balu
2014-10-06 10:56:41 UTC
Dear experts ,
I am using kamailio with rtp proxy module. I have 2 questions /issues .
1. When caller or callee ends the call the other end call is not
disocnnecting .
UA is pjsip based and behind NAT router. Present call flow is
pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP
proxy----> ThridParty SIP Server
UA local ip : 192.168.2.11
UA public IP : 89.78.92.23
Kamailio Public ip: 94.50.203.32
Third party Sip server : 76.42.89.25
Here When I disconnect call from either side , it is not disconnecting
other side .
2. My second requirement is , how can I define port of third party server .
for example if have 3 or 4 sip servers with different sip registration
ports other tahn 5060
How can I route registration requests coming from UAs to different ports of
third party servers.
Please bear my ignorance I am new to kamailio .Hope some experts will help
me here .
Attached kamailio config and SIP trace taken from kamailio server
Thank you
INVITE sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p.
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org
Contact: <sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23~58429~1>
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12140 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9
Content-Type: application/sdp
Content-Length: 369
P-hint: outbound
v=0
o=- 3620097527 3620097527 IN IP4 94.50.203.32
s=pjmedia
c=IN IP4 94.50.203.32
t=0 0
a=X-nat:0
m=audio 60822 RTP/AVP 104 18 0 8 96
c=IN IP4 94.50.203.32
a=rtcp:60823
a=sendrecv
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=nortpproxy:yes
SIP/2.0 407 Proxy Authentication Required
CSeq: 12140 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p.
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Proxy-Authenticate: DIGEST realm="sip.testcalls.com",
nonce="141110871219020226209383237537"
Content-Length: 0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
ACK sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12140 ACK
Content-Length: 0
INVITE sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.4f21;nat=yes>
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org
Contact: <sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23~58429~1>
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12141 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9
Proxy-Authorization: Digest username="test", realm="sip.testcalls.com",
nonce="141110871219020226209383237537", uri="
sip:18792356789-Ioo0U+***@public.gmane.org:5060",
response="e98f243028e20a3d864dc54149db8ab1"
Content-Type: application/sdp
Content-Length: 369
P-hint: outbound
v=0
o=- 3620097527 3620097527 IN IP4 94.50.203.32
s=pjmedia
c=IN IP4 94.50.203.32
t=0 0
a=X-nat:0
m=audio 47574 RTP/AVP 104 18 0 8 96
c=IN IP4 94.50.203.32
a=rtcp:47575
a=sendrecv
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=nortpproxy:yes
SIP/2.0 183 Session Progress
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 180 Ringing
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 180 Ringing
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 200 OK
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
ACK sip:76.42.89.25:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.3f67e202ad7510de3a935364e562755c.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPjpDeX-U.AQN7jSbWJ7Tqg4Sbl0Lby6nkE
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12141 ACK
Content-Length: 0
BYE sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23%7E58429%7E1 SIP/2.0
Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
CSeq: 1 BYE
Via: SIP/2.0/UDP 76.42.89.25:5060;branch=z9hG4bK190938140255193499457203
From: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Content-Length: 0
Max-Forwards: 70
SIP/2.0 200 OK
CSeq: 1 BYE
Via: SIP/2.0/UDP 76.42.89.25:5060
;branch=z9hG4bK190938140255193499457203;rport=5060
From: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Server: kamailio (4.1.3 (i386/linux))
Content-Length: 0
-----------
config file below
-----------
#!KAMAILIO
#
check_via=no
rev_dns=no
dns=no
#!define WITH_NAT
#!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 #########
#!ifdef WITH_MYSQL
#!ifndef DBURL
#!define DBURL "mysql://kamailio:***@localhost/kamailio"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
#!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
#!define WITH_DEBUG
#!ifdef WITH_DEBUG
debug=2
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
memdbg=2
memlog=2
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes
#sreekanth commented no
auto_aliases=no
/* add local domain aliases */
#alias="sip.mydomain.com"
listen=udp:94.50.203.32:8304
/* port to listen to
* - can be specified more than once if needed to listen on many ports */
#port=3074
#!ifdef WITH_TLS
enable_tls=yes
#!endif
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
#!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/local/lib/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.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 "mangler.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 1)
#sreekanth changed above value from 0 to 1
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# 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=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger-j+PTJLu+***@public.gmane.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(i:42)")
#sreekanth changed above $avp(RECEIVED) to $avp(i:42)
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
#added by sreekanth
#!define DLG_FLAG 28
#!define CC_FLAG 29
loadmodule "dialog.so"
modparam("dialog", "hash_size", 2048)
modparam("dialog", "default_timeout", 3600)
modparam("dialog", "db_mode", 0)
modparam("dialog", "dlg_flag", DLG_FLAG)
loadmodule "rtimer.so";
#!ifdef CNXCC_CHANNEL
modparam("rtimer", "timer", "name=ta;interval=1;mode=1;")
modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT")
#!endif
loadmodule "cnxcc.so"
modparam("cnxcc", "dlg_flag", CC_FLAG)
modparam("cnxcc", "credit_check_period", 1) #check every 1 second
#sreekanth-End
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
#setflag(DLG_FLAG);
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
if (is_method("INVITE")) {
route(RELAY);
}
# 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"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
sl_send_reply("100", "Trying");
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);
#t_newtran();
#t_reply("200", "OK");
# exit;
}
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.
record_route();
}
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;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
exit;
}
}
# USER location service
route[LOCATION] {
#!ifdef WITH_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
$avp(oexten) = $rU;
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] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
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;
}
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
}
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
} else if(is_method("SUBSCRIBE")) {
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$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();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop())
set_contact_alias();
}
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;
rtpproxy_manage("co");
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
#if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
#{
# rtpproxy_manage();
# return;
#}
#from here sreekanth commented
#if (is_request()) {
#rtpproxy_manage("co"); }
#if (is_reply()) {
#rtpproxy_manage("z50"); }
#if (is_request()) {
# if (!has_totag()) {
# if(t_is_branch_route()) {
# if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))) {
# add_rr_param(";nat=yes");
# }
# }
# }
# }
#sreekanth till here
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()) {
handle_ruri_alias();
}
#!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] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!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
}
I am using kamailio with rtp proxy module. I have 2 questions /issues .
1. When caller or callee ends the call the other end call is not
disocnnecting .
UA is pjsip based and behind NAT router. Present call flow is
pjsipUA (LAN_ip)----->Router (Publicip)-------->Kamailio_with_RTP
proxy----> ThridParty SIP Server
UA local ip : 192.168.2.11
UA public IP : 89.78.92.23
Kamailio Public ip: 94.50.203.32
Third party Sip server : 76.42.89.25
Here When I disconnect call from either side , it is not disconnecting
other side .
2. My second requirement is , how can I define port of third party server .
for example if have 3 or 4 sip servers with different sip registration
ports other tahn 5060
How can I route registration requests coming from UAs to different ports of
third party servers.
Please bear my ignorance I am new to kamailio .Hope some experts will help
me here .
Attached kamailio config and SIP trace taken from kamailio server
Thank you
INVITE sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p.
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org
Contact: <sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23~58429~1>
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12140 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9
Content-Type: application/sdp
Content-Length: 369
P-hint: outbound
v=0
o=- 3620097527 3620097527 IN IP4 94.50.203.32
s=pjmedia
c=IN IP4 94.50.203.32
t=0 0
a=X-nat:0
m=audio 60822 RTP/AVP 104 18 0 8 96
c=IN IP4 94.50.203.32
a=rtcp:60823
a=sendrecv
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=nortpproxy:yes
SIP/2.0 407 Proxy Authentication Required
CSeq: 12140 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPjUMSliGvGug7LTZQNqrrpmLN7hggWq7p.
