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http://opensips.com.br/wiki/index.php?title=Opensips_1.9

 

https://github.com/OpenSIPS/opensips/blob/master/INSTALL

 

 http://opensips.com.br/wp/index.php/nova-versao-1-10/


http://opensips.com.br/wp/

 

Opensips 1.9

Este tutorial tem por finalidade, demonstrar o processo de instalação e configuração do OpenSIPS 1.9 utilizando a opção do mesmo para gerar o arquivo de script.

Índice

 [ocultar

Informacoes

Para este tutorial foi utilizado um Debian 6.0 com kernel 64bits A versão utilizada do OpenSIPS foi 1.9.1


Dependencias

apt-get install gcc make libncurses5-dev libnewt-dev libxml2-dev unixodbc \ 
unixodbc-dev libmysqlclient15-dev libxmlrpc-c3-dev libexpat1-dev zlib1g-dev \ 
m4 bison flex libpcre3-dev mysql-server vim apache2-mpm-prefork libapache2-mod-php5 \
 php5-mysql php5-xmlrpc php-pear  ngrep g++ libjpeg62-dev libssl-dev 

Download

cd /usr/src/
wget -c http://opensips.org/pub/opensips/1.9.1/src/opensips-1.9.1_src.tar.gz
tar -xzvf opensips-1.9.1_src.tar.gz
cd opensips-1.9.1-tls

Compilação

./configure
make menuconfig

No menu selecione a opção Configure Compile Options , depois selecione Configure Excluded Modules

Selecione os seguintes módulos

  • db_mysql
  • dialplan
  • regex
  • mi_xmlrpc
  • presence
  • presence_dialoginfo
  • presence_mwi
  • presence_xml
  • xcap
  • xcap_client

Após as alterações salve os parametros e selecione no primeiro menu a opção Compile & Install OpenSIPS

O processo deverá demorar um pouco, ao termino você voltará ao menu, neste selecione a opção Generate OpenSIPS Script , selecione então Residential Script, e depois Configure Residencial Script

Neste marque as seguintes opções

  • USE_ALIASES
  • USE_AUTH
  • USE_DBACC
  • USE_DBUSRLOC
  • USE_DIALOG
  • USE_MULTIDOMAIN
  • USE_NAT
  • USE_PRESENCE
  • USE_DIALPLAN
  • HAVE_INBOUND_PSTN
  • HAVE_OUTBOUND_PSTN
  • USE_DR_PSTN

Com todas as opções selecionadas selecione para gerar o script, o script será gerado no diretório etc dentro do diretório dos fontes, no meu caso o nome do arquivo foi opensips_residential_2013-8-17_11:37:45.cfg , no seu caso a data obviamente estará diferente.

Configuração

Precisamos executar alguns procedimentos para darmos continuidade, seguem os comandos abaixo.

ln -s /usr/etc/opensips /etc/
cp packaging/debian/opensips.default /etc/default/opensips
cp packaging/debian/opensips.init /etc/init.d/opensips
chmod +x /etc/init.d/opensips
update-rc.d opensips defaults

Edite o arquivo /etc/default/opensips , a altere o parametro RUN_OPENSIPS para yes

groupadd opensips
mkdir /var/run/opensips
useradd -d /var/run/opensips/ -s /bin/false -g opensips opensips
chown -R opensips.opensips /var/run/opensips


Agora podemos dar continuidade.

opensipsctlrc

O arquivo /etc/opensips/opensipsctlrc possue alguns parametros que precisamos definir referente a nossa estrutura, você deve analisar o script para verificar oque mais você pode usar, mas em ambito geral abaixo estão as opções que precisamos.

SIP_DOMAIN:

  • este parametro é o dominio que o script usará para gerar os dados no banco de dados, você pode usar o ip do seu servidor ou um nome de dominio se você possuir.

DBENGINE DBHOST DBNAME DBRWUSER DBRWPW DBROOTUSER

* Estas opções são referentes ao banco de dados que utilizaremos, configure de acordo com seu ambiente

ALIAS_TYPE

* Esta opcão deve ser definida como DB, assim os alias das contas SIP estarão no banco de dados

MI_CONNECTOR_FIFO

* Defina esta opção como /tmp/opensips_fifo

Após configurar, vamos executar o comando abaixo para criar nosso banco de dados.

opensipsdbctl create opensips


Instalando o RTP Proxy

cd /usr/src/
wget -c http://b2bua.org/chrome/site/rtpproxy-1.2.1.tar.gz
tar -xzvf rtpproxy-1.2.1.tar.gz
cd rtpproxy-1.2.1
./configure
make
make install
groupadd rtpproxy
useradd -d /var/run/rtpproxy -s /bin/true -g rtpproxy rtpproxy
mkdir /var/log/rtpproxy
mkdir /var/run/rtpproxy
chown -R rtpproxy.rtpproxy /var/log/rtpproxy
chown -R rtpproxy.rtpproxy /var/run/rtpproxy

Arquivo de inicialização

/etc/init.d/rtpproxy

#!/bin/bash
#
# Este script e de autoria de Mike Tesliuk
# qualquer falha no mesmo por favor informe 
# atraves do email mike (a) tesliuk.com
#
### BEGIN INIT INFO
# Provides:          rtpproxy
# Required-Start:    $syslog $network $local_fs $time
# Required-Stop:     $syslog $network $local_fs
# Default-Start:     2 3 4 5
# Default-Stop:      0 1 6
# Short-Description: Start the RTPPROXY server
# Description:       Start the RTPPROXY server
### END INIT INFO

PATH=/usr/local/bin:/usr/local/sbin:/usr/bin:/usr/sbin:/bin:/sbin
USELOG=1
USER=rtpproxy
# Altere o ip abaixo para o ip de seu sistema
IPADDR="_SEU_IP_AQUI_"

. /lib/lsb/init-functions


start(){
        echo "Iniciando RTP PROXY "
        if [ -z $(pidof rtpproxy) ]; then
                if [ "${USELOG}" = "1" ]; then
                        echo "Iniciando com LOG"
                        /usr/local/bin/rtpproxy -l $IPADDR -s udp:127.0.0.1:7890 -u $USER -F -f d DBUG 2&> /var/log/rtpproxy/rtpproxy.log &
                else
                        echo "Iniciando sem LOG"
                        /usr/local/bin/rtpproxy -l $IPADDR -s udp:127.0.0.1:7890 -u $USER  -F -f d DBUG 2&> /dev/null
                fi

                if [ -n $(pidof rtpproxy) ]; then
                        echo "START OK"
                fi
        else
                echo "Processo ja em execucao"
        fi
}


stop(){

        if [ -z $(pidof rtpproxy) ]; then
                echo "Processo nao encontrado"
        else
                kill -9 $(pidof rtpproxy)
                if [ -n $(pidof rtpproxy) ]; then
                        echo "STOP OK"
                else
                        echo "Falha em realizar stop do servico"
                fi

        fi
}

case $1 in
        start)
                start
        ;;
        stop)
                stop
        ;;
        restart)
                stop
                start

        ;;
        *)
                echo "Utilize: stop | start | restart"
        ;;
esac

Vamos dar permissao e colocar na inicialização

chmod +x /etc/init.d/rtpproxy 
update-rc.d rtpproxy defaults
/etc/init.d/rtpproxy start

Arquivo autogerado

copie o arquivo que foi gerado para você para /etc/opensips/opensips.cfg

O Arquivo padrão precisa de alguns ajustes, basicamente localize as opções CUSTOMIZE ME existentes no arquivo.

Um parametro extra que vamos adicionar é a opção db_url para o modulo uri, para isso localize a opção onde está sendo carregado modulo uri.so e então adicione abaixo.

modparam("uri", "db_url",
        "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME

Termine de alterar os parametros onde traz a opção CUTOMIZE ME para os dados referentes.


Ao terminar as configurações temos que então configurar o nosso plano de discagem para chamadas externas.

Localize a opção dp_translate, comente a mesma, nós não a usaremos aqui, abaixo acrescente

        # definimos aqui pstn como padrao 0
        $avp(pstn)=0;
        xlog("Verificando $rU para do_routing");        

        # Ligacao local (achar uma forma para pegar o ddd na base do usuario)
        # routeid 0 para gateway de ligacoes locais (acrescentar 5511)
        if ($rU=~"^0[0-9]{8}$") {
                $avp(pstn)=1;
                $avp(routeid)=0;

        # ligacao ddd (0 + ddd + numero) 
        # routeid 1 para gateway para ddd (acrescenta 55)
        }else if( $rU =~ "^0[0-9]{2}[0-9]{8}$" ){
                $avp(pstn)=1;
                $avp(routeid)=1;


        # Ligacao ldn (00 + numero)
        # routeid 2 para remover o 00 e enviar a chamada diretamente
        }else if( $rU =~ "^00[0-9]+$" ){
                $avp(pstn)=1;
                $avp(routeid)=2;

