<div dir="ltr">We use Kamailio as sip proxy, loadbalance and registration of SIP users. And asterisk for main business logic, outbound and inbound calls.<div>All calls go through some of Asterisk servers.</div><div>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.</div>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Aug 21, 2013 at 3:54 PM, Nishar M.H <span dir="ltr"><<a href="mailto:nisharmh85@gmail.com" target="_blank">nisharmh85@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><div><div><div><div><div>Hi All,<br><br></div>I have created a Two VMs for Asterisk & Kamailio.<br>
</div>Asterisk IP : 192.168.20.196<br></div>Kamailio IP : 192.168.20.208<br><br></div>I have installed both successfully. I have created trunk between two and successfully started kamailio.<br>
<br></div><div>What i am looking for :<br><br></div><div>1) Asterisk as media server and Kamailio as SIP Server.<br></div><div>2) Call from Asterisk to Kamailio and vice versa.<br></div><div>3) Enable Outbound call for Kamailio User through Asterisk Server.<br>
</div><div>4) Inbound call from outside through Asterisk to Kamailio Users.<br></div><div><br></div>The issues i am facing here is :<br><br></div>1) Unable to call Internally. Kamailio SIP to SIP. If i tried to call from 1000 to 1001, "No route to destination.<br>
<br><br><div>2) Unable to call from Asterisk to Kamailio and vice versa.<br><br></div><div><br></div><div><br></div><div><br><br></div><div>Here is the log below: <br><br>/var/log# tail /var/log/syslog<br>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)<br>
Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [../outbound/api.h:49]: Failed to import bind_ob<br>Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: rr [rr_mod.c:159]: outbound module not available<br>
Aug 21 09:37:36 kamailio-VirtualBox /usr/local/sbin/kamailio[17495]: INFO: usrloc [hslot.c:53]: locks array size 512<br>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<br>
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<br>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<br>
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<br>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)<br>
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<br>
<br><br></div><div>One more thing i have to ask is :<br><br></div><div>1) Is there any command to check Kamailio has a trunk with Asterisk PBX server ?<br>Like in Asterisk SIP show peers.<br></div><div>I have tried "kamctl ul show". But it shows kamailio user 1000 and 1001.<br>
<br><br></div><div>My kamailio.cfg file :<br><br><br>#!KAMAILIO<br>#<br># Kamailio (OpenSER) SIP Server v4.0 - default configuration script<br># - web: <a href="http://www.kamailio.org" target="_blank">http://www.kamailio.org</a><br>
# - git: <a href="http://sip-router.org" target="_blank">http://sip-router.org</a><br>#<br># Direct your questions about this file to: <<a href="mailto:sr-users@lists.sip-router.org" target="_blank">sr-users@lists.sip-router.org</a>><br>
#<br># Refer to the Core CookBook at <a href="http://www.kamailio.org/wiki/" target="_blank">http://www.kamailio.org/wiki/</a><br>
# for an explanation of possible statements, functions and parameters.<br>#<br># Several features can be enabled using '#!define WITH_FEATURE' directives:<br>#<br># *** To run in debug mode: <br># - define WITH_DEBUG<br>
#<br># *** To enable mysql: <br># - define WITH_MYSQL<br>#<br># *** To enable authentication execute:<br># - enable mysql<br># - define WITH_AUTH<br># - add users using 'kamctl'<br>#<br># *** To enable IP authentication execute:<br>
# - enable mysql<br># - enable authentication<br># - define WITH_IPAUTH<br># - add IP addresses with group id '1' to 'address' table<br>#<br># *** To enable persistent user location execute:<br>
# - enable mysql<br># - define WITH_USRLOCDB<br>#<br># *** To enable presence server execute:<br># - enable mysql<br># - define WITH_PRESENCE<br>#<br># *** To enable nat traversal execute:<br># - define WITH_NAT<br>
# - install RTPProxy: <a href="http://www.rtpproxy.org" target="_blank">http://www.rtpproxy.org</a><br># - start RTPProxy:<br># rtpproxy -l _your_public_ip_ -s udp:localhost:7722<br>#<br># *** To enable PSTN gateway routing execute:<br>
# - define WITH_PSTN<br># - set the value of pstn.gw_ip<br># - check route[PSTN] for regexp routing condition<br>#<br># *** To enable database aliases lookup execute:<br># - enable mysql<br># - define WITH_ALIASDB<br>
#<br># *** To enable speed dial lookup execute:<br># - enable mysql<br># - define WITH_SPEEDDIAL<br>#<br># *** To enable multi-domain support execute:<br># - enable mysql<br># - define WITH_MULTIDOMAIN<br>
#<br># *** To enable TLS support execute:<br># - adjust CFGDIR/tls.cfg as needed<br># - define WITH_TLS<br>#<br># *** To enable XMLRPC support execute:<br># - define WITH_XMLRPC<br># - adjust route[XMLRPC] for access policy<br>
#<br># *** To enable anti-flood detection execute:<br># - adjust pike and htable=>ipban settings as needed (default is<br># block if more than 16 requests in 2 seconds and ban for 300 seconds)<br># - define WITH_ANTIFLOOD<br>
#<br># *** To block 3XX redirect replies execute:<br># - define WITH_BLOCK3XX<br>#<br># *** To enable VoiceMail routing execute:<br># - define WITH_VOICEMAIL<br># - set the value of voicemail.srv_ip<br># - adjust the value of voicemail.srv_port<br>
#<br># *** To enhance accounting execute:<br># - enable mysql<br># - define WITH_ACCDB<br># - add following columns to database<br>#!ifdef ACCDB_COMMENT<br> ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';<br>
ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';<br> ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default '';<br> ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';<br>
ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';<br> ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';<br> ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT '';<br>
ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT '';<br> ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default '';<br> ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT '';<br>
ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT '';<br> ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT '';<br>#!endif<br><br>####### Include Local Config If Exists #########<br>
import_file "kamailio-local.cfg"<br><br>####### Defined Values #########<br><br># *** Value defines - IDs used later in config<br>#!ifdef WITH_MYSQL<br># - database URL - used to connect to database server by modules such<br>
# as: auth_db, acc, usrloc, a.s.o.<br>#!ifndef DBURL<br>#!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio"<br>#!endif<br>#!endif<br>#!ifdef WITH_MULTIDOMAIN<br># - the value for 'use_domain' parameters<br>
#!define MULTIDOMAIN 1<br>#!else<br>#!define MULTIDOMAIN 0<br>#!endif<br><br># - flags<br># FLT_ - per transaction (message) flags<br># FLB_ - per branch flags<br>#!define FLT_ACC 1<br>#!define FLT_ACCMISSED 2<br>#!define FLT_ACCFAILED 3<br>