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Proxy-Authenticate: DIGEST realm="sip.testcalls.com",
nonce="141110871219020226209383237537"
Content-Length: 0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
ACK sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12140 ACK
Content-Length: 0
INVITE sip:18792356789-Ioo0U+***@public.gmane.org:5060 SIP/2.0
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.4f21;nat=yes>
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org
Contact: <sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23~58429~1>
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12141 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER,
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Siphon PjSip v2.0.0-beta/arm-apple-darwin9
Proxy-Authorization: Digest username="test", realm="sip.testcalls.com",
nonce="141110871219020226209383237537", uri="
sip:18792356789-Ioo0U+***@public.gmane.org:5060",
response="e98f243028e20a3d864dc54149db8ab1"
Content-Type: application/sdp
Content-Length: 369
P-hint: outbound
v=0
o=- 3620097527 3620097527 IN IP4 94.50.203.32
s=pjmedia
c=IN IP4 94.50.203.32
t=0 0
a=X-nat:0
m=audio 47574 RTP/AVP 104 18 0 8 96
c=IN IP4 94.50.203.32
a=rtcp:47575
a=sendrecv
a=rtpmap:104 iLBC/8000
a=fmtp:104 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=nortpproxy:yes
SIP/2.0 183 Session Progress
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 180 Ringing
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 180 Ringing
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
SIP/2.0 200 OK
CSeq: 12141 INVITE
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.4e8266c84727e8ceeaa14c37b4b48ef7.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;branch=z9hG4bKPj.fUOV5Ds9oW3oGonVCpIwM9tEpoNUacJ
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Contact: <sip:76.42.89.25:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 224
Record-Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
v=0
o=VoipSwitch 6220 7220 IN IP4 76.42.89.25
s=VoipSIP
i=Audio Session
c=IN IP4 76.42.89.25
t=0 0
m=audio 6220 RTP/AVP 8 96
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=sendrecv
ACK sip:76.42.89.25:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 94.50.203.32:8764
;branch=z9hG4bK08c3.3f67e202ad7510de3a935364e562755c.0
Via: SIP/2.0/UDP 192.168.2.11:58429
;received=89.78.92.23;rport=58429;branch=z9hG4bKPjpDeX-U.AQN7jSbWJ7Tqg4Sbl0Lby6nkE
Max-Forwards: 69
From: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
To: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
CSeq: 12141 ACK
Content-Length: 0
BYE sip:test-***@public.gmane.org:58429;ob;alias=89.78.92.23%7E58429%7E1 SIP/2.0
Route: <sip:94.50.203.32:8764
;lr=on;ftag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE;did=691.3f21;nat=yes>
CSeq: 1 BYE
Via: SIP/2.0/UDP 76.42.89.25:5060;branch=z9hG4bK190938140255193499457203
From: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Content-Length: 0
Max-Forwards: 70
SIP/2.0 200 OK
CSeq: 1 BYE
Via: SIP/2.0/UDP 76.42.89.25:5060
;branch=z9hG4bK190938140255193499457203;rport=5060
From: sip:18792356789-Ioo0U+***@public.gmane.org;tag=19093814023234994340686221
Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz
To: sip:test-Ioo0U+***@public.gmane.org;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE
Server: kamailio (4.1.3 (i386/linux))
Content-Length: 0
-----------
config file below
-----------
#!KAMAILIO
#
check_via=no
rev_dns=no
dns=no
#!define WITH_NAT
#!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 #########
#!ifdef WITH_MYSQL
#!ifndef DBURL
#!define DBURL "mysql://kamailio:***@localhost/kamailio"
#!endif
#!endif
#!ifdef WITH_MULTIDOMAIN
# - the value for 'use_domain' parameters
#!define MULTIDOMAIN 1
#!else
#!define MULTIDOMAIN 0
#!endif
#!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
#!define WITH_DEBUG
#!ifdef WITH_DEBUG
debug=2
log_stderror=no
#!else
debug=2
log_stderror=no
#!endif
memdbg=2
memlog=2
log_facility=LOG_LOCAL0
fork=yes
children=4
/* uncomment the next line to disable TCP (default on) */
disable_tcp=yes
#sreekanth commented no
auto_aliases=no
/* add local domain aliases */
#alias="sip.mydomain.com"