        }

        # pstn esta definido e vamos rotear
        if($avp(pstn) == 1){
                xlog("Regra pre do_routing");

                # utilizamos o routeid que definimos para achar o gateway
                # correto para este perfil de chamada
                if (!do_routing("$avp(routeid)")) {
                        send_reply("500","No PSTN Route found");
                        exit;
                }

                route(relay);
                exit;
        }


Descendo um pouco o arquivo, após a sessão do if(!lookup('location','m'), acrescente a seguinte opção.

rtpproxy_offer();

Esta opção deve estar na linha anterior a linha abaixo

if (isbflagset(NAT)) setflag(NAT);


Seu arquivo deverá estar semelhante ao arquivo abaixo.

#
# $Id: opensips_residential.m4 9742 2013-02-05 10:24:48Z vladut-paiu $
#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team@opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
#   the "Residential" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

debug=3
log_stderror=no
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */
debug=6
fork=no
log_stderror=yes

/* uncomment the next line to enable the auto temporary blacklisting of 
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns 
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* comment the next line to enable the auto discovery of local aliases
   based on revers DNS on IPs */
auto_aliases=no


listen=udp:_SEU_IP_AQUI_:5060   # CUSTOMIZE ME


disable_tcp=yes

disable_tls=yes


####### Modules Section ########

#set module path
mpath="/usr/lib/opensips/modules/"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timer", 5)
modparam("tm", "fr_inv_timer", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)


#### URI module
loadmodule "uri.so"
modparam("uri", "use_uri_table", 0)
modparam("uri", "db_url", "mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips")

  





#### MYSQL module
loadmodule "db_mysql.so"



#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "db_mode",   2)
modparam("usrloc", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME


#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "failed_transaction_flag", "ACC_FAILED")
/* account triggers (flags) */
modparam("acc", "db_flag", "ACC_DO")
modparam("acc", "db_missed_flag", "ACC_MISSED")
modparam("acc", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME


#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")


#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME


#### DOMAIN module
loadmodule "domain.so"
modparam("domain", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME
modparam("domain", "db_mode", 1)   # Use caching
modparam("auth_db|usrloc|uri", "use_domain", 1)

### XCAP
loadmodule "xcap.so"
modparam("xcap", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME
modparam("xcap", "integrated_xcap_server", 1)

#### PRESENCE modules
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("presence", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address", "sip:127.0.0.1:5060") # CUSTOMIZE ME


#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME


####  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "received_avp", "$avp(received_nh)")

loadmodule "rtpproxy.so"
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7890") # CUSTOMIZE ME


####  DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME


####  DYNAMMIC ROUTING module
loadmodule "drouting.so"
modparam("drouting", "db_url",
	"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME




####### Routing Logic ########

# main request routing logic

route{
	force_rport();
	
	if (nat_uac_test("8")) {
		if (is_method("REGISTER")) {
			fix_nated_register();
			setbflag(NAT);
		} else {
			fix_nated_contact();
			setflag(NAT);
		}
	}
 	

	if (!mf_process_maxfwd_header("10")) {
		sl_send_reply("483","Too Many Hops");
		exit;
	}

	if (has_totag()) {
		# sequential request withing a dialog should
		# take the path determined by record-routing
		if (loose_route()) {
			
			# validate the sequential request against dialog
			if ( $DLG_status!=NULL && !validate_dialog() ) {
				xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
				## exit;
			}
			
			if (is_method("BYE")) {
				setflag(ACC_DO); # do accounting ...
				setflag(ACC_FAILED); # ... even if the transaction fails
			} else if (is_method("INVITE")) {
				# even if in most of the cases is useless, do RR for
				# re-INVITEs alos, as some buggy clients do change route set
				# during the dialog.
				record_route();
			}

			if (check_route_param("nat=yes")) 
				setflag(NAT);

			# route it out to whatever destination was set by loose_route()
			# in $du (destination URI).
			route(relay);
		} else {
			if (is_method("SUBSCRIBE") && $rd == "_SEU_IP_AQUI_:5060") { # CUSTOMIZE ME
				# in-dialog subscribe requests
				route(handle_presence);
				exit;
			}
			if ( is_method("ACK") ) {
				if ( t_check_trans() ) {
					# non loose-route, but stateful ACK; must be an ACK after 
					# a 487 or e.g. 404 from upstream server
					t_relay();
					exit;
				} else {
					# ACK without matching transaction ->
					# ignore and discard
					exit;
				}
			}
			sl_send_reply("404","Not here");
		}
		exit;
	}

