-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
I updated this config to work with version 4. See below. I have calls working over TLS between jitsi sip clients registered to the same proxy of either one of two proxies I have built using this updated config. I can make calls to/from clients over TLS registered to the openrcs.com proxy from both of these proxies. I can't make calls between the two proxies configured with the config below. Each domain has commercial SSL certs. I have rtpproxy configured and working. I would be very grateful if somebody would check the config and see if I have made a mistake. Many thanks. I will post the final working config so it may be of help to others.
Regards, John
#!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS #!define WITH_MULTIDOMAIN # # Kamailio (OpenSER) SIP Server v4.0 - By Daniel-Constantin Mierla # http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour # updated for version 4.0 by John Cahill johnc<AT>aktivix.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 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 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 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 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 "mysql://kamailio:kamailiorw@localhost/kamailio" #!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=3 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: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
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" #!endif
####### Modules Section ########
# set paths to location of modules #!ifdef LOCAL_TEST_RUN mpath="modules_k:modules" #!else mpath="/usr/lib64/kamailio/modules_k/:/usr/lib64/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 "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_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
# ----------------- 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)
# ----- 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=$fd;dst_ouser=$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;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif
# ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif
# ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif
#!endif
# ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif
# ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching #modparam("domain", "db_mode", 1) JC commented # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif
#!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL)
# ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif
#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@somedomain.com")
# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif
#!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/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
####### Routing Logic ########
# Main SIP request routing logic # - processing of any incoming SIP request starts with this route route {
# per request initial checks route(REQINIT);
# NAT detection route(NAT);
# 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 }
# 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(RELAY); }
route[RELAY] { #!ifdef WITH_NAT if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (isflagset(FLT_NATS) || isbflagset(FLB_NATB)) { route(RTPPROXY); } #!endif
/* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("BRANCH_ONE"); t_on_reply("REPLY_ONE"); t_on_failure("FAIL_ONE"); }
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()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } 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 ... 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_ALIASDB # search in DB-based aliases alias_db_lookup("dbaliases"); #!endif
if (!lookup("location")) { switch ($rc) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } }
# when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } }
# 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 if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("$td", "subscriber")) { www_challenge("$td", "0"); exit; }
if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else {
#!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif
# authenticate if from local subscriber if (from_uri==myself) { if (!proxy_authorize("$fd", "subscriber")) { proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } }
consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; }
# Caller NAT detection route route[NAT] { #!ifdef WITH_NAT force_rport(); setflag(FLT_NATS); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
# RTPProxy control route[RTPPROXY] { #!ifdef WITH_NAT if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ rtpproxy_offer(); } if (!has_totag()) add_rr_param(";nat=yes"); #!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; }
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY); exit; #!endif
return; }
# XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif
# Sample branch router branch_route[BRANCH_ONE] { xdbg("new branch at $ru\n"); }
# Sample onreply route onreply_route[REPLY_ONE] { xdbg("incoming reply\n"); #!ifdef WITH_NAT if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB)) && status=~"(183)|(2[0-9][0-9])") { rtpproxy_answer(); } if (isbflagset("6")) { fix_nated_contact(); } #!endif }
# Sample failure route failure_route[FAIL_ONE] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif
if (t_is_canceled()) { 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; ##}
# uncomment the following lines if you want to redirect the failed # calls to a different new destination ##if (t_check_status("486|408")) { ## sethostport("192.168.2.100:5060"); ## append_branch(); ## # do not set the missed call flag again ## t_relay(); ##} }
Hello,
On 5/20/13 12:47 PM, johnc wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
I updated this config to work with version 4. See below. I have calls working over TLS between jitsi sip clients registered to the same proxy of either one of two proxies I have built using this updated config. I can make calls to/from clients over TLS registered to the openrcs.com proxy from both of these proxies. I can't make calls between the two proxies configured with the config below. Each domain has commercial SSL certs. I have rtpproxy configured and working. I would be very grateful if somebody would check the config and see if I have made a mistake. Many thanks. I will post the final working config so it may be of help to others.
it is rather impossible to test configs from other people and not easy to review large configs. So I recommend you run with debug=3, see what is printed when you try a call that fails. If you cannot figure out from there what's the solution, then send the debug messages here.
Cheers, Daniel
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
Here is a The kamailio log from the sip proxy (atoll.dmcip.com) of the caller. There is nothing in the called server's (unspeak.im) logs from the call.
johnc@atoll.dmcip.com ----Call---> gnuday@unspeak.im
The call gets as far as "SIP 100 trying".
Both servers have only port 5061 TCP open on the firewall to only allow TLS.
Any help would be greatly appreciated.
Many thanks.
Regards, John
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
I've worked out what's going wrong and I fear there is a mistake in your config at:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
inter-domain calls are going in plain text over UDP. I can see the invites at the reciving end doing a packet capture using tcpdump. My firewall blocks these thus explaining why nothing was seen in the kamailio log at that end.
tcpdump -vvv -i eth0 src 46.43.48.253 and udp and port 5060 >debug.txt
How do I make all inter-domain traffic only TLS?
I've ensured that there are just valid TLS SRV records at both ends:
host -t SRV _sips._tcp.unspeak.im _sips._tcp.unspeak.im has SRV record 0 0 5061 unspeak.im. host -t SRV _sips._tcp.atoll.dmcip.com _sips._tcp.atoll.dmcip.com has SRV record 0 10 5061 atoll.dmcip.com.
Many thanks for all your help so far.
