-------- Mensaje original -------- Return-Path: <[email protected]> Delivered-To: <[email protected]> Received: from palosanto.com by mail.palosanto.com (Dovecot) with LMTP id 1RBBAn9f2VPPOgAA3RMWGw for <[email protected]>; Wed, 30 Jul 2014 16:14:09 -0500 Received: from localhost (mail.palosanto.com [127.0.0.1]) by palosanto.com (Postfix) with ESMTP id BD82513C027E for <[email protected]>; Wed, 30 Jul 2014 16:14:09 -0500 (ECT) X-Virus-Scanned: Debian amavisd-new at mail.palosanto.com X-Spam-Flag: NO X-Spam-Score: -2.231 X-Spam-Level: X-Spam-Status: No, score=-2.231 tagged_above=-1000 required=6.31 tests=[AWL=-0.333, BAYES_00=-1.9, NORMAL_HTTP_TO_IP=0.001, WEIRD_PORT=0.001] autolearn=unavailable Received: from palosanto.com ([127.0.0.1]) by localhost (mail.palosanto.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id LEfbzVEOoMof for <[email protected]>; Wed, 30 Jul 2014 16:14:04 -0500 (ECT) Received: from www.kamailio.org (main.kamailio.org [193.22.119.66]) by palosanto.com (Postfix) with ESMTPS id 4412B13C0280 for <[email protected]>; Wed, 30 Jul 2014 16:13:56 -0500 (ECT) Received: from localhost ([127.0.0.1] helo=main.kamailio.org ident=list) by www.kamailio.org with esmtp (Exim 4.72) (envelope-from <[email protected]>) id 1XCbCY-0000J5-PL; Wed, 30 Jul 2014 23:14:22 +0200 Received: from lab2.palosanto.com ([201.234.196.173] helo=palosanto.com) by www.kamailio.org with esmtp (Exim 4.72) (envelope-from <[email protected]>) id 1XCbCU-0000II-SD for [email protected]; Wed, 30 Jul 2014 23:14:19 +0200 Received: from localhost (mail.palosanto.com [127.0.0.1]) by palosanto.com (Postfix) with ESMTP id 5BC5613C0280 for <[email protected]>; Wed, 30 Jul 2014 16:13:43 -0500 (ECT) X-Virus-Scanned: Debian amavisd-new at mail.palosanto.com Received: from palosanto.com ([127.0.0.1]) by localhost (mail.palosanto.com [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id jY5HkVjL2_h0 for <[email protected]>; Wed, 30 Jul 2014 16:13:39 -0500 (ECT) Received: from avillacis.palosanto.com (avillacis.palosanto.com [192.168.3.2]) by palosanto.com (Postfix) with ESMTPSA id 3DFFD13C0279 for <[email protected]>; Wed, 30 Jul 2014 16:13:39 -0500 (ECT) Message-ID: <[email protected]> Date: Wed, 30 Jul 2014 16:14:08 -0500 From: Alex Villacís Lasso <[email protected]> User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101 Thunderbird/24.7.0 MIME-Version: 1.0 To: Kamailio (SER) - Users Mailing List <[email protected]> Content-Type: multipart/mixed; boundary="------------030501020507030206060205" Subject: [SR-Users] kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header X-BeenThere: [email protected] X-Mailman-Version: 2.1.13 Precedence: list Reply-To: Kamailio (SER) - Users Mailing List <[email protected]> List-Id: "Kamailio \(SER\) - Users Mailing List" <sr-users.lists.sip-router.org> List-Unsubscribe: <http://lists.sip-router.org/cgi-bin/mailman/options/sr-users>, <mailto:[email protected]?subject=unsubscribe> List-Archive: <http://lists.sip-router.org/pipermail/sr-users> List-Post: <mailto:[email protected]> List-Help: <mailto:[email protected]?subject=help> List-Subscribe: <http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>, <mailto:[email protected]?subject=subscribe> Sender: [email protected] Errors-To: [email protected] I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an problem when handling SUBSCRIBE requests (as required for BLF and voicemail indications). My configuration is written so that these SUBSCRIBE requests are not handled by kamailio, but instead routed to asterisk. There is a failure to check From/To headers to see whether NOTIFY packets generated as part of a subscription can be restored using the information in Record-Route. The end result is that kamailio ends up sending packets with garbled tags that are (rightly) rejected by the SIP endpoint. The following is an example that demonstrates the issue (using Jitsi as endpoint): After registration, Jitsi sends a SUBSCRIBE request: SUBSCRIBE sip:[email protected] SIP/2.0 Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:[email protected]>;tag=bf427f4a To: "avillacisIM" <sip:[email protected]> Max-Forwards: 70 Contact: "avillacisIM" <sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 192.168.3.2:5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Proxy-Authorization: Digest username="avillacisIM",realm="pbx.villacis.com",nonce="U9lZJlPZV/r06Xep/ukc1UzAIO0V3TbS",uri="sip:[email protected]",response="0e18f4913c2693f6154c91f158fb17fe" Content-Length: 0 This packet is mangled by the configuration, and is sent to asterisk like this: SUBSCRIBE sip:[email protected] SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE From: "avillacisIM" <sip:[email protected]:5080>;tag=bf427f4a To: "avillacisIM" <sip:[email protected]:5080> Max-Forwards: 69 Contact: "avillacisIM" <sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com> User-Agent: Jitsi2.5.5255Linux Event: message-summary Accept: application/simple-message-summary Expires: 3600 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Content-Length: 0 The asterisk response for the SUBSCRIBE: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bKd941.2ab9cf36e41dc48855ae2cbe9a309d0a.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:[email protected]:5080>;tag=bf427f4a To: "avillacisIM" <sip:[email protected]:5080>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:[email protected]:5080>;expires=3600 Content-Length: 0 This is in turn transformed back by kamailio, and sent to Jitsi like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.2:5060;rport=5060;branch=z9hG4bK-343638-bd3ea073eb8920481b32962f3221eb6f Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "avillacisIM" <sip:[email protected]>;tag=bf427f4a To: "avillacisIM" <sip:[email protected]>;tag=as5562e95e Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 2 SUBSCRIBE Server: Asterisk PBX 11.11.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Expires: 3600 Contact: <sip:[email protected]:5080;alias=127.0.0.1~5080~1>;expires=3600 Content-Length: 0 Now asterisk wants to send a NOTIFY to the endpoint for the subscription. The NOTIFY looks like this: NOTIFY sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport Max-Forwards: 70 Route: <sip:127.0.0.1;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes>,<sip:192.168.2.18;r2=on;lr=on;ftag=bf427f4a;vsf=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;vst=AAAAAAAAAAAAAAAAAAAAHwAAAAAAAAAAAAAAAAAAAABAMTI3LjAuMC4xOjUwODA-;nat=yes> From: "asterisk" <sip:[email protected]:5080>;tag=as5562e95e To: <sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:[email protected]:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*[email protected]:5080 Voice-Message: 0/0 (0/0) Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:[email protected]:5080>;tag=as5562e95e) from the above request has nothing to do with the saved information (vsf parameter). Instead, it blindly mangles the From header, and does not even run a sanity check on the result before routing it. The end result is shown below. NOTIFY sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com SIP/2.0 Record-Route: <sip:192.168.2.18;r2=on;lr=on;ftag=as5562e95e> Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=as5562e95e> Via: SIP/2.0/UDP 192.168.2.18;branch=z9hG4bK8333.8bfe7bc2bd554a8631f0d00d463b28ee.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK658fa5fc;rport=5080 Max-Forwards: 69 From: "asterisk" <sip:asterisk@12(.0.0.1:[email protected]:5080>;tag=as5562e95e To: <sip:[email protected]:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a Contact: <sip:[email protected]:5080> Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0 CSeq: 102 NOTIFY User-Agent: Asterisk PBX 11.11.0 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 89 Messages-Waiting: no Message-Account: sip:*[email protected]:5080 Voice-Message: 0/0 (0/0) From examination of the source code, the vsf and vst strings are base64 encodings of the result of XORing the byte strings of the old and new tags. For this to work, the headers of future packets should match. However, here kamailio does not realize that the header does not match (by the ftag), and also does not check that the resulting "restored" header is a valid header.
