sr-users October 2014

sr-users@lists.kamailio.org

101 participants
186 discussions

rtpengine with rejected re-invites to new RTP ports

by Jeff Pyle

Hello, Given the following scenario with Kamailio and rtpengine in the middle: - call establishes with G.711 RTP - b-leg re-invites to T.38, indicating a different port number then he is using for G.711 - a-leg refuses the re-invite with a 488 - call continues using G.711 on original port numbers as if re-invite never happened Will Kamailio and rtpengine handle it? Today I have a functional configuration with rtpproxy on another SIP proxy. It handles every scenario I have tried except this one. rtpproxy sees the new ports in the re-invite and adjusts its session accordingly. If the re-invite is rejected, the old media ports are no longer valid through rