June 2009 - sr-users - kamailio.org

[Kamailio-Users] Kamailio / OpenSer billing platform
by Meftah Tayeb 30 Jun '09

30 Jun '09

hello all, please any billing modules or billing application to use with Kamailio / OpenSer? i need a prepay / Postpay application sam to Asterisk to billing (A2Billing) thanks for your help __________ Information from ESET NOD32 Antivirus, version of virus signature database 4191 (20090626) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com

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[Kamailio-Users] new feature: fine access to xml documents
by Daniel-Constantin Mierla 29 Jun '09

29 Jun '09

Hello, several days ago I committed support for accessing components of xml documents using pseudo-variables based on XPath syntax. The new $xml(...) PV is implemented in the presence_xml module as it works with xml docs and links libxml2. Short description in the cookbook: https://sip-router.org/wiki/cookbooks/pseudo-variables/devel#presence_xml_p… As an example: - publish has the body: <?xml version='1.0' encoding='UTF-8'?> <presence xmlns='urn:ietf:params:xml:ns:pidf' xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model' xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid' xmlns:c='urn:ietf:params:xml:ns:pidf:cipid' entity='sip:daniel@asipto.com'> <tuple id='t8b484917'> <status><basic>open</basic></status> </tuple> <dm:person id='p7562ef60'><rpid:activities><rpid:unknown/></rpid:activities></dm:person> </presence> Getting the presence status: # this is needed because libxml2 cannot work with default no-prefix ns, so we alias the "p" prefix. modparam("presence_xml", "xml_ns", "p=urn:ietf:params:xml:ns:pidf") $xml(a=>doc) = $rb; $xml(a=>xpath:/p:presence/p:tuple/p:status/p:basic) Next is to update the presence modules to be able to work with the documents that have been updated in the config. Comments, opinions and improvements are welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com/

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[Serusers] ser avpops {Scanned}
by sszasz＠digicomm.ro 29 Jun '09

29 Jun '09

Hi, I'm using SER and OpenSIPS for tests and I have a problem with avpops. How can I translate the following code snipplet ( working in opensips) to work in SER? if(avp_db_load("$fu/username","$avp(s:CBlocal)")) { sl_send_reply("403", "Call is barred"); exit; }; Thanks in advance Szasz Szabolcs -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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[Kamailio-Users] is avp the best option?
by Juan Perez 29 Jun '09

29 Jun '09

hello I need to load into memory a list of ip address associated with an index: ex: index, IP 1, 192.168.1.1 2. 192.168.1.2 3, 192.168.1.3 ......... ......... n, 192.168.1.n then at some point in the config I will call the variable, passing the index as argument and retrieving the IP as result can I do it with the avp module? cheers jp

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Re: [Kamailio-Users] sipsak
by Saúl Ibarra 29 Jun '09

29 Jun '09

Well, I think you should just need to locate the user and relay the request... but first you need to figure out why you afecta replying 501 to a notify... El 29 de jun de 2009, 11:32 a.m., "Chandrakant Solanki" < solanki.chandrakant(a)gmail.com> escribió: How can I explicitly permitting notify request from kamailio config...??? On Mon, Jun 29, 2009 at 2:45 PM, Saúl Ibarra <saghul(a)gmail.com> wrote: > > Are you explicitly not ... -- Regards, Chandrakant Solanki

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[Kamailio-Users] sipsak
by Chandrakant Solanki 29 Jun '09

29 Jun '09

Hi I am using sipsak for following details... Below details are store in /tmp/mwi-immediate file NOTIFY sip:1212@mytest.com <sip%3A1212(a)mytest.com> SIP/2.0 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> Contact: <sip:1212@72.18.100.73 <sip%3A1212(a)72.18.100.73>> Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Event: message-summary Content-Type: application/simple-message-summary Content-Length: 39 Messages-Waiting: yes Voicemail: 2/0 But when I execute sipsak command it gives following bad request #sipsak shoot -a number4 -f /tmp/mwi-immediate -s sip:1212@mytest.com<sip%3A1212(a)mytest.com>-v SIP/2.0 501 Method Not Supported Here Via: SIP/2.0/UDP 127.0.0.1:34389 ;branch=z9hG4bK.3242d1c3;rport=34389;alias;received=172.18.100.73 From: <sip:asterisk@172.18.100.73 <sip%3Aasterisk(a)172.18.100.73>> To: <sip:1212@mytest.com <sip%3A1212(a)mytest.com>> ;tag=d17c868d3732913d731aeb2b10d8dba7.1b83 Call-ID: 023798(a)72.18.100.73 CSeq: 023798 NOTIFY Server: Kamailio (1.5.0-notls (i386/linux)) Content-Length: 0 Any idea... regarding the above problem.... -- Regards, Chandrakant Solanki

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[Kamailio-Users] Problem upgrading from 1.4.x to 1.5.1 notls
by Uriel Rozenbaum 29 Jun '09

29 Jun '09

Guys, I've successfully upgraded schema and binaries for Kamailio, but when I start it, I get: Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:db_check_table_version: invalid version 2 for table carrierfailureroute found, expected 3 (check table structure and table "version") Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:24:34 sbci1pri /usr/local/sbin/kamailio[2611]: ERROR:core:main: error while initializing modules If I try and set 3 to the version table (for carrierfailureroute) I get the following: Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:db_check_table_version: invalid version 3 for table carrierfailureroute found, expected 2 (check table structure and table "version") Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:carrierroute:carrierroute_db_init: during table version check. Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:init_mod: failed to initialize module carrierroute Jun 25 18:42:34 sbci1pri /usr/local/sbin/kamailio[2660]: ERROR:core:main: error while initializing modules Is there anything I have to do (appart from following the guidelines on http://kamailio.org/dokuwiki/doku.php/install:1.4.x-to-1.5.0)? Thanks! Uriel

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[Serusers] auto-attendant problem
by IL'dar AKHmetgaleev 28 Jun '09

28 Jun '09

Greetings. I have a problem with auto-attendant. It accepts call immediately without playing my greeting message. How this problem can be fixed? My sip is akhilman(a)iptel.org -- Ильдар Ахметгалеев aka AkhIL Сбт Июн 27 23:52:48 KRAST 2009 Sat Jun 27 15:52:48 UTC 2009 ---------------------------------- Visit my home page http://akhilman.blogspot.com/ jabber: akhil(a)jabber.ru ---------------------------------- Маг формирует намерение совершить то, что он намерен совершить, просто за счет того, что он намеревается это совершить. (К.Кастанеда) ---------------------------------- Linux artstation 2.6.29-gentoo-r5 #1 SMP Mon Jun 1 19:50:26 KRAST 2009 x86_64 AMD Phenom(tm) 9550 Quad-Core Processor AuthenticAMD GNU/Linux up 12 days, 14:35, 1 user, load average: 0.20, 0.35, 0.38

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[Kamailio-Users] No Audio with clients Behind NAT, audio is fine with clients using public IPs (Im using rtpproxy)
by rubenrojas - Trc.es 26 Jun '09

