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:avillaci...@pbx.villacis.com SIP/2.0
Call-ID: 87ff107c2665316f4e257b358c54b3d4@0:0:0:0:0:0:0:0
CSeq: 2 SUBSCRIBE
From: "avillacisIM" <sip:avillaci...@pbx.villacis.com>;tag=bf427f4a
To: "avillacisIM" <sip:avillaci...@pbx.villacis.com>
Max-Forwards: 70
Contact: "avillacisIM"
<sip:avillacisIM@192.168.3.2:5060;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:avillaci...@pbx.villacis.com",response="0e18f4913c2693f6154c91f158fb17fe"
Content-Length: 0
This packet is mangled by the configuration, and is sent to asterisk like this:
SUBSCRIBE sip:avillaci...@pbx.villacis.com 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:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a
To: "avillacisIM" <sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>
Max-Forwards: 69
Contact: "avillacisIM"
<sip:avillacisIM@192.168.3.2:5060;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:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;tag=bf427f4a
To: "avillacisIM"
<sip:avillacisIM_pbx.villacis.com@127.0.0.1:5080>;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:avillacisIM@127.0.0.1:5080>;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:avillaci...@pbx.villacis.com>;tag=bf427f4a
To: "avillacisIM" <sip:avillaci...@pbx.villacis.com>;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:avillacisIM@127.0.0.1:5080;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:avillacisIM@192.168.3.2:5060;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:asterisk@127.0.0.1:5080>;tag=as5562e95e
To:
<sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a
Contact: <sip:asterisk@127.0.0.1:5080>
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:*97@127.0.0.1:5080
Voice-Message: 0/0 (0/0)
Here is where the bug appears. The autoprocessing does not recognize that the From header (From: "asterisk" <sip:asterisk@127.0.0.1:5080>;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:avillacisIM@192.168.3.2:5060;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(.0.0.1:5080.....@127.0.0.1:5080>;tag=as5562e95e
To:
<sip:avillacisIM@192.168.3.2:5060;transport=udp;registering_acc=pbx_villacis_com>;tag=bf427f4a
Contact: <sip:asterisk@127.0.0.1:5080>
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:*97@127.0.0.1:5080
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: <sr-users@lists.sip-router.org>
#
# 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:pin...@kamailio.org")
# 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:rls@127.0.0.1:5060")
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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