#!KAMAILIO #!define WITH_ODBC #!define WITH_AUTH #!define WITH_IPAUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_PRESENCE #!define WITH_NAT #!define WITH_MULTIDOMAIN #!define WITH_XHTTP #!define WITH_WEBSOCKET #!define WITH_TLS #!define WITH_XCAPSRV # # Kamailio (OpenSER) SIP Server v4.1 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <[email protected]> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_ODBC #!ifndef DBURL #!define DBURL "unixodbc:///kamailio-connector" #!endif #!ifdef WITH_ASTERISK #!define DBASTURL "unixodbc:///elxpbx-connector" #!endif #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 #log_facility=LOG_LOCAL0 log_facility=LOG_LOCAL6 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 # force_rport=yes #!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 #!ifdef WITH_XHTTP tcp_accept_no_cl=yes #!endif tcp_rd_buf_size=32768 pv_buffer_size=2048 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_ODBC loadmodule "db_unixodbc.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #loadmodule "topoh.so" loadmodule "sqlops.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" loadmodule "ipops.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_ASTERISK loadmodule "uac.so" #!endif #!ifdef WITH_XHTTP loadmodule "xhttp.so" #!ifdef WITH_XCAPSRV loadmodule "pua.so" loadmodule "rls.so" loadmodule "xcap_server.so" #!endif #!ifdef WITH_WEBSOCKET loadmodule "msrp.so" loadmodule "websocket.so" loadmodule "sdpops.so" #endif #!endif #!ifdef WITH_XHTTP_RPC loadmodule "xhttp_rpc.so" #!endif #!ifdef WITH_XHTTP_PI loadmodule "xhttp_pi.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) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) #!ifdef WITH_ASTERISK /* set values to match defaults from asterisk */ modparam("registrar", "default_expires", 120) #!endif # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK # subscriber table is actually a view in DBASTURL modparam("auth_db", "use_domain", 1) modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "db_update_period", 20) #modparam("presence", "server_address", "sip:@127.0.0.1:5060") # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 0) modparam("presence_xml", "integrated_xcap_server", 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:[email protected]") # 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 #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif #!ifdef WITH_XHTTP #!ifdef WITH_XCAPSRV # ----- xcap_server params ----- modparam("xcap_server", "db_url", DBURL) modparam("xcap_server", "buf_size", 32768) modparam("pua", "db_url", DBURL) modparam("rls", "db_url", DBURL) modparam("rls", "integrated_xcap_server", 1) modparam("rls", "to_presence_code", 1024) modparam("rls", "server_address", "sip:[email protected]:5060") modparam("msrp", "cmap_size", 8) #!endif #!ifdef WITH_WEBSOCKET modparam("websocket", "keepalive_mechanism", 2) #!endif #!endif #!ifdef WITH_XHTTP_RPC modparam("xhttp_rpc", "xhttp_rpc_root", "http_rpc") #!endif #!ifdef WITH_XHTTP_PI modparam("xhttp_pi", "xhttp_pi_root", "http_pi") modparam("xhttp_pi", "framework", "//etc/kamailio/pi_framework.xml") #!endif modparam("sqlops", "sqlcon", "elxpbx=>unixodbc:///elxpbx-connector") #modparam("topoh", "mask_key", "_elastix_3_") #!ifdef WITH_ODBC modparam("db_unixodbc", "use_escape_common", 1) #!endif ####### Routing Logic ######## import_file "kamailio-mhomed-elastix.cfg" # 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); # run rtpproxy resolution route(MHOMED_ELASTIX); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { route(MHOMED_RR); } # 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] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. route(MHOMED_RR); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Value of $var(rr_advertise_address) is set in route(MHOMED_ELASTIX) route[MHOMED_RR] { if ($var(rr_advertise_address) != 0) { record_route_advertised_address("$var(rr_advertise_address)"); } else { record_route(); } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK # route(REGFWD); route(TOASTERISK); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK)) && (!sdp_content() || !sdp_with_media("message"))) { # 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; } #!endif $avp(oexten) = $rU; #xlog("L_ALERT", "ALERT: received routing request for ru=$ru rU=$rU rd=$rd ou=$ou fu=$fu