We use Kamailio as sip proxy, loadbalance and registration of SIP users. And asterisk for main business logic, outbound and inbound calls. All calls go through some of Asterisk servers. In kamailio.conf - I do not want authorization from calls coming from Asterisk and I forward calls to local (for kamailio) SIP users. All calls to Kamailio are sent to some of Asterisk servers, and in sip.conf - all calls from Kamailio are authorized.

On Wed, Aug 21, 2013 at 3:54 PM, Nishar M.H nisharmh85@gmail.com wrote:

Hi All,

I have created a Two VMs for Asterisk & Kamailio. Asterisk IP : 192.168.20.196 Kamailio IP : 192.168.20.208

I have installed both successfully. I have created trunk between two and successfully started kamailio.

What i am looking for :

1. Asterisk as media server and Kamailio as SIP Server.
2. Call from Asterisk to Kamailio and vice versa.
3. Enable Outbound call for Kamailio User through Asterisk Server.
4. Inbound call from outside through Asterisk to Kamailio Users.

The issues i am facing here is :

1. Unable to call Internally. Kamailio SIP to SIP. If i tried to call from

1000 to 1001, "No route to destination.

2. Unable to call from Asterisk to Kamailio and vice versa.

Here is the log below:

/var/log# tail /var/log/syslog Aug 21 09:37:36 kamailio-VirtualBox kamailio: INFO: <core> [tcp_main.c:4846]: init_tcp: using epoll_lt as the io watch method (auto detected) Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [rr_mod.c:159]: outbound module not available Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: usrloc [hslot.c:53]: locks array size 512 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17507]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Aug 21 09:39:59 kamailio-VirtualBox /usr/local/sbin/kamailio[17504]: NOTICE: acc [acc.c:275]: ACC: call missed: timestamp=1377063599;method=INVITE;from_tag=ea6d986d;to_tag=;call_id=ZDZhMzkxN2RjZjhhYmJhMmJiNTM0OTRlYzA3NzJkMTA.;code=408;reason=Request Timeout;src_user=1000;src_domain=192.168.20.208;src_ip=192.168.20.186;dst_ouser=1000;dst_user=1000;dst_domain=192.168.20.186

One more thing i have to ask is :

1. Is there any command to check Kamailio has a trunk with Asterisk PBX

server ? Like in Asterisk SIP show peers. I have tried "kamctl ul show". But it shows kamailio user 1000 and 1001.

My kamailio.cfg file :

#!KAMAILIO # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users@lists.sip-router.org

...

# # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif

####### Include Local Config If Exists ######### import_file "kamailio-local.cfg"

####### Defined Values #########

# *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif

# - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5

#!define FLB_NATB 6 #!define FLB_NATSIPPING 7

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

### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif

memdbg=5 memlog=5

log_facility=LOG_LOCAL0

fork=yes children=4

/* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes

/* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no

/* add local domain aliases */ #alias="sip.mydomain.com"

/* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060

/* port to listen to

• • can be specified more than once if needed to listen on many ports */

port=5060

#!ifdef WITH_TLS enable_tls=yes #!endif

# life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605

####### Custom Parameters #########

# These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id #

#!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif

#!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif

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

# set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif

#!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif

loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so"

#!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif

#!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif

#!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif

#!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif

#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif

#!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif

#!ifdef WITH_TLS loadmodule "tls.so" #!endif

#!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif

#!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif

#!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif

# ----------------- setting module-specific parameters ---------------

# ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")

# ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000)

# ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0)

# ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0)

# ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww 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) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif

# ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif

# ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN)

# ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif

#!endif

# ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif

# ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif

# ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif

#!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL)

# ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif

#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")

# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org")

# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif

#!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif

#!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4)

# ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif

#!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif

#!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif

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

# Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route {

# per request initial checks
route(REQINIT);

# NAT detection
route(NATDETECT);

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

# handle requests within SIP dialogs
route(WITHINDLG);

### only initial requests (no To tag)

t_check_trans();

