InnoDB;

CREATE VIEW vmusers AS
SELECT phplib_id as uniqueid,
  username as customer_id,
  'default' as context,
  username as mailbox,
  vmail_password as password,
  CONCAT(first_name,' ',last_name) as fullname,
  email_address as email,
  NULL as pager,
  datetime_created as stamp 
FROM openser.subscriber;

CREATE VIEW sipusers AS
SELECT username as name,
  username,
  'friend' as type,
  NULL as secret,
  domain as host,
  CONCAT(rpid, ' ','<',username,'>') as callerid,
  'default' as context,
  username as mailbox,
  'yes' as nat,
  'no' as qualify,
  username as fromuser,
  NULL as authuser,
  domain as fromdomain,
  NULL as insecure,
  'no' as canreinvite,
  NULL as disallow,
  NULL as allow,
  NULL as restrictcid,
  domain as defaultip,
  domain as ipaddr,
  '5060' as port,
  NULL as regseconds
FROM openser.subscriber;


Add a ​​MySQL​​ user which will have full access right to 'asterisk' database.


GRANT ALL ON asterisk.* to asterisk@localhost IDENTIFIED BY 'some_password';




Configure UnixODBC





In the file ‘/usr/local/etc/odbcinst.ini’ you must add:


[MySQL]
Description = MySQL driver
Driver = /usr/lib/odbc/libmyodbc.so
Setup = /usr/lib/odbc/libodbcmyS.so
CPTimeout =
CPReuse =
UsageCount = 1


In the file '/usr/local/etc/odbc.ini' you must add:


[MySQL-asterisk]
Description = MySQL Asterisk database
Trace = Off
TraceFile = stderr
Driver = MySQL
SERVER = localhost
USER = asterisk
PASSWORD = some_password
PORT = 3306
DATABASE = asterisk




Configure Asterisk




In '/etc/asterisk/res_odbc.conf':


[asterisk]
enabled => yes
dsn => MySQL-asterisk
username => asterisk
password => asterisk
pre-connect => yes


In '/etc/asterisk/extconfig.conf':


sipusers => odbc,asterisk,sipusers
sippeers => odbc,asterisk,sipusers
voicemail => odbc,asterisk,vmusers


In '/etc/asterisk/sip.conf':

If you want to enable ​​MWI​​, do not forget to set checkmwi attribute.


checkmwi=10


Guidelines about configuring the ​​SIP​​ channel you find at ​​http://www.voip-info.org/wiki-Asterisk+config+sip.conf​​​. You do not need to add any ​​SIP​​ user or peer in the configuration file, they will be loaded from database.

In ‘/etc/asterisk/voicemail.conf’ you do not need to add any mailbox. They will be loaded from database. The general configuration part of voicemail application is presented at ​​​http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf​​​

For our tutorial, we consider that the users will have 4-digit ID. To implement a clear dialing plan in ​​Asterisk​​ which allow extensibility and clear extentions for different services, the calls to voicemail will be prefixed with '1' in ​​OpenSER​​ proxy. This prefix will be transpartent for users. If voice mailbox does not exist, ​​Asterisk​​ will play "invalid extension" message.

In '/etc/asterisk/extensions.conf':


exten => 1,1,Ringing
exten => 1,2,VoicemailMain(${CALLERIDNUM})
exten => 1,3,Hangup

exten => 11,1,Ringing
exten => 11,2,VoicemailMain()
exten => 11,3,Hangup

exten => _1XXXX,1,Ringing
exten => _1XXXX,2,MailboxExists(${EXTEN:1})
exten => _1XXXX,3,Playback(invalid)
exten => _1XXXX,4,Hangup
exten => _1XXXX,103,Voicemail(u${EXTEN:1})
exten => _1XXXX,104,Hangup




Configure OpenSER




Dialing plan:
- local users have 4-digit extension
- to listen its voice messages from its ​​SIP​​ phone, the user has to dial *98 (Asterisk will prompt only for PIN)
- to listen its voice messages from another ​​SIP​​ phone, the user has to dial *981 (Asterisk will prompt for mailbox ID and PIN)
- to call directly to leave voice message to user XXXX, the user has to dial *89XXXX

In '/usr/local/etc/openser/openser.cfg': 


