Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio
The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private network.
The only problem I got is that called client get the call as it was originated by one of the media server (the call contact is some like ext@media.server.ip), which is wrong in my opinion.
Here it is the configuration, can you help me dealing with this?
Thank you
#!KAMAILIO
#!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT
#!define WITH_DISPATCHER
# # 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/dokuwiki/doku.php # 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
####### 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. #!define DBURL “database stuff" #!define DBASTURL "database stuff" #!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 #########
#!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=yes #!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:172.20.20.29:5060 listen=tcp:172.20.20.29:5060
listen=udp:PUBLIC IP:5060 listen=tcp:PUBLIC IP:5060
mhomed=1
/* 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 #
asterisk.bindip = "172.20.20.80" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "172.20.20.29" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port"
####### Modules Section ########
# set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif
#!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif
loadmodule "mi_fifo.so" loadmodule "kex.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
#!ifdef WITH_DISPATCHER loadmodule "dispatcher.so" loadmodule "ipops.so" loadmodule "sqlops.so" #!endif
loadmodule "uac.so"
# ----------------- 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", 1)
# ----- 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)
# ----- 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=$rd") #!endif
# ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif
# ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "")
modparam("auth_db", "user_column", "name") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0)
# ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif
#!endif
# ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif
# ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif
# ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif
#!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL)
# ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif
#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", “sip:pinger@PUBLIC IP")
# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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
#!ifdef WITH_DISPATCHER # ------- Load-balancer params ------ modparam("dispatcher", "db_url”,”MYSQL STUFF") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "setid_col", "setid") modparam("dispatcher", "destination_col", "destination") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dispatcher", "ds_ping_from", "sip:kamailio-sbc@172.20.20.29") modparam("dispatcher", "ds_ping_interval",15) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;class=3")
modparam("sqlops","sqlcon",”ca=>OTHER MYSQL STUFF") #!endif
####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route {
# per request initial checks route(REQINIT);
# NAT detection route(NATDETECT);
# handle requests within SIP dialogs route(WITHINDLG);
### only initial requests (no To tag)
# CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; }
t_check_trans();
# authentication route(AUTH);
# record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route();
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting }
# handle presence related requests route(PRESENCE);
# handle registrations route(REGISTRAR);
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
# user location service route(LOCATION);
route(RELAY); }
# Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } 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 t_relay(); exit; } else { # ACK without matching transaction ... nore and discard exit; } } sl_send_reply("404","Not here"); } exit; } }
# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
# Authentication route route[AUTH] {
#if(route(FROMASTERISK)) # return;
# 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; }
#!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("$fd", "sipusers", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; }
# 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; }
# 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();
route(REGFWD);
exit; } }
# USER location service route[LOCATION] {
if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; }
$avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($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] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); }
if (!t_relay()) { sl_reply_error(); } exit; }
# 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;
route(RTPPROXY);
if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; }
# 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; } }
# RTPProxy control route[RTPPROXY] { if (is_method("INVITE")){ sql_query("ca", "select destination from dispatcher where destination like '%$dd%'","ra"); if($dbr(ra=>rows)>0){ $avp(duip)=$(du{s.select,-2,:}); if (is_ip_rfc1918("$avp(duip)")) { xlog("L_INFO", "Call is going to private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("cwei","172.20.20.29"); #rtpproxy_manage("rwei"); }
} else if(ds_is_from_list()){ if (is_ip_rfc1918("$si")) { xlog("L_INFO", " Call is coming from a private IPv4 Media Server Engage RTPProxy Now\n"); rtpproxy_manage("cwie”,”PUBLIC IP"); #rtpproxy_manage("rwie");
} }else if(!ds_is_from_list()){ rtpproxy_manage("rwei"); xlog("L_INFO", "NONE! RTP PROXY BOH\n"); } } }
# Send to Asterisk route[TOASTERISK] { ds_mark_dst("P"); if(!ds_select_dst("1", "4")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","[$fU@$si:$sp]{$rm} No destinations available for $rd \n"); exit; }
xlog("L_INFO","[$fU@$si:$sp]{$rm} From Outside World to Asterisk Box $du\n"); rtpproxy_manage("cawie");
route(RELAY); exit; }
# Test if coming from Asterisk route[FROMASTERISK] { if(ds_is_from_list()){ rtpproxy_manage("cawei"); xlog("L_INFO","[$fU@$si:$sp]{$rm} Call from Media-Server Cluster\n"); return 1; } return -1; }
route[REGFWD] {
xlog("L_INFO","regfwd "+"sip:" +"\n");
if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); }
-- Alex PRE s.r.l. Backbone Network Mission Control System Engineering&Architecture Task Force alex@presrl.net http://www.presrl.it +39 0971 471 430 +39 388 150 6886
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Hello,
understanding the config file is going to take some time, so it is unlikely many will have the spare time for it.
You have to provide more specific details, like the sip trace (ngrep on kamailio server sip ports) for the issue, presenting what are the parties involved in sending and receiving, then what is wrong and what should be there.
Then we can eventually discover what is the part of the config dealing with that and how can be fixed.