tu=$tu\n"); uac_restore_from(); uac_restore_to(); if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server route route[PRESENCE] { #!ifdef WITH_XCAPSRV if(!is_method("PUBLISH|SUBSCRIBE|NOTIFY")) #!else if(!is_method("PUBLISH|SUBSCRIBE")) #!endif return; #!ifdef WITH_ASTERISK if (!route(FROMASTERISK)) { if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOASTERISK); } } #!else if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!endif #!ifdef WITH_ASTERISK # if routing to asterisk, asterisk should handle dialog if (!route(FROMASTERISK)) { if(is_method("SUBSCRIBE") && $hdr(Event)=="dialog") { route(TOASTERISK); } } #!endif #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { $var(ret_code) = rls_handle_subscribe(); if ($var(ret_code) == 1024) handle_subscribe(); t_release(); } else if (is_method("NOTIFY")) { rls_handle_notify(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif $var(tempfU) = $fU; #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address() && $au == "") { # Attempt to create a P-Asserted-Identity if none exists, to preserve # incoming Caller-ID if (!is_present_hf("P-Asserted-Identity")) { #append_hf("P-Asserted-Identity: <sip:$xavp(ra=>number)@$fd>\r\n"); append_hf("P-Asserted-Identity: <sip:$fU@$fd>\r\n"); } # Loading $fU from database using IP sql_pvquery("elxpbx", "SELECT name FROM sip WHERE host = '$si' AND sippasswd IS NULL", "$var(tempfU)"); # source IP allowed return; } #!endif if (is_method("REGISTER|INVITE") || from_uri==myself) { # authenticate requests #if (!auth_check("$fd", "subscriber", "1")) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); # # 1 - Contact header field is searched for occurrence of RFC1918 or rfc6598 addresses. # 2 - the "received" test is used: address in Via is compared against source IP address of signaling # 16 - test if the source port is different from the port in Via # 64 - test if the source connection of signaling is a WebSocket # ----- # 83 #!ifdef WITH_WEBSOCKET # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. if (nat_uac_test("83")) { #!else if (nat_uac_test("19")) { #!endif if (is_method("REGISTER")) { fix_nated_register(); } else { if (is_first_hop()) if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { # if(has_totag()) { # if(check_route_param("nat=yes")) { setbflag(FLB_NATB); # } # } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { return; } set_rtp_proxy_set("$var(rtpproxy_set)"); rtpproxy_manage("co", $var(rtpproxy_if)); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { # run rtpproxy resolution route(MHOMED_ELASTIX); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); # manage websocket reply if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { # run rtpproxy resolution route(MHOMED_ELASTIX); route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_XHTTP event_route[xhttp:request] { #!ifdef WITH_XHTTP_RPC $var(xhttp_rpc_root) = $(hu{s.substr,0,9}); if ($var(xhttp_rpc_root) == "/http_rpc") { dispatch_xhttp_rpc(); } #!endif #!ifdef WITH_XHTTP_PI $var(xhttp_rpc_root) = $(hu{s.substr,0,8}); if ($var(xhttp_rpc_root) == "/http_pi") { dispatch_xhttp_pi(); } #!endif #!ifdef WITH_XCAPSRV if ($hu =~ "^/xcap-root/") { route(XCAPSRV); } #!endif set_reply_close(); set_reply_no_connect(); # if ($Rp != 80 ##!ifdef WITH_TLS # && $Rp != 443 ##!endif # ) { # # xlog("L_WARN", "HTTP request received on $Rp\n"); # xhttp_reply("403", "Forbidden", "text/html", "Forbidden"); # exit; # } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN"; # && $hdr(Origin) != "https://communicator.MY_DOMAIN";) { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } #!endif #!ifdef WITH_XCAPSRV route[XCAPSRV] { # Remove port specification from Host header and use as authentication domain $var(http_domain) = $(hdr(Host){s.select,0,:}); if (!www_authorize("$var(http_domain)", "subscriber")) { www_challenge("$var(http_domain)", "0"); exit; } # xlog("L_ALERT", "===== xhttp: request [$rv] $rm => $hu\n"); set_reply_close(); set_reply_no_connect(); # Jitsi... if ($hu=~"^/xcap-root/resource-lists/users/.*/index$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/resource-lists\/users\/.*\/).*$/\1generallist.xml/}); else if ($hu=~"^/xcap-root/pres-rules/users/.