#
# $Id$
#

# ----------- global configuration parameters ------------------------

debug=3           # debug level (cmd line: -dddddddddd)
fork=yes          # daemonize
log_stderror=no   # (cmd line: -E)

check_via=no      # (cmd. line: -v)
dns=no            # (cmd. line: -r)
rev_dns=no        # (cmd. line: -R)
children=4
fifo="/tmp/openser_fifo"

listen=udp:10.10.10.10:5060

#
# ------------------ module loading ----------------------------------

# Uncomment this if you want to use SQL database
mpath="/usr/local/lib/openser/modules"
loadmodule "mysql.so"

loadmodule "xlog.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "avpops.so"
loadmodule "auth.so"
loadmodule "auth_db.so"
loadmodule "group.so"
loadmodule "uri.so"

# ----------------- setting module-specific parameters ---------------
modparam("usrloc|auth_db|avpops|group",
    "db_url", "mysql://openser:openserrw@localhost/openser")

# -- usrloc params --
# persistent storage
modparam("usrloc", "db_mode", 2)

# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config), 
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")

# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)


modparam("avpops", "avp_table", "usr_preferences")

# -------------------------  request routing logic -------------------

# main routing logic

route{

    # initial sanity checks -- messages with
    # max_forwards==0, or excessively long requests
    if (!mf_process_maxfwd_header("10")) {
        sl_send_reply("483","Too Many Hops");
        exit;
    };

    if (msg:len >=  2048 ) {
        sl_send_reply("513", "Message too big");
        exit;
    };

    # we record-route all messages -- to make sure that
    # subsequent messages will go through our proxy; that's
    # particularly good if upstream and downstream entities
    # use different transport protocol
    if (!method=="REGISTER")
        record_route();

    # subsequent messages withing a dialog should take the
    # path determined by record-routing
    if (loose_route()) {
        # mark routing logic in request
        append_hf("P-hint: rr-enforced/r/n"); 
        route(1);
    };

    if (!uri==myself) {
        # mark routing logic in request
        append_hf("P-hint: outbound/r/n"); 
        route(1);
    };

    # if the request is for other domain use UsrLoc
    # (in case, it does not work, use the following command
    # with proper names and addresses in it)
    if (uri==myself) {

        if (method=="REGISTER") {
            if (!www_authorize("openser.org", "subscriber")) {
                www_challenge("openser.org", "0");
                exit;
            };

            save("location");
            exit;
        };

        # requests for Media server
        if(is_method("INVITE") && !has_totag() && uri=~"sip:/*9") {
            route(3);
            exit;
        }

        # mark transaction if user is in voicemail group
        if(is_method("INVITE") && !has_totag()
            && is_user_in("Request-URI","voicemail"))
        {
            xdbg("user [$ru] has voicemail redirection enabled/n");
            # backup R-URI
            avp_write("$ruri", "i:10");
            setflag(2);
        };
        # native SIP destinations are handled using our USRLOC DB
        if (!lookup("location")) {
            if(isflagset(2)) {
                # route to Asterisk Media Server
                prefix("1");
                rewritehostport("10.10.10.11:5060");
                route(1);
            } else {
                sl_send_reply("404", "Not Found");
                exit;
            }
        };
        append_hf("P-hint: usrloc applied/r/n"); 
    };

    route(1);
}


route[1] {
    
    if(isflagset(2))
        t_on_failure("1");

    if (!t_relay()) {
        sl_reply_error();
    };
    exit;
}


# voicemail access
# - *98 - listen caller's voice messages, being prompted for pin
# - *981 - listen voice messages, being promted for mailbox and pin
# - *98XXXX - leave voice message to XXXX
#
route[3] {
    # direct voicemail
    if (uri =~ "sip:/*98@" ) {
            rewriteuser("1");
        xdbg("voicemail access/n");
    } else if (uri =~ "sip:/*981@" ) {
        strip(4);
        rewriteuser("11");
    } else if (uri =~ "sip:/*98.+@" ) {
        strip(3);
        prefix("1");
    } else {
        xlog("unknown media extension $rU/n");
        sl_send_reply("404", "Unknown media service");
        exit;
    }

    # route to Asterisk Media Server
    rewritehostport("10.10.10.11:5060");
    route(1);
}

failure_route[1] {
    if (t_was_cancelled()) {
        xdbg("transaction was cancelled by UAC/n");
        return;
    }
    # restore initial uri
    avp_pushto("$ruri", "i:10");
    prefix("1");
    # route to Asterisk Media Server
    rewritehostport("10.10.10.11:5060");
    resetflag(2);
    route(1);
}





See also