#!define FLT_NATS 5<br><br>#!define FLB_NATB 6<br>#!define FLB_NATSIPPING 7<br><br>####### Global Parameters #########<br><br>### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR<br>#!ifdef WITH_DEBUG<br>debug=4<br>log_stderror=yes<br>
#!else<br>debug=2<br>log_stderror=no<br>#!endif<br><br>memdbg=5<br>memlog=5<br><br>log_facility=LOG_LOCAL0<br><br>fork=yes<br>children=4<br><br>/* uncomment the next line to disable TCP (default on) */<br>#disable_tcp=yes<br>
<br>/* uncomment the next line to disable the auto discovery of local aliases<br> based on reverse DNS on IPs (default on) */<br>#auto_aliases=no<br><br>/* add local domain aliases */<br>#alias="<a href="http://sip.mydomain.com" target="_blank">sip.mydomain.com</a>"<br>
<br>/* uncomment and configure the following line if you want Kamailio to <br> bind on a specific interface/port/proto (default bind on all available) */<br>#listen=udp:<a href="http://10.0.0.10:5060" target="_blank">10.0.0.10:5060</a><br>
<br>/* port to listen to<br> * - can be specified more than once if needed to listen on many ports */<br>port=5060<br><br>#!ifdef WITH_TLS<br>enable_tls=yes<br>#!endif<br><br># life time of TCP connection when there is no traffic<br>
# - a bit higher than registration expires to cope with UA behind NAT<br>tcp_connection_lifetime=3605<br><br>####### Custom Parameters #########<br><br># These parameters can be modified runtime via RPC interface<br># - see the documentation of 'cfg_rpc' module.<br>
#<br># Format: <a href="http://group.id" target="_blank">group.id</a> = value 'desc' description<br># Access: $sel(<a href="http://cfg_get.group.id" target="_blank">cfg_get.group.id</a>) or @<a href="http://cfg_get.group.id" target="_blank">cfg_get.group.id</a><br>
#<br><br>#!ifdef WITH_PSTN<br># PSTN GW Routing<br>#<br># - pstn.gw_ip: valid IP or hostname as string value, example:<br># pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"<br>#<br># - by default is empty to avoid misrouting<br>
pstn.gw_ip = "" desc "PSTN GW Address"<br>pstn.gw_port = "" desc "PSTN GW Port"<br>#!endif<br><br>#!ifdef WITH_VOICEMAIL<br># VoiceMail Routing on offline, busy or no answer<br>#<br>
# - by default Voicemail server IP is empty to avoid misrouting<br>voicemail.srv_ip = "" desc "VoiceMail IP Address"<br>voicemail.srv_port = "5060" desc "VoiceMail Port"<br>#!endif<br>
<br>####### Modules Section ########<br><br># set paths to location of modules (to sources or installation folders)<br>#!ifdef WITH_SRCPATH<br>mpath="modules_k:modules"<br>#!else<br>mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/"<br>
#!endif<br><br>#!ifdef WITH_MYSQL<br>loadmodule "db_mysql.so"<br>#!endif<br><br>loadmodule "mi_fifo.so"<br>loadmodule "kex.so"<br>loadmodule "corex.so"<br>loadmodule "tm.so"<br>
loadmodule "tmx.so"<br>loadmodule "sl.so"<br>loadmodule "rr.so"<br>loadmodule "pv.so"<br>loadmodule "maxfwd.so"<br>loadmodule "usrloc.so"<br>loadmodule "registrar.so"<br>
loadmodule "textops.so"<br>loadmodule "siputils.so"<br>loadmodule "xlog.so"<br>loadmodule "sanity.so"<br>loadmodule "ctl.so"<br>loadmodule "cfg_rpc.so"<br>loadmodule "mi_rpc.so"<br>
loadmodule "acc.so"<br><br>#!ifdef WITH_AUTH<br>loadmodule "auth.so"<br>loadmodule "auth_db.so"<br>#!ifdef WITH_IPAUTH<br>loadmodule "permissions.so"<br>#!endif<br>#!endif<br><br>#!ifdef WITH_ALIASDB<br>