	# CANCEL processing
	if (is_method("CANCEL"))
	{
		if (t_check_trans())
			t_relay();
		exit;
	}

	t_check_trans();

	if ( !(is_method("REGISTER")  || is_from_gw() ) ) {
		
		if (is_from_local())
		{
			
			# authenticate if from local subscriber
			# authenticate all initial non-REGISTER request that pretend to be
			# generated by local subscriber (domain from FROM URI is local)
			if (!proxy_authorize("", "subscriber")) {
				proxy_challenge("", "0");
				exit;
			}
			if (!db_check_from()) {
				sl_send_reply("403","Forbidden auth ID");
				exit;
			}
		
			consume_credentials();
			# caller authenticated
			
		} else {
			# if caller is not local, then called number must be local
			
			if (!is_uri_host_local()) {
				send_reply("403","Rely forbidden");
				exit;
			}
		}

	}

	# preloaded route checking
	if (loose_route()) {
		xlog("L_ERR",
		"Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");
		if (!is_method("ACK"))
			sl_send_reply("403","Preload Route denied");
		exit;
	}

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route();

	# account only INVITEs
	if (is_method("INVITE")) {
		
		if(has_totag()){
			engage_rtp_proxy();
		}
		# create dialog with timeout
		if ( !create_dialog("B") ) {
			send_reply("500","Internal Server Error");
			exit;
		}
		
		setflag(ACC_DO); # do accounting
	}

	
	if (!is_uri_host_local()) {
		append_hf("P-hint: outbound\r\n"); 
		
		route(relay);
	}

	# requests for my domain
	
	if( is_method("PUBLISH|SUBSCRIBE"))
			route(handle_presence);

	if (is_method("REGISTER"))
	{
		
		# authenticate the REGISTER requests
		if (!www_authorize("", "subscriber"))
		{
			www_challenge("", "0");
			exit;
		}
		
		if (!db_check_to()) 
		{
			sl_send_reply("403","Forbidden auth ID");
			exit;
		}

		if (   0 ) setflag(TCP_PERSISTENT);

		if (!save("location"))
			sl_reply_error();

		exit;
	}

	if ($rU==NULL) {
		# request with no Username in RURI
		sl_send_reply("484","Address Incomplete");
		exit;
	}

	
	# apply DB based aliases
	alias_db_lookup("dbaliases");

	
	# apply transformations from dialplan table
	#xlog("Verificando se temos rota para este destino $ru/$ru");
	#dp_translate("0","$rU/$rU");
	

	# definimos aqui pstn como padrao 0
	$avp(pstn)=0;
	xlog("Verificando $rU para do_routing");	
	
	# Ligacao local (achar uma forma para pegar o ddd na base do usuario)
	# routeid 0 para gateway de ligacoes locais (acrescentar 5511)
	if ($rU=~"^0[0-9]{8}$") {
		$avp(pstn)=1;
		$avp(routeid)=0;

	# ligacao ddd (0 + ddd + numero) 
	# routeid 1 para gateway para ddd (acrescenta 55)
	}else if( $rU =~ "^0[0-9]{2}[0-9]{8}$" ){
		$avp(pstn)=1;
		$avp(routeid)=1;


	# Ligacao ldn (00 + numero)
	# routeid 2 para remover o 00 e enviar a chamada diretamente
	}else if( $rU =~ "^00[0-9]+$" ){
		$avp(pstn)=1;
		$avp(routeid)=2;

	}

	# pstn esta definido e vamos rotear
	if($avp(pstn) == 1){
		xlog("Regra pre do_routing");
		
		# utilizamos o routeid que definimos para achar o gateway
		# correto para este perfil de chamada
		if (!do_routing("$avp(routeid)")) {
			send_reply("500","No PSTN Route found");
			exit;
		}
		
		route(relay);
		exit;
	}
	 

	# do lookup with method filtering
	if (!lookup("location","m")) {
		if (!db_does_uri_exist()) {
			send_reply("420","Bad Extension");
			exit;
		}
		
		t_newtran();
		t_reply("404", "Not Found");
		exit;
	} 

	# esta entrada fez a magica do sdp
	rtpproxy_offer();
	if (isbflagset(NAT)) setflag(NAT);

	# when routing via usrloc, log the missed calls also
	setflag(ACC_MISSED);
	route(relay);
}


route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		
		if (isflagset(NAT)) {
			rtpproxy_offer();
		}

		t_on_branch("per_branch_ops");
		t_on_reply("handle_nat");
		t_on_failure("missed_call");
	}