Regards, John
- -------- Original Message -------- Subject: Re: [SR-Users] http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour Date: Tue, 21 May 2013 05:28:31 +0100 From: johnc johnc@aktivix.org Reply-To: Kamailio (SER) - Users Mailing List sr-users@lists.sip-router.org To: miconda@gmail.com, "Kamailio (SER) - Users Mailing List" sr-users@lists.sip-router.org
- -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
Here is a The kamailio log from the sip proxy (atoll.dmcip.com) of the caller. There is nothing in the called server's (unspeak.im) logs from the call.
johnc@atoll.dmcip.com ----Call---> gnuday@unspeak.im
The call gets as far as "SIP 100 trying".
Both servers have only port 5061 TCP open on the firewall to only allow TLS.
Any help would be greatly appreciated.
Many thanks.
Regards, John
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
I've got my Kamailio script working, hurray! I had created SRV and NAPTR records for the proxies but was missing a critical line in the script:
dns_try_naptr=yes
This enables Kamailio to query the NAPTR and SRV records and then decide on the appropriate traffic to talk to the other proxy for non local calls.
http://kamailio.net/dokuwiki/doku.php/core-cookbook:3.0.x#dns_try_naptr
This was tough for a Kamailio noob like myself :-) I've put the whole thing on my wiki in case anybody else would like to build a federated SIP system. https://www.johncahill.net/wiki/index.php/Skype_like_conferencing_System
Thanks again for the help Daniel. I will update your wiki also, (which I shamelessly ripped off), if you give me write access.
Cheers, John
On 21/05/13 13:15, johnc wrote:
Hi,
I've worked out what's going wrong and I fear there is a mistake in
your config at:
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
inter-domain calls are going in plain text over UDP. I can see the
invites at the reciving end doing a packet capture using tcpdump. My firewall blocks these thus explaining why nothing was seen in the kamailio log at that end.
tcpdump -vvv -i eth0 src 46.43.48.253 and udp and port 5060 >debug.txt
How do I make all inter-domain traffic only TLS?
I've ensured that there are just valid TLS SRV records at both ends:
host -t SRV _sips._tcp.unspeak.im _sips._tcp.unspeak.im has SRV record 0 0 5061 unspeak.im. host -t SRV _sips._tcp.atoll.dmcip.com _sips._tcp.atoll.dmcip.com has SRV record 0 10 5061 atoll.dmcip.com.
Many thanks for all your help so far.
Regards, John
-------- Original Message -------- Subject: Re: [SR-Users]
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
Date: Tue, 21 May 2013 05:28:31 +0100 From: johnc johnc@aktivix.org Reply-To: Kamailio (SER) - Users Mailing List
To: miconda@gmail.com, "Kamailio (SER) - Users Mailing List"
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
Here is a The kamailio log from the sip proxy (atoll.dmcip.com) of the caller. There is nothing in the called server's (unspeak.im) logs from the call.
johnc@atoll.dmcip.com ----Call---> gnuday@unspeak.im
The call gets as far as "SIP 100 trying".
Both servers have only port 5061 TCP open on the firewall to only allow TLS.
Any help would be greatly appreciated.
Many thanks.
Regards, John -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
iQIcBAEBAgAGBQJRmvfnAAoJELy1jPQ1KER7O8EP/RfxYhBP4zhP6ioZq/piMIQX 6zCEX+dC087MHtST3lOqx1oQf5ERHbR8vRpbuZyoflnKHspAijwEvuQwWuPfGXlR RQlMowAMcKuEZ6RL4FS+Tm2VlFz+HFxH4KxJgOi721DGdzUrT4eKUf4hD3vsRlj/ QY7fLWiNj5TgHFu4yRu/OSVQ5DYr+PFrV8T7GtwgbCCq2Wzu1TdtuLTKJeUlHQJ/ v9uQxLJyFTUsjGWdWj1ynEBJ1CoYMwWj85eorjsylW/PQ49SJupNd1w4CkagCpAO 78h+qgcm2znxJajiA/sqD+31qzGCx2muSEVfFApg5JsLanr62xQuhAGYxLhDMR6u yda9ItUlFWrzrAy+TGf9NZyBN2l+pjAU2O4jWuEJRZ0znf9AshwS/zvRdbkGecqa E26eMSnfWks+JvE5YjIShyaKxXEcrNYm9RdrUQbKqzHwM0zS+/V42iZp9pZA73+T pWdwWasi0P5H1bBhLkdIO6mxidsahMbkUjeOeAzP7JVKszvjWow4ks/Pd/b+Rk6m /3A2bAyGn3SkPo4PveXBDGGdqA5sEMivm4WOBAGkXN2dIDtXCZ2bVGv7ucsm8qzC gAJ80cdiH2n3x8WFp9xlRBmcqG89b8JUW8I2KDjSCkEnltUByA7MC3YXQ+MdrlrI lBnxCsMyckAXGJnZniD+ =Zcfa -----END PGP SIGNATURE-----
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
Hello,
On 5/21/13 8:47 PM, johnc wrote:
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1
Hi,
I've got my Kamailio script working, hurray! I had created SRV and NAPTR records for the proxies but was missing a critical line in the script:
dns_try_naptr=yes
This enables Kamailio to query the NAPTR and SRV records and then decide on the appropriate traffic to talk to the other proxy for non local calls.
http://kamailio.net/dokuwiki/doku.php/core-cookbook:3.0.x#dns_try_naptr
This was tough for a Kamailio noob like myself :-) I've put the whole thing on my wiki in case anybody else would like to build a federated SIP system. https://www.johncahill.net/wiki/index.php/Skype_like_conferencing_System
Thanks again for the help Daniel. I will update your wiki also, (which I shamelessly ripped off), if you give me write access.
Thanks for sharing the result. I am planning an update of my tutorial as well, with couple of other adjustments.
Cheers, Daniel