# authentication
route(AUTH);

# record routing for dialog forming requests (in case they are routed)
# - remove preloaded route headers
remove_hf("Route");
if (is_method("INVITE|SUBSCRIBE"))
    record_route();

# account only INVITEs

• if (is_method("INVITE"))

{
#    setflag(FLT_ACC); # do accounting
    setflag(1); # do accouting

    if (uri=~"sip:4000@192.168.20.196:5080")
    {
        route(2);
    }
}*

# dispatch requests to foreign domains
route(SIPOUT);

### requests for my local domains

# handle presence related requests
route(PRESENCE);

# handle registrations
route(REGISTRAR);

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

# dispatch destinations to PSTN
route(PSTN);

# user location service
route(LOCATION);

}

*route[2] {

    rewritehostport("192.168.20.196:5080"); # change the IP here with

the IP of your Asterisk Server t_relay(); exit; }*

route[RELAY] {

# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
    if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
    if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");
}
if (is_method("INVITE")) {
    if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");
}

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

}

# Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif

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

if(!sanity_check("1511", "7"))
{
    xlog("Malformed SIP message from $si:$sp\n");
    exit;
}

}

# Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } }

# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error();

    exit;
}

}

# USER location service route[LOCATION] {

#!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif

#!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif

$avp(oexten) = $rU;
if (!lookup("location")) {
    $var(rc) = $rc;
    route(TOVOICEMAIL);
    t_newtran();
    switch ($var(rc)) {
        case -1:
        case -3:
            send_reply("404", "Not Found");
            exit;
        case -2:
            send_reply("405", "Method Not Allowed");
            exit;
    }
}

# when routing via usrloc, log the missed calls also
if (is_method("INVITE"))
{
    setflag(FLT_ACCMISSED);
}

route(RELAY);
exit;

}

# Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return;

#!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; };

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

#!endif

# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
    sl_send_reply("404", "Not here");
    exit;
}
return;

}

# Authentication route route[AUTH] { #!ifdef WITH_AUTH

#!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif

if (is_method("REGISTER") || from_uri==myself)
{
    # authenticate requests
    if (!auth_check("$fd", "subscriber", "1")) {
        auth_challenge("$fd", "0");
        exit;
    }
    # user authenticated - remove auth header
    if(!is_method("REGISTER|PUBLISH"))
        consume_credentials();
}
# if caller is not local subscriber, then check if it calls
# a local destination, otherwise deny, not an open relay here
if (from_uri!=myself && uri!=myself)
{
    sl_send_reply("403","Not relaying");
    exit;
}

#!endif return; }

# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; }

# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;

rtpproxy_manage();

if (is_request()) {
    if (!has_totag()) {
        add_rr_param(";nat=yes");
    }
}
if (is_reply()) {
    if(isbflagset(FLB_NATB)) {
        add_contact_alias();
    }
}

#!endif return; }

# URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; }

# Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } }

# PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; }

# route to PSTN dialed numbers starting with '+' or '00'
#     (international format)
# - update the condition to match your dialing rules for PSTN routing
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))
    return;

# only local users allowed to call
if(from_uri!=myself) {
    sl_send_reply("403", "Not Allowed");
    exit;
}

if (strempty($sel(cfg_get.pstn.gw_port))) {
    $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
} else {
    $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"
                + $sel(cfg_get.pstn.gw_port);
}

route(RELAY);
exit;

#!endif

return;

}

# XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif

# route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return;

# check if VoiceMail server IP is defined
if (strempty($sel(cfg_get.voicemail.srv_ip))) {
    xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");
    return;
}
if($avp(oexten)==$null)
    return;

$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)
            + ":" + $sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;

#!endif

return;

}

# manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }

# manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); }

# manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE);

if (t_is_canceled()) {
    exit;
}

#!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif

#!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif }

--

Thanks & Regards,

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*Nishar Hamsa

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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users