	if (isflagset(NAT)) {
		add_rr_param(";nat=yes");
		}

	if (!t_relay()) {
		send_reply("500","Internal Error");
	};
	exit;
}


# Presence route
route[handle_presence]
{
	if (!t_newtran())
	{
		sl_reply_error();
		exit;
	}

	if(is_method("PUBLISH"))
	{
		handle_publish();
	}
	else
	if( is_method("SUBSCRIBE"))
	{
		handle_subscribe();
	}

	exit;
}


branch_route[per_branch_ops] {
	xlog("new branch at $ru\n");
}


onreply_route[handle_nat] {
        #fix_nated_sdp("3");
	rtpproxy_answer();

	if (nat_uac_test("1"))
		fix_nated_contact();
	if ( isflagset(NAT) )
		rtpproxy_answer("ro");
	xlog("incoming reply\n");
}


failure_route[missed_call] {
	if (t_was_cancelled()) {
		exit;
	}

	# uncomment the following lines if you want to block client 
	# redirect based on 3xx replies.
	##if (t_check_status("3[0-9][0-9]")) {
	##t_reply("404","Not found");
	##	exit;
	##}

	
}



local_route {
	if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
		
		acc_db_request("200 Dialog Timeout", "acc");
		
	}
}


Agora precisamos criar usuários, alias, e rota de saida.

Criando um usuario

opensipsctl add NOME_DO_USUARIO SENHA_DO_USUARIO
opensipsctl add USUARIO2 SENHA_USUARIO2

Criando um alias

O alias normalmente é utilizado para um número referente a este usuário, seja um número de telefone ou um ramal.

opensipsctl alias_db add 05551234 NOME_DO_USUARIO

Neste caso, se um usuário ou uma chamada de fora vier para 05551234 ela será direcionada para o usuário especificado

Criando os troncos

Agora precisamos criar os troncos de saida, você deve inserir no banco de dados, veja abaixo um exemplo.

mysql> select * from dr_gateways;
+----+------+------+-------------+-------+------------+-------+------------+-------------+
| id | gwid | type | address     | strip | pri_prefix | attrs | probe_mode | description |
+----+------+------+-------------+-------+------------+-------+------------+-------------+
|  1 | 1    |    2 | IP_DO_TRONCO|     0 | 5511       |       |          2 | LOCAL SP    |
|  3 | 2    |    2 | IP_DO_TRONCO|     1 | 55         |       |          0 | LDN         |
|  4 | 3    |    2 | IP_DO_TRONCO|     2 |            |       |          0 | LDI         |
+----+------+------+-------------+-------+------------+-------+------------+-------------+

Veja que temos duas opções especificas ali, que é o strip, o strip é para remover um digito do numero recebido, a opção pri_prefix é para adicionar um prefixo antes da discagem

Com esta opção agora precisamos criar a regra que vai entrar para um tronco o outro, abaixo segue as entradas do banco de dados.

mysql> select * from dr_rules;
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+
| ruleid | groupid | prefix | timerec | priority | routeid | gwlist | attrs | description |
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+
|      4 | 0       |        |         |        0 | 1       | 1      |       | Local       |
|      5 | 1       |        |         |        0 | 1       | 2      |       | LDN         |
|      6 | 2       |        |         |        0 | 1       | 3      |       | LDI         |
+--------+---------+--------+---------+----------+---------+--------+-------+-------------+

Neste caso o detalhe está gwlist e no groupid, nós no nosso arquivo opensips.cfg nas regras para as ligações externas, definimos que ele buscará pelo grupo 0 para local, pelo grupo 1 para ldn , e pelo grupo 2 para internacional, e cada um destes grupos corresponde a um ip da tabela da anterior.


Com estes dados criados, podemos dar inicio nos testes.

Iniciando o OpenSIPS

Precisamos iniciar o OpenSIPS antes de mais nada, então para isso execute o comando abaixo.

/etc/init.d/opensips start

Se o sistema levantar, então é só dar sequencia, se não levantar, habilite as opções de debug do arquivo e então execute o comando abaixo para tnetar localizar o erro.

/etc/init.d/opensips debug


Fazendo chamadas

Para fazer as chamadas as instruções são:

LOCAL: 0 + 8 DIGITOS
LDN  : 0 + DDD + 8 DIGITOS
LDI  : 00 + NUMERO

Para chamar um usuário você pode discar o alias que voce atribuiu ou o proprio nome do usuário se for softphone

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21:16:46 (*.251.139.148)
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