26 Jun '09

Hello everyone, this is my first post on this list, I have installed kamailio 1.5.1 and set up a vanilla default kamailio.cfg, then I have modified the cfg to activate mysql, domain, presence, nathelper and authentication with md5, everything works as supposed to, and the clients can register, send txt messages and talk to each other. The only problem is with the audio when the two clients are behind a NAT, the phones can make a call and it does ring too, but when you pick up there is no audio both ways. when the phones have a public IP everything goes fine, it also works when I use a Linksys PAP2T whith the options to "Insert VIA received", "Insert VIA rport", "Handle VIA received", "Handle VIA rport" and "NAT mapping enable" turned on, with the Qutecom softphone works too. This is happening with thomson phones (model ST 2022), and GrandStream Budge Tone 200, it happens no matter what options I set for NATting on the phones, I've even used stun with stunserver.org or the ekiga stunserver, the phones register and can make and recieve calls, but there is no audio when you pick up the call. With a kamctl ul show, you can see that the phones have registered the Contact with their local IPs and the Received have the public IPs and ports for the NAT The only difference with the working Linksys is that they register the Contact with the public IP. Here you can see two NATed phones on the proxy Domain:: location table=512 records=2 max_slot=1 AOR:: 20000004(a)212.4.107.250 Contact:: sip:20000004@192.168.254.110:5060;transport=udp;user=phone Q= Expires:: 1150 Callid:: 72ed03f6d2f390f9(a)192.168.254.110 Cseq:: 10003 User-agent:: Grandstream BT200 1.1.6.27 Received:: sip:212.4.97.115:35379 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 7807 AOR:: 20000000(a)212.4.107.250 Contact:: sip:20000000@192.168.254.101:5060;user=phone Q= Expires:: 2945 Callid:: 17fe-c0a80101-5-1(a)192.168.254.101 Cseq:: 6 User-agent:: THOMSON ST2022 hw2 fw3.56 00-18-F6-B5-7E-06 Received:: sip:212.4.97.115:55128 State:: CS_NEW Flags:: 0 Cflag:: 0 Socket:: udp:212.4.107.250:5060 Methods:: 4294967295 Im using rtpproxy and there is no log error that indicates that rttpproxy isn't working, in fact doing a SIP trace shows rtpproxy setting ports for the audio. I run rtpproxy with this command: rtpproxy -l 212.4.107.250 -s udp:localhost:7722 -F Any help would be greatly appreciated, I've been two weeks looking for a solution Im attaching my kamailio.cfg so you can take a look, at the end of the message Im gonna attache the SIP Trace of a call between two NATed phones (a Thomson and a GrandStream) in case anyone can help me decypher whats wrong here: this is my cfg file ************************************************************************************************** # # $Id: kamailio.cfg 5800 2009-04-20 11:01:49Z miconda $ # # Kamailio (OpenSER) SIP Server - basic configuration script # - web: http://www.kamailio.org # - svn: http://openser.svn.sourceforge.net/viewvc/openser/ # # Direct your questions about this file to: <users(a)lists.kamailio.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # There are comments showing how to enable different features in th econfig # file. Such commented code starts with #X# where X is a letter to identify # a feature. Delete entire #X# if you want to enable that feature. Next are # sed commands that help you enable such features. # # *** To enamble mysql execute: # sed -i 's/#m#//g' kamailio.cfg # # *** To enamble authentication execute: # - enable mysql # sed -i 's/#a#//g' kamailio.cfg # - add users using 'kamctl' # # *** To enamble persistent user location execute: # - enable mysql # sed -i 's/#u#//g' kamailio.cfg # # *** To enamble presence server execute: # - enable mysql # sed -i 's/#p#//g' kamailio.cfg # # *** To enamble nat traversal execute: # sed -i 's/#n#//g' kamailio.cfg # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enhance accounting execute: # - enable mysql # sed -i 's/#c#//g' kamailio.cfg # - add following columns to database # 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_call ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; # ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; # ####### Global Parameters ######### debug=3 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the following lines to enable debugging */ #debug=6 #fork=no #log_stderror=yes /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to enable the auto temporary blacklisting of not available destinations (default disabled) */ #disable_dns_blacklist=no /* uncomment the next line to enable IPv6 lookup after IPv4 dns lookup failures (default disabled) */ #dns_try_ipv6=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ #auto_aliases=no /* uncomment the following lines to enable TLS support (default off) */ #disable_tls = no #listen = tls:your_IP:5061 #tls_verify_server = 1 #tls_verify_client = 1 #tls_require_client_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/kamailio/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/kamailio/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/kamailio/tls/user/user-calist.pem" port=5060 /* 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:192.168.1.2:5060 ####### Modules Section ######## #set module path mpath="/usr/local/lib/kamailio/modules/" /* uncomment next line for MySQL DB support */ loadmodule "db_mysql.so" loadmodule "mi_fifo.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri_db.