Cheers, Daniel
On 11/09/14 00:55, [PRE s.r.l.] - Alex wrote:
Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio
The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private network.
The only problem I got is that called client get the call as it was originated by one of the media server (the call contact is some like ext@media.server.ip mailto:ext@media.server.ip), which is wrong in my opinion.
Here it is the configuration, can you help me dealing with this?
Hello, the situation is:
client -> kamailio + rtproxy -> asterisk -> rtpproxy + kamailio -> other client
the idea is that rtpproxy has to proxy the whole rtp traffic between asterisk or other media proxy and the client.
the strange problem is that my client which is receiving the call (csipsimple or other one) views the call origination as it was originated by asterisk private ip (172.20.20.80) (the callid/contact is extension@172.20.20.80). the correct behaviour is in my opinion that the call has to appear as it was originated by kamailio (extension@151.xx.xx.xx)
i'm using the dispatcher module to route the call from kamailio to asterisk.
Thank you for your help
On 11 Sep 2014, at 12:46, Daniel-Constantin Mierla miconda@gmail.com wrote:
Hello,
understanding the config file is going to take some time, so it is unlikely many will have the spare time for it.
You have to provide more specific details, like the sip trace (ngrep on kamailio server sip ports) for the issue, presenting what are the parties involved in sending and receiving, then what is wrong and what should be there.
Then we can eventually discover what is the part of the config dealing with that and how can be fixed.
Cheers, Daniel
On 11/09/14 00:55, [PRE s.r.l.] - Alex wrote:
Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio
The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private network.
The only problem I got is that called client get the call as it was originated by one of the media server (the call contact is some like ext@media.server.ip), which is wrong in my opinion.
Here it is the configuration, can you help me dealing with this?
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 - http://www.asipto.com Sep 22-25, Berlin, Germany _______________________________________________ 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
-- Alex PRE s.r.l. Backbone Network Mission Control System Engineering&Architecture Task Force alex@presrl.net http://www.presrl.it +39 0971 471 430 +39 388 150 6886
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Hello,
On 12/09/14 14:49, [PRE s.r.l.] - Alex wrote:
Hello, the situation is:
client -> kamailio + rtproxy -> asterisk -> rtpproxy + kamailio -> other client
the idea is that rtpproxy has to proxy the whole rtp traffic between asterisk or other media proxy and the client.
the strange problem is that my client which is receiving the call (csipsimple or other one) views the call origination as it was originated by asterisk private ip (172.20.20.80) (the callid/contact is extension@172.20.20.80 mailto:extension@172.20.20.80). the correct behaviour is in my opinion that the call has to appear as it was originated by kamailio (extension@151.xx.xx.xx mailto:extension@151.xx.xx.xx)
i'm using the dispatcher module to route the call from kamailio to asterisk.
kamailio being a proxy it does not changes caller/callee identification. You have options in asterisk dialplan to set caller domain, and this is the best way to do it.
In kamailio, you can do updates to caller id using uac module, but again, it is better to do it in asterisk and let kamailio act as a proxy.
Cheers, Daniel
Thank you for your help
On 11 Sep 2014, at 12:46, Daniel-Constantin Mierla <miconda@gmail.com mailto:miconda@gmail.com> wrote:
Hello,
understanding the config file is going to take some time, so it is unlikely many will have the spare time for it.
You have to provide more specific details, like the sip trace (ngrep on kamailio server sip ports) for the issue, presenting what are the parties involved in sending and receiving, then what is wrong and what should be there.
Then we can eventually discover what is the part of the config dealing with that and how can be fixed.
Cheers, Daniel
On 11/09/14 00:55, [PRE s.r.l.] - Alex wrote:
Hi to everyone, I’m trying to implement a SBC for my network based on Kamailio
The base idea is putting Kamailio on a dual home machine (one public interface for clients and a private interface for media servers and database), using rtpproxy to handle the relaying from public to private network.
The only problem I got is that called client get the call as it was originated by one of the media server (the call contact is some like ext@media.server.ip mailto:ext@media.server.ip), which is wrong in my opinion.
Here it is the configuration, can you help me dealing with this?
-- Daniel-Constantin Mierla http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda Next Kamailio Advanced Trainings 2014 -http://www.asipto.com Sep 22-25, Berlin, Germany _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
-- Alex PRE s.r.l. Backbone Network Mission Control System Engineering&Architecture Task Force alex@presrl.net mailto:alex@presrl.net http://www.presrl.it +39 0971 471 430 +39 388 150 6886
Questo messaggio e i suoi allegati sono indirizzati esclusivamente alle persone indicate. La diffusione, copia o qualsiasi altra azione derivante dalla conoscenza di queste informazioni sono rigorosamente vietate. Qualora abbiate ricevuto questo documento per errore siete cortesemente pregati di darne immediata comunicazione al mittente e di provvedere alla sua distruzione, Grazie.
/This e-mail and any attachments//is //confidential and may contain privileged information intended for the addressee(s) only. Dissemination, copying, printing or use by anybody else is unauthorised. If you are not the intended recipient, please delete this message and any attachments and advise the sender by return e-mail, Thanks./
*Rispetta l'ambiente. Non stampare questa mail se non è necessario.*
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