loadmodule "alias_db.so"<br>#!endif<br><br>#!ifdef WITH_SPEEDDIAL<br>loadmodule "speeddial.so"<br>#!endif<br><br>#!ifdef WITH_MULTIDOMAIN<br>loadmodule "domain.so"<br>#!endif<br><br>#!ifdef WITH_PRESENCE<br>
loadmodule "presence.so"<br>loadmodule "presence_xml.so"<br>#!endif<br><br>#!ifdef WITH_NAT<br>loadmodule "nathelper.so"<br>loadmodule "rtpproxy.so"<br>#!endif<br><br>#!ifdef WITH_TLS<br>
loadmodule "tls.so"<br>#!endif<br><br>#!ifdef WITH_ANTIFLOOD<br>loadmodule "htable.so"<br>loadmodule "pike.so"<br>#!endif<br><br>#!ifdef WITH_XMLRPC<br>loadmodule "xmlrpc.so"<br>#!endif<br>
<br>#!ifdef WITH_DEBUG<br>loadmodule "debugger.so"<br>#!endif<br><br># ----------------- setting module-specific parameters ---------------<br><br><br># ----- mi_fifo params -----<br>modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")<br>
<br><br># ----- tm params -----<br># auto-discard branches from previous serial forking leg<br>modparam("tm", "failure_reply_mode", 3)<br># default retransmission timeout: 30sec<br>modparam("tm", "fr_timer", 30000)<br>
# default invite retransmission timeout after 1xx: 120sec<br>modparam("tm", "fr_inv_timer", 120000)<br><br><br># ----- rr params -----<br># add value to ;lr param to cope with most of the UAs<br>modparam("rr", "enable_full_lr", 1)<br>
# do not append from tag to the RR (no need for this script)<br>modparam("rr", "append_fromtag", 0)<br><br><br># ----- registrar params -----<br>modparam("registrar", "method_filtering", 1)<br>
/* uncomment the next line to disable parallel forking via location */<br># modparam("registrar", "append_branches", 0)<br>/* uncomment the next line not to allow more than 10 contacts per AOR */<br>#modparam("registrar", "max_contacts", 10)<br>
# max value for expires of registrations<br>modparam("registrar", "max_expires", 3600)<br># set it to 1 to enable GRUU<br>modparam("registrar", "gruu_enabled", 0)<br><br><br># ----- acc params -----<br>
/* what special events should be accounted ? */<br>modparam("acc", "early_media", 0)<br>modparam("acc", "report_ack", 0)<br>modparam("acc", "report_cancels", 0)<br>
/* by default ww do not adjust the direct of the sequential requests.<br> if you enable this parameter, be sure the enable "append_fromtag"<br> in "rr" module */<br>modparam("acc", "detect_direction", 0)<br>
/* account triggers (flags) */<br>modparam("acc", "log_flag", FLT_ACC)<br>modparam("acc", "log_missed_flag", FLT_ACCMISSED)<br>modparam("acc", "log_extra", <br>
"src_user=$fU;src_domain=$fd;src_ip=$si;"<br>
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")<br>modparam("acc", "failed_transaction_flag", FLT_ACCFAILED)<br>/* enhanced DB accounting */<br>#!ifdef WITH_ACCDB<br>modparam("acc", "db_flag", FLT_ACC)<br>
modparam("acc", "db_missed_flag", FLT_ACCMISSED)<br>modparam("acc", "db_url", DBURL)<br>modparam("acc", "db_extra",<br> "src_user=$fU;src_domain=$fd;src_ip=$si;"<br>
"dst_ouser=$tU;dst_user=$rU;dst_domain=$rd")<br>#!endif<br><br><br># ----- usrloc params -----<br>/* enable DB persistency for location entries */<br>#!ifdef WITH_USRLOCDB<br>modparam("usrloc", "db_url", DBURL)<br>
modparam("usrloc", "db_mode", 2)<br>modparam("usrloc", "use_domain", MULTIDOMAIN)<br>#!endif<br><br><br># ----- auth_db params -----<br>#!ifdef WITH_AUTH<br>modparam("auth_db", "db_url", DBURL)<br>
modparam("auth_db", "calculate_ha1", yes)<br>modparam("auth_db", "password_column", "password")<br>modparam("auth_db", "load_credentials", "")<br>
modparam("auth_db", "use_domain", MULTIDOMAIN)<br><br># ----- permissions params -----<br>#!ifdef WITH_IPAUTH<br>modparam("permissions", "db_url", DBURL)<br>modparam("permissions", "db_mode", 1)<br>
#!endif<br><br>#!endif<br><br><br># ----- alias_db params -----<br>#!ifdef WITH_ALIASDB<br>modparam("alias_db", "db_url", DBURL)<br>modparam("alias_db", "use_domain", MULTIDOMAIN)<br>
#!endif<br><br><br># ----- speeddial params -----<br>#!ifdef WITH_SPEEDDIAL<br>modparam("speeddial", "db_url", DBURL)<br>modparam("speeddial", "use_domain", MULTIDOMAIN)<br>#!endif<br>
<br><br># ----- domain params -----<br>#!ifdef WITH_MULTIDOMAIN<br>modparam("domain", "db_url", DBURL)<br># register callback to match myself condition with domains list<br>modparam("domain", "register_myself", 1)<br>
#!endif<br><br><br>#!ifdef WITH_PRESENCE<br># ----- presence params -----<br>modparam("presence", "db_url", DBURL)<br><br># ----- presence_xml params -----<br>modparam("presence_xml", "db_url", DBURL)<br>
modparam("presence_xml", "force_active", 1)<br>#!endif<br><br><br>#!ifdef WITH_NAT<br># ----- rtpproxy params -----<br>modparam("rtpproxy", "rtpproxy_sock", "udp:<a href="http://127.0.0.1:7722" target="_blank">127.0.0.1:7722</a>")<br>
<br># ----- nathelper params -----<br>modparam("nathelper", "natping_interval", 30)<br>modparam("nathelper", "ping_nated_only", 1)<br>modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)<br>
modparam("nathelper", "sipping_from", "<a href="mailto:sip%3Apinger@kamailio.org" target="_blank">sip:pinger@kamailio.org</a>")<br><br># params needed for NAT traversal in other modules<br>modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)")<br>
modparam("usrloc", "nat_bflag", FLB_NATB)<br>#!endif<br><br><br>#!ifdef WITH_TLS<br># ----- tls params -----<br>modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")<br>
#!endif<br><br>#!ifdef WITH_ANTIFLOOD<br># ----- pike params -----<br>modparam("pike", "sampling_time_unit", 2)<br>modparam("pike", "reqs_density_per_unit", 16)<br>modparam("pike", "remove_latency", 4)<br>
<br># ----- htable params -----<br># ip ban htable with autoexpire after 5 minutes<br>modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")<br>#!endif<br><br>#!ifdef WITH_XMLRPC<br>
# ----- xmlrpc params -----<br>modparam("xmlrpc", "route", "XMLRPC");<br>modparam("xmlrpc", "url_match", "^/RPC")<br>#!endif<br><br>#!ifdef WITH_DEBUG<br># ----- debugger params -----<br>
modparam("debugger", "cfgtrace", 1)<br>#!endif<br><br>####### Routing Logic ########<br><br><br># Main SIP request routing logic<br># - processing of any incoming SIP request starts with this route<br>
# - note: this is the same as route { ... }<br>request_route {<br><br> # per request initial checks<br> route(REQINIT);<br><br> # NAT detection<br> route(NATDETECT);<br><br> # CANCEL processing<br> if (is_method("CANCEL"))<br>
{<br> if (t_check_trans()) {<br> route(RELAY);<br> }<br> exit;<br> }<br><br> # handle requests within SIP dialogs<br> route(WITHINDLG);<br><br> ### only initial requests (no To tag)<br>
<br> t_check_trans();<br><br> # authentication<br> route(AUTH);<br><br> # record routing for dialog forming requests (in case they are routed)<br> # - remove preloaded route headers<br> remove_hf("Route");<br>
if (is_method("INVITE|SUBSCRIBE"))<br> record_route();<br><br> # account only INVITEs<br> <b> if (is_method("INVITE"))<br> {<br> # setflag(FLT_ACC); # do accounting<br> setflag(1); # do accouting<br>