*/presrules$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/)pres-rules(\/users\/.*\/)presrules/\1org.openmobilealliance.pres-rules\2pres-rules/}); else if ($hu=~"^/xcap-root/oma_status-icon/users/.*/.*$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/)oma_status-icon(\/users\/.*\/).*$/\1org.openmobilealliance.pres-content\2oma_status-icon\/index/}); # Bria... else if ($hu=~"^/xcap-root/resource-lists/users/.*/contacts-resource-list.xml$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/resource-lists\/users\/.*\/).*$/\1generallist.xml/}); else if ($hu=~"^/xcap-root/resource-lists/users/.*/resource-list.xml$") $var(doc_uri) = $(hu{re.subst,/(^\/xcap-root\/resource-lists\/users\/.*\/).*$/\1generallist.xml/}); else $var(doc_uri) = $hu; # xlog("L_ALERT", "===== xhttp: request will serve [$rv] $rm => $var(doc_uri)\n"); # xcap ops $xcapuri(u=>data) = $var(doc_uri); if($xcapuri(u=>xuid)=~"^sip:.+@.+") $var(uri) = $xcapuri(u=>xuid); else if($xcapuri(u=>xuid)=~".+@.+") $var(uri) = "sip:" + $xcapuri(u=>xuid); else $var(uri) = "sip:"+ $xcapuri(u=>xuid) + "@" + $Ri; # xlog("L_ALERT", "===== xhttp: $xcapuri(u=>auid) : $xcapuri(u=>xuid)\n"); if ($xcapuri(u=>auid) == "xcap-caps") { if ($rm == "GET") { $var(xbody) = "<?xml version='1.0' encoding='UTF-8'?> <xcap-caps xmlns='urn:ietf:params:xml:ns:xcap-caps'> <auids> <!-- <auid>org.openxcap.watchers</auid> --> <auid>rls-services</auid> <auid>resource-lists</auid> <auid>xcap-caps</auid> <!-- <auid>org.openxcap.dialog-rules</auid> --> <auid>org.openmobilealliance.pres-content</auid> <!-- <auid>org.openxcap.purge</auid> --> <auid>pres-rules</auid> <auid>org.openmobilealliance.pres-rules</auid> <auid>pidf-manipulation</auid> <auid>org.openmobilealliance.xcap-directory</auid> <auid>org.openmobilealliance.user-profile</auid> <auid>org.openmobilealliance.search</auid> </auids> <extensions></extensions> <namespaces> <!-- <namespace>http://openxcap.org/ns/watchers</namespace> --> <namespace>urn:ietf:params:xml:ns:rls-services</namespace> <namespace>urn:ietf:params:xml:ns:resource-lists</namespace> <namespace>urn:ietf:params:xml:ns:xcap-caps</namespace> <!-- <namespace>http://openxcap.org/ns/dialog-rules</namespace> --> <namespace>urn:oma:xml:prs:pres-content</namespace> <!-- <namespace>http://openxcap.org/ns/purge</namespace> --> <namespace>urn:ietf:params:xml:ns:pres-rules</namespace> <namespace>urn:ietf:params:xml:ns:pidf</namespace> <namespace>urn:oma:xml:xdm:xcap-directory</namespace> <namespace>urn:oma:xml:xdm:user-profile</namespace> <namespace>urn:oma:xml:xdm:search</namespace> </namespaces> </xcap-caps>"; xhttp_reply("200", "OK", "application/xcap-caps+xml", "$var(xbody)"); } else { append_to_reply("Allow: GET\r\n"); xhttp_reply("405", "Method Not Allowed", "", ""); } exit; } # be sure auth user access only its documents if ($au != $(var(uri){uri.user})) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>$si:$sp</body></html>"); exit; } switch($rm) { case "PUT": xcaps_put("$var(uri)", "$var(doc_uri)", "$rb"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "GET": xcaps_get("$var(uri)", "$var(doc_uri)"); exit; break; case "DELETE": xcaps_del("$var(uri)", "$var(doc_uri)"); if ($xcapuri(u=>auid) =~ "pres-rules") { pres_update_watchers("$var(uri)", "presence"); pres_refresh_watchers("$var(uri)", "presence", 1); } else if ($xcapuri(u=>auid) =~ "rls-services" || $xcapuri(u=>auid) =~ "resource-lists") { rls_update_subs("$var(uri)", "presence"); } else if ($xcapuri(u=>auid) =~ "pidf-manipulation") { pres_refresh_watchers("$var(uri)", "presence", 2, "$xcapuri(u=>uri_adoc)", "$xcapuri(u=>file)"); } exit; break; case "POST": if ($xcapuri(u=>auid) =~ "search") { xhttp_reply("501", "Not Implemented", "", ""); } else { if ($xcapuri(u=>auid) =~ "xcap-directory") { append_to_reply("Allow: GET\r\n"); } else { append_to_reply("Allow: DELETE, GET, PUT\r\n"); } xhttp_reply("405", "Method Not Allowed", "", ""); } exit; break; } # other http requests xhttp_reply("404", "Not Found", "", ""); exit; } #!endif #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $var(rip) = $sel(cfg_get.asterisk.bindip); $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); # If authorization user is identical to From: username, I will assume this is # a call coming from an extension within a domain. Otherwise, this request # will be unchanged, for incoming SIP trunks. if ($au == $fU || $ad != "") { # Further check - test whether unmagled name is a global trunk $var(istrunk) = 0; sql_pvquery("elxpbx", "SELECT COUNT(name) FROM sip WHERE name = '$au'", "$var(istrunk)"); if ($var(istrunk) != 0) { $var(istrunk) = 1; } else { $var(realfd) = $fd; if ($ad != "") { $var(realfd) = $ad; } # Encode domain part into username $var(newfrom) = $au + "_" + $var(realfd); $var(newfromuri) = "sip:" + $(var(newfrom){s.escape.user}) + "@" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); if ($ad != "" && $fU == $tU && $fd == $td) { $var(newto) = $var(newfrom); } else { $var(newto) = $tU + "_" + $td; } $var(newtouri) = "sip:" + $(var(newto){s.escape.user}) + "@" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); uac_replace_from("$var(newfromuri)"); uac_replace_to("$var(newtouri)"); consume_credentials(); } } else { if ($au != "") { $var(newfromuri) = "sip:" + $au + "@" + $fd; uac_replace_from("$var(newfromuri)"); } else { if ($au != $var(tempfU)) { $var(newfromuri) = "sip:" + $var(tempfU) + "@" + $fd; uac_replace_from("$var(newfromuri)"); } } } route(RELAY); exit; } event_route[msrp:frame-in] { xlog("L_ALERT","============#[[$msrp(method)]]===========\n"); xlog("L_ALERT","============*[[$si:$sp]]\n"); xlog("L_ALERT","============ crthop: [$msrp(crthop)]\n"); xlog("L_ALERT","============ prevhop: [$msrp(prevhop)]\n"); xlog("L_ALERT","============ nexthop: [$msrp(nexthop)]\n"); xlog("L_ALERT","============ firsthop: [$msrp(firsthop)]\n"); xlog("L_ALERT","============ lasthop: [$msrp(lasthop)]\n"); xlog("L_ALERT","============ prevhops: [$msrp(prevhops)]\n"); xlog("L_ALERT","============ nexthops: [$msrp(nexthops)]\n"); xlog("L_ALERT","============ srcaddr: [$msrp(srcaddr)]\n"); xlog("L_ALERT","============ srcsock: [$msrp(srcsock)]\n"); xlog("L_ALERT","============ sessid: [$msrp(sessid)]\n"); xlog("L_ALERT","============ From-Path:[$hdr(From-Path)]\n"); xlog("L_ALERT","============ To-Path: [$hdr(To-Path)]\n"); xlog("L_ALERT","============ ad: [$ad]\n"); msrp_reply_flags("1"); if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if ($msrp(nexthops)>0) { msrp_relay(); exit; } # Kamailio 4.1.4 is currently unable to build a RFC-compliant MSRP challenge or # check a RFC-compliant authentication response. Skipping authentication altogether # is also not RFC-compliant, but at least some progress is made. # # $var(msrprealm) = $(hdr(To-Path){msrpuri.host}); # xlog("L_ALERT","============ msrprealm: [$var(msrprealm)]\n"); # if (!www_authenticate("$var(msrprealm)", "subscriber", "$msrp(method)")) { # if(auth_get_www_authenticate("$var(msrprealm)", "1", "$var(wauth)")) { # $var(s1) = "qop=\"auth\""; # $var(s2) = "qop=\"auth\", opaque=\"0123456789abcdef\""; # $var(wauth) = $(var(wauth){s.replace,$var(s1),$var(s2)}); # msrp_reply("401", "Unauthorized", "$var(wauth)"); # } else { # msrp_reply("500", "Server Error"); # } # exit; # } msrp_cmap_save(); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } if (msrp_cmap_lookup()) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_relay(); } else { msrp_reply("481", "Session-does-not-exist"); } } else { msrp_reply("501", "Request-method-not-understood"); } } # Forward REGISTER to Asterisk #route[REGFWD] { # 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); # # # Encode domain part into username # $var(newfrom) = $fU + "_" + $fd; # $var(newfromuri) = "sip:" + $(var(newfrom){s.escape.user}) + "@" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); # uac_replace_from("$var(newfromuri)"); # $var(newto) = $tU + "_" + $td; # $var(newtouri) = "sip:" + $(var(newto){s.escape.user}) + "@" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); # uac_replace_to("$var(newtouri)"); # # $var(encodeuser) = $au + "_" + $fd; # $uac_req(furi)=$var(newfromuri); # # $uac_req(turi)=$var(newtouri); # # $var(encodeuser) = $au + "_" + $fd; # $uac_req(hdrs)="Contact: <sip:" + $(var(encodeuser){s.escape.user}) + "@" # + $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(); #} #!endif
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