listen=udp:94.50.203.32:8304
/* port to listen to
* - can be specified more than once if needed to listen on many ports */
#port=3074
#!ifdef WITH_TLS
enable_tls=yes
#!endif
# life time of TCP connection when there is no traffic
# - a bit higher than registration expires to cope with UA behind NAT
tcp_connection_lifetime=3605
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface
# - see the documentation of 'cfg_rpc' module.
#
# Format: group.id = value 'desc' description
# Access: $sel(cfg_get.group.id) or @cfg_get.group.id
#
#!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/local/lib/kamailio/modules/"
#!endif
#!ifdef WITH_MYSQL
loadmodule "db_mysql.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 "mangler.so"
#!ifdef WITH_AUTH
loadmodule "auth.so"
loadmodule "auth_db.so"
#!ifdef WITH_IPAUTH
loadmodule "permissions.so"
#!endif
#!endif
#!ifdef WITH_ALIASDB
loadmodule "alias_db.so"
#!endif
#!ifdef WITH_SPEEDDIAL
loadmodule "speeddial.so"
#!endif
#!ifdef WITH_MULTIDOMAIN
loadmodule "domain.so"
#!endif
#!ifdef WITH_PRESENCE
loadmodule "presence.so"
loadmodule "presence_xml.so"
#!endif
#!ifdef WITH_NAT
loadmodule "nathelper.so"
loadmodule "rtpproxy.so"
#!endif
#!ifdef WITH_TLS
loadmodule "tls.so"
#!endif
#!ifdef WITH_ANTIFLOOD
loadmodule "htable.so"
loadmodule "pike.so"
#!endif
#!ifdef WITH_XMLRPC
loadmodule "xmlrpc.so"
#!endif
#!ifdef WITH_DEBUG
loadmodule "debugger.so"
#!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params -----
modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo")
# ----- tm params -----
# auto-discard branches from previous serial forking leg
modparam("tm", "failure_reply_mode", 3)
# default retransmission timeout: 30sec
modparam("tm", "fr_timer", 30000)
# default invite retransmission timeout after 1xx: 120sec
modparam("tm", "fr_inv_timer", 120000)
# ----- rr params -----
# add value to ;lr param to cope with most of the UAs
modparam("rr", "enable_full_lr", 1)
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 1)
#sreekanth changed above value from 0 to 1
# ----- registrar params -----
modparam("registrar", "method_filtering", 1)
/* uncomment the next line to disable parallel forking via location */
# modparam("registrar", "append_branches", 0)
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# max value for expires of registrations
modparam("registrar", "max_expires", 3600)
# 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=$rd")
#!endif
# ----- usrloc params -----
/* enable DB persistency for location entries */
#!ifdef WITH_USRLOCDB
modparam("usrloc", "db_url", DBURL)
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "use_domain", MULTIDOMAIN)
#!endif
# ----- auth_db params -----
#!ifdef WITH_AUTH
modparam("auth_db", "db_url", DBURL)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "")
modparam("auth_db", "use_domain", MULTIDOMAIN)
# ----- permissions params -----
#!ifdef WITH_IPAUTH
modparam("permissions", "db_url", DBURL)
modparam("permissions", "db_mode", 1)
#!endif
#!endif
# ----- alias_db params -----
#!ifdef WITH_ALIASDB
modparam("alias_db", "db_url", DBURL)
modparam("alias_db", "use_domain", MULTIDOMAIN)
#!endif
# ----- speeddial params -----
#!ifdef WITH_SPEEDDIAL
modparam("speeddial", "db_url", DBURL)
modparam("speeddial", "use_domain", MULTIDOMAIN)
#!endif
# ----- domain params -----
#!ifdef WITH_MULTIDOMAIN
modparam("domain", "db_url", DBURL)
# register callback to match myself condition with domains list
modparam("domain", "register_myself", 1)
#!endif
#!ifdef WITH_PRESENCE
# ----- presence params -----
modparam("presence", "db_url", DBURL)
# ----- presence_xml params -----
modparam("presence_xml", "db_url", DBURL)
modparam("presence_xml", "force_active", 1)
#!endif
#!ifdef WITH_NAT
# ----- rtpproxy params -----
modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722")
# ----- nathelper params -----
modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", FLB_NATSIPPING)
modparam("nathelper", "sipping_from", "sip:pinger-j+PTJLu+***@public.gmane.org")
# params needed for NAT traversal in other modules
modparam("nathelper|registrar", "received_avp", "$avp(i:42)")