<br> if (uri=~"<a href="http://sip:4000@192.168.20.196:5080" target="_blank">sip:4000@192.168.20.196:5080</a>")<br> {<br> route(2);<br> }<br> }</b><br><br> # dispatch requests to foreign domains<br>
route(SIPOUT);<br><br> ### requests for my local domains<br><br> # handle presence related requests<br> route(PRESENCE);<br><br> # handle registrations<br> route(REGISTRAR);<br><br> if ($rU==$null)<br>
{<br> # request with no Username in RURI<br> sl_send_reply("484","Address Incomplete");<br> exit;<br> }<br><br> # dispatch destinations to PSTN<br> route(PSTN);<br><br>
# user location service<br> route(LOCATION);<br>}<br><br><b>route[2] {<br><br> rewritehostport("<a href="http://192.168.20.196:5080" target="_blank">192.168.20.196:5080</a>"); # change the IP here with the IP of your Asterisk Server<br>
t_relay();<br> exit;<br>}</b><br><br><br><br><br>route[RELAY] {<br><br> # enable additional event routes for forwarded requests<br> # - serial forking, RTP relaying handling, a.s.o.<br> if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {<br>
if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH");<br> }<br> if (is_method("INVITE|SUBSCRIBE|UPDATE")) {<br> if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY");<br>
}<br> if (is_method("INVITE")) {<br> if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE");<br> }<br><br> if (!t_relay()) {<br> sl_reply_error();<br> }<br>
exit;<br>}<br><br># Per SIP request initial checks<br>route[REQINIT] {<br>#!ifdef WITH_ANTIFLOOD<br> # flood dection from same IP and traffic ban for a while<br> # be sure you exclude checking trusted peers, such as pstn gateways<br>
# - local host excluded (e.g., loop to self)<br> if(src_ip!=myself)<br> {<br> if($sht(ipban=>$si)!=$null)<br> {<br> # ip is already blocked<br> xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n");<br>
exit;<br> }<br> if (!pike_check_req())<br> {<br> xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n");<br> $sht(ipban=>$si) = 1;<br>
exit;<br> }<br> }<br>#!endif<br><br> if (!mf_process_maxfwd_header("10")) {<br> sl_send_reply("483","Too Many Hops");<br> exit;<br> }<br><br> if(!sanity_check("1511", "7"))<br>
{<br> xlog("Malformed SIP message from $si:$sp\n");<br> exit;<br> }<br>}<br><br># Handle requests within SIP dialogs<br>route[WITHINDLG] {<br> if (has_totag()) {<br> # sequential request withing a dialog should<br>
# take the path determined by record-routing<br> if (loose_route()) {<br> route(DLGURI);<br> if (is_method("BYE")) {<br> setflag(FLT_ACC); # do accounting ...<br>
setflag(FLT_ACCFAILED); # ... even if the transaction fails<br> }<br> else if ( is_method("ACK") ) {<br> # ACK is forwarded statelessy<br> route(NATMANAGE);<br>
}<br> else if ( is_method("NOTIFY") ) {<br> # Add Record-Route for in-dialog NOTIFY as per RFC 6665.<br> record_route();<br> }<br> route(RELAY);<br>
} else {<br> if (is_method("SUBSCRIBE") && uri == myself) {<br> # in-dialog subscribe requests<br> route(PRESENCE);<br> exit;<br> }<br>
if ( is_method("ACK") ) {<br> if ( t_check_trans() ) {<br> # no loose-route, but stateful ACK;<br> # must be an ACK after a 487<br> # or e.g. 404 from upstream server<br>
route(RELAY);<br> exit;<br> } else {<br> # ACK without matching transaction ... ignore and discard<br> exit;<br> }<br>
}<br> sl_send_reply("404","Not here");<br> }<br> exit;<br> }<br>}<br><br># Handle SIP registrations<br>route[REGISTRAR] {<br> if (is_method("REGISTER"))<br>