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "acc.so" /* uncomment next lines for MySQL based authentication support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "auth.so" loadmodule "auth_db.so" /* uncomment next line for aliases support NOTE: a DB (like db_mysql) module must be also loaded */ #loadmodule "alias_db.so" /* uncomment next line for multi-domain support NOTE: a DB (like db_mysql) module must be also loaded NOTE: be sure and enable multi-domain support in all used modules (see "multi-module params" section ) */ loadmodule "domain.so" /* uncomment the next two lines for presence server support NOTE: a DB (like db_mysql) module must be also loaded */ loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "presence_mwi.so"#manually added loadmodule "nathelper.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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) # ----- rr 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) # ----- uri_db params ----- /* by default we disable the DB support in the module as we do not need it in this configuration */ modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "") # ----- acc params ----- /* what sepcial events should be accounted ? */ modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* 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", "failed_transaction_flag", 3) modparam("acc", "log_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") /* uncomment the following lines to enable DB accounting also */ #c#modparam("acc", "db_flag", 1) #c#modparam("acc", "db_missed_flag", 2) #c#modparam("domain", "db_url", #c# "mysql://openser:openserrw@localhost/openser") #c#modparam("acc", "db_extra", #c# "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") # ----- usrloc params ----- /* uncomment the following lines if you want to enable DB persistency for location entries */ #u#modparam("usrloc", "db_mode", 2) #u#modparam("usrloc", "db_url", #u# "mysql://openser:openserrw@localhost/openser") # ----- auth_db params ----- /* uncomment the following lines if you want to enable the DB based authentication */ #a#modparam("auth_db", "calculate_ha1", yes) #a#modparam("auth_db", "password_column", "password") #a#modparam("auth_db", "db_url", #a# "mysql://openser:openserrw@localhost/openser") #a#modparam("auth_db", "load_credentials", "") #parametros de autentificacion modificados manualmente modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "ha1") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 0) #modparam("auth_db", "use_domain", 0) modparam("auth_db", "use_domain", 1)#0 encendemos con 1 porque utilizaremos multi-domain modparam("auth_db", "load_credentials", "rpid") modparam("auth_db", "db_url", "mysql://openser:openserrw@localhost/openser") # ----- alias_db params ----- /* uncomment the following lines if you want to enable the DB based aliases */ #modparam("alias_db", "db_url", # "mysql://openser:openserrw@localhost/openser") # ----- domain params ----- /* uncomment the following lines to enable multi-domain detection support */ modparam("domain", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("domain", "db_mode", 1) # Use caching # ----- multi-module params ----- /* uncomment the following line if you want to enable multi-domain support in the modules (dafault off) */ modparam("alias_db|auth_db|usrloc|uri_db", "use_domain", 1) # ----- presence params ----- /* uncomment the following lines if you want to enable presence */ modparam("presence|presence_xml", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("presence_xml", "force_active", 1) modparam("presence", "server_address", "sip:212.4.107.250:5060") # -- nathelper modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@212.4.107.250") modparam("registrar|nathelper", "received_avp", "$avp(i:80)") modparam("usrloc", "nat_bflag", 6) modparam("nathelper", "sipping_method", "OPTIONS") ####### Routing Logic ######## # main request routing logic route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # NAT detection route(4); if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(1); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(2); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } } sl_send_reply("404","Not here"); } exit; } #initial requests # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { t_relay(); } exit; } t_check_trans(); # authentication route(3); # record routing if (!is_method("REGISTER|MESSAGE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } ##if (!uri==myself) /* replace with following line if multi-domain support is used */ if (!is_uri_host_local()) { append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS ##if($rd=="tls_domain1.net") { ## t_relay("tls:domain1.net"); ## exit; ##} else if($rd=="tls_domain2.net") { ## t_relay("tls:domain2.net"); ## exit; ##} route(1); } # requests for my domain if( is_method("PUBLISH|SUBSCRIBE")) { route(2); } if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } if ($rU==NULL) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # apply DB based aliases (uncomment to enable) ##alias_db_lookup("dbaliases"); if (!lookup("location")) { switch ($retcode) { 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 setflag(2); route(1); } route[1] { if (check_route_param("nat=yes")) { setbflag(6); setbflag(7);# sipping } if (isflagset(5) || isbflagset(6)) { route(5); } /* example how to enable some additional event routes */ if (is_method("INVITE")) { #t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); } exit; } # Presence route /* uncomment the whole following route for enabling presence server */ route[2] { 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; # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route /* uncomment the whole following route for enabling authentication */ route[3] { if (is_method("REGISTER")) { # authenticate the REGISTER requests (uncomment to enable auth) if (!www_authorize("", "subscriber")) { www_challenge("", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } # Auth only on registration #a# } else { #a# # authenticate if from local subscriber (uncomment to enable auth) #a# if (from_uri==myself) #a# { #a# if (!proxy_authorize("", "subscriber")) { #a# proxy_challenge("", "0"); #a# exit; #a# } #a# if (is_method("PUBLISH")) #a# { #a# if ($au!=$tU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } else { #a# if ($au!=$fU) { #a# sl_send_reply("403","Forbidden auth ID"); #a# exit; #a# } #a# } #a# #a# consume_credentials(); #a# # caller authenticated #a# } #a# } return; } # Caller NAT detection route /* uncomment the whole following route for enabling Caller NAT Detection */ route[4]{ force_rport(); if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); } setflag(5); } return; } # RTPProxy control /* uncomment the whole following route for enabling RTPProxy Control */ route[5] { if (is_method("BYE")) { unforce_rtp_proxy(); } else if (is_method("INVITE")){ force_rtp_proxy(); } if (!has_totag()) add_rr_param(";nat=yes"); return; } branch_route[1] { xdbg("new branch at $ru\n"); } onreply_route[1] { xdbg("incoming reply\n"); if ((isflagset(5) || isbflagset(6)) && status=~"(183)|(2[0-9][0-9])") { force_rtp_proxy(); } if (isbflagset(6)) { fix_nated_contact(); } } failure_route[1] { if (is_method("INVITE") && (isbflagset(6) || isflagset(5))) { unforce_rtp_proxy(); } if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} # 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(); ##} } ************************************************************************************************** ************************************************************************************************** And here goes the SIP Trace for a NATed to NATed hardphones: ************************************************************************************************** U +0.161561 212.4.97.115:35379 -> 212.4.107.250:5060 INVITE sip:20000000@212.4.107.250;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 332 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 192.168.254.110 t=0 0 m=audio 40000 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 # U +0.000407 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 100 Giving a try Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bK8f809670adc00668;rport=35379;received=212.4.97.115 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Server: Kamailio (1.5.1-notls (i386/linux)) Content-Length: 0 # U +0.000034 212.4.107.250:5060 -> 212.4.97.115:55128 INVITE sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: replaces, timer, path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 348 v=0 o=20000004 8000 8000 IN IP4 192.168.254.110 s=SIP Call c=IN IP4 212.4.107.250 t=0 0 m=audio 35752 RTP/AVP 4 3 18 0 8 9 97 a=sendrecv a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=ptime:60 a=nortpproxy:yes # U +0.019311 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone> Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Content-Length: 0 # U +0.030480 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +0.000083 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Length: 0 # U +6.510103 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000365 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.034122 212.4.97.115:35379 -> 212.4.107.250:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.254.110:5060;branch=z9hG4bKdf5e0ceed72f3797 Route: <sip:212.4.107.250;lr=on;nat=yes> From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@192.168.254.110:5060;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.000245 212.4.107.250:5060 -> 192.168.254.101:5060 ACK sip:20000000@192.168.254.101:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.2 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bKdf5e0ceed72f3797 From: "20000004" <sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Contact: <sip:20000004@212.4.97.115:35379;transport=udp;user=phone> Supported: path Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 ACK User-Agent: Grandstream BT200 1.1.6.27 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0 # U +0.458031 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000246 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes # U +0.999724 212.4.97.115:55128 -> 212.4.107.250:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 212.4.107.250;branch=z9hG4bK5974.c5c9aa24.0 Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 151 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 192.168.254.101 t=0 0 m=audio 32448 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv # U +0.000295 212.4.107.250:5060 -> 212.4.97.115:35379 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.110:5060;rport=35379;received=212.4.97.115;branch=z9hG4bK8f809670adc00668 From: "20000004"<sip:20000004@212.4.107.250;user=phone>;tag=ab6ba13b2f38a04e To: <sip:20000000@212.4.107.250;user=phone>;tag=c0a80101-21188 Call-ID: c177cae013da224d(a)192.168.254.110 CSeq: 29653 INVITE Require: timer Session-Expires: 100;refresher=uac Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:20000000@192.168.254.101:5060;user=phone> Record-Route: <sip:212.4.107.250;lr=on;nat=yes> Allow-Events: refer,dialog,message-summary,check-sync,talk,hold Content-Type: application/sdp Content-Length: 167 v=0 o=20000000 138812 138812 IN IP4 192.168.254.101 s=- c=IN IP4 212.4.107.250 t=0 0 m=audio 35754 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=nortpproxy:yes

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[Kamailio-Users] NAThelper and multipart SDP
by Pascal Maugeri 26 Jun '09

26 Jun '09

Hi Does NATHelper support properly multipart SDP (eg. INVITE with application/sdp and application/resource-lists+xml) ? Regards, Pascal

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