#sreekanth changed above $avp(RECEIVED) to $avp(i:42)
modparam("usrloc", "nat_bflag", FLB_NATB)
#!endif
#!ifdef WITH_TLS
# ----- tls params -----
modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg")
#!endif
#!ifdef WITH_ANTIFLOOD
# ----- pike params -----
modparam("pike", "sampling_time_unit", 2)
modparam("pike", "reqs_density_per_unit", 16)
modparam("pike", "remove_latency", 4)
# ----- htable params -----
# ip ban htable with autoexpire after 5 minutes
modparam("htable", "htable", "ipban=>size=8;autoexpire=300;")
#!endif
#!ifdef WITH_XMLRPC
# ----- xmlrpc params -----
modparam("xmlrpc", "route", "XMLRPC");
modparam("xmlrpc", "url_match", "^/RPC")
#!endif
#!ifdef WITH_DEBUG
# ----- debugger params -----
modparam("debugger", "cfgtrace", 1)
#!endif
#added by sreekanth
#!define DLG_FLAG 28
#!define CC_FLAG 29
loadmodule "dialog.so"
modparam("dialog", "hash_size", 2048)
modparam("dialog", "default_timeout", 3600)
modparam("dialog", "db_mode", 0)
modparam("dialog", "dlg_flag", DLG_FLAG)
loadmodule "rtimer.so";
#!ifdef CNXCC_CHANNEL
modparam("rtimer", "timer", "name=ta;interval=1;mode=1;")
modparam("rtimer", "exec", "timer=ta;route=SHOW_CHANNEL_COUNT")
#!endif
loadmodule "cnxcc.so"
modparam("cnxcc", "dlg_flag", CC_FLAG)
modparam("cnxcc", "credit_check_period", 1) #check every 1 second
#sreekanth-End
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
#setflag(DLG_FLAG);
# per request initial checks
route(REQINIT);
# NAT detection
route(NATDETECT);
if (is_method("INVITE")) {
route(RELAY);
}
# 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"))
record_route();
# account only INVITEs
if (is_method("INVITE"))
{
sl_send_reply("100", "Trying");
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);
#t_newtran();
#t_reply("200", "OK");
# exit;
}
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.
record_route();
}
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;
}
}
# Handle SIP registrations
route[REGISTRAR] {
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if (!save("location"))
sl_reply_error();
exit;
}
}
# USER location service
route[LOCATION] {
#!ifdef WITH_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
$avp(oexten) = $rU;
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] {
if(!is_method("PUBLISH|SUBSCRIBE"))
return;
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;
}
#!ifdef WITH_PRESENCE
if (!t_newtran())
{
sl_reply_error();
exit;
}
if(is_method("PUBLISH"))
{
handle_publish();
t_release();
} else if(is_method("SUBSCRIBE")) {
handle_subscribe();
t_release();
}
exit;
#!endif
# if presence enabled, this part will not be executed
if (is_method("PUBLISH") || $rU==$null)
{
sl_send_reply("404", "Not here");
exit;
}
return;
}
# Authentication route
route[AUTH] {
#!ifdef WITH_AUTH
#!ifdef WITH_IPAUTH
if((!is_method("REGISTER")) && allow_source_address())
{
# source IP allowed
return;
}
#!endif
if (is_method("REGISTER") || from_uri==myself)
{
# authenticate requests
if (!auth_check("$fd", "subscriber", "1")) {
auth_challenge("$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();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
if(is_first_hop())
set_contact_alias();
}
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;
rtpproxy_manage("co");
if (is_request()) {
if (!has_totag()) {
if(t_is_branch_route()) {
add_rr_param(";nat=yes");
}
}
}
#if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
#{
# rtpproxy_manage();
# return;
#}
#from here sreekanth commented
#if (is_request()) {
#rtpproxy_manage("co"); }
#if (is_reply()) {
#rtpproxy_manage("z50"); }
#if (is_request()) {
# if (!has_totag()) {
# if(t_is_branch_route()) {
# if ((isflagset(FLT_NATS) || isbflagset(FLB_NATB))) {
# add_rr_param(";nat=yes");
# }
# }
# }
# }
#sreekanth till here
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()) {
handle_ruri_alias();
}
#!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] {
xdbg("incoming reply\n");
if(status=~"[12][0-9][0-9]")
route(NATMANAGE);
}
# manage failure routing cases
failure_route[MANAGE_FAILURE] {
route(NATMANAGE);
if (t_is_canceled()) {
exit;
}
#!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
}