{<br> if(isflagset(FLT_NATS))<br> {<br> setbflag(FLB_NATB);<br> # uncomment next line to do SIP NAT pinging <br> ## setbflag(FLB_NATSIPPING);<br> }<br> if (!save("location"))<br>
sl_reply_error();<br><br> exit;<br> }<br>}<br><br># USER location service<br>route[LOCATION] {<br><br>#!ifdef WITH_SPEEDDIAL<br> # search for short dialing - 2-digit extension<br> if($rU=~"^[0-9][0-9]$")<br>
if(sd_lookup("speed_dial"))<br> route(SIPOUT);<br>#!endif<br><br>#!ifdef WITH_ALIASDB<br> # search in DB-based aliases<br> if(alias_db_lookup("dbaliases"))<br> route(SIPOUT);<br>
#!endif<br><br> $avp(oexten) = $rU;<br> if (!lookup("location")) {<br> $var(rc) = $rc;<br> route(TOVOICEMAIL);<br> t_newtran();<br> switch ($var(rc)) {<br> case -1:<br>
case -3:<br> send_reply("404", "Not Found");<br> exit;<br> case -2:<br> send_reply("405", "Method Not Allowed");<br>
exit;<br> }<br> }<br><br> # when routing via usrloc, log the missed calls also<br> if (is_method("INVITE"))<br> {<br> setflag(FLT_ACCMISSED);<br> }<br><br> route(RELAY);<br>
exit;<br>}<br><br># Presence server route<br>route[PRESENCE] {<br> if(!is_method("PUBLISH|SUBSCRIBE"))<br> return;<br><br>#!ifdef WITH_PRESENCE<br> if (!t_newtran())<br> {<br> sl_reply_error();<br>
exit;<br> };<br><br> if(is_method("PUBLISH"))<br> {<br> handle_publish();<br> t_release();<br> }<br> else<br> if( is_method("SUBSCRIBE"))<br> {<br> handle_subscribe();<br>
t_release();<br> }<br> exit;<br>#!endif<br> <br> # if presence enabled, this part will not be executed<br> if (is_method("PUBLISH") || $rU==$null)<br> {<br> sl_send_reply("404", "Not here");<br>
exit;<br> }<br> return;<br>}<br><br># Authentication route<br>route[AUTH] {<br>#!ifdef WITH_AUTH<br><br>#!ifdef WITH_IPAUTH<br> if((!is_method("REGISTER")) && allow_source_address())<br>
{<br> # source IP allowed<br> return;<br> }<br>#!endif<br><br> if (is_method("REGISTER") || from_uri==myself)<br> {<br> # authenticate requests<br> if (!auth_check("$fd", "subscriber", "1")) {<br>
auth_challenge("$fd", "0");<br> exit;<br> }<br> # user authenticated - remove auth header<br> if(!is_method("REGISTER|PUBLISH"))<br> consume_credentials();<br>
}<br> # if caller is not local subscriber, then check if it calls<br> # a local destination, otherwise deny, not an open relay here<br> if (from_uri!=myself && uri!=myself)<br> {<br> sl_send_reply("403","Not relaying");<br>
exit;<br> }<br><br>#!endif<br> return;<br>}<br><br># Caller NAT detection route<br>route[NATDETECT] {<br>#!ifdef WITH_NAT<br> force_rport();<br> if (nat_uac_test("19")) {<br> if (is_method("REGISTER")) {<br>
fix_nated_register();<br> } else {<br> add_contact_alias();<br> }<br> setflag(FLT_NATS);<br> }<br>#!endif<br> return;<br>}<br><br># RTPProxy control<br>route[NATMANAGE] {<br>
#!ifdef WITH_NAT<br> if (is_request()) {<br> if(has_totag()) {<br> if(check_route_param("nat=yes")) {<br> setbflag(FLB_NATB);<br> }<br> }<br> }<br> if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))<br>
return;<br><br> rtpproxy_manage();<br><br> if (is_request()) {<br> if (!has_totag()) {<br> add_rr_param(";nat=yes");<br> }<br> }<br> if (is_reply()) {<br> if(isbflagset(FLB_NATB)) {<br>
add_contact_alias();<br> }<br> }<br>#!endif<br> return;<br>}<br><br># URI update for dialog requests<br>route[DLGURI] {<br>#!ifdef WITH_NAT<br> if(!isdsturiset()) {<br> handle_ruri_alias();<br>
}<br>#!endif<br> return;<br>}<br><br># Routing to foreign domains<br>route[SIPOUT] {<br> if (!uri==myself)<br> {<br> append_hf("P-hint: outbound\r\n");<br> route(RELAY);<br> }<br>}<br>
<br># PSTN GW routing<br>route[PSTN] {<br>#!ifdef WITH_PSTN<br> # check if PSTN GW IP is defined<br> if (strempty($sel(cfg_get.pstn.gw_ip))) {<br> xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n");<br>
return;<br> }<br><br> # route to PSTN dialed numbers starting with '+' or '00'<br> # (international format)<br> # - update the condition to match your dialing rules for PSTN routing<br>
if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$"))<br> return;<br><br> # only local users allowed to call<br> if(from_uri!=myself) {<br> sl_send_reply("403", "Not Allowed");<br>
exit;<br>
}<br><br> if (strempty($sel(cfg_get.pstn.gw_port))) {<br> $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);<br> } else {<br> $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":"<br>
+ $sel(cfg_get.pstn.gw_port);<br> }<br><br> route(RELAY);<br> exit;<br>#!endif<br><br> return;<br>}<br><br># XMLRPC routing<br>#!ifdef WITH_XMLRPC<br>route[XMLRPC] {<br> # allow XMLRPC from localhost<br>
if ((method=="POST" || method=="GET")<br> && (src_ip==127.0.0.1)) {<br> # close connection only for xmlrpclib user agents (there is a bug in<br> # xmlrpclib: it waits for EOF before interpreting the response).<br>
if ($hdr(User-Agent) =~ "xmlrpclib")<br> set_reply_close();<br> set_reply_no_connect();<br> dispatch_rpc();<br> exit;<br> }<br> send_reply("403", "Forbidden");<br>
exit;<br>}<br>#!endif<br><br># route to voicemail server<br>route[TOVOICEMAIL] {<br>#!ifdef WITH_VOICEMAIL<br> if(!is_method("INVITE"))<br> return;<br><br> # check if VoiceMail server IP is defined<br>
if (strempty($sel(cfg_get.voicemail.srv_ip))) {<br> xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n");<br> return;<br> }<br> if($avp(oexten)==$null)<br> return;<br><br>
$ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip)<br> + ":" + $sel(cfg_get.voicemail.srv_port);<br> route(RELAY);<br> exit;<br>#!endif<br><br> return;<br>
}<br><br># manage outgoing branches<br>branch_route[MANAGE_BRANCH] {<br> xdbg("new branch [$T_branch_idx] to $ru\n");<br> route(NATMANAGE);<br>}<br><br># manage incoming replies<br>onreply_route[MANAGE_REPLY] {<br>
xdbg("incoming reply\n");<br> if(status=~"[12][0-9][0-9]")<br> route(NATMANAGE);<br>}<br><br># manage failure routing cases<br>failure_route[MANAGE_FAILURE] {<br> route(NATMANAGE);<br>
<br> if (t_is_canceled()) {<br> exit;<br> }<br><br>#!ifdef WITH_BLOCK3XX<br> # block call redirect based on 3xx replies.<br> if (t_check_status("3[0-9][0-9]")) {<br> t_reply("404","Not found");<br>
exit;<br> }<br>#!endif<br><br>#!ifdef WITH_VOICEMAIL<br> # serial forking<br> # - route to voicemail on busy or no answer (timeout)<br> if (t_check_status("486|408")) {<br> route(TOVOICEMAIL);<br>
exit;<br> }<br>#!endif<br>}<br><br></div><div>
<br></div><div><div><div><div><div><div><div><br>-- <br><div dir="ltr"><pre><br><code><span><code><span>Thanks & Regards,<br><br></span></code>--------------------------------------------------------------------------------------------<br>
<br><span style="font-family:arial narrow,sans-serif"><font><b>Nishar Hamsa<br><br></b></font></span><br><br><br><br><br></span></code><code><span><code><span>--------------------------------------------------------------------------------------------</span></code></span><br>
<br><br><span></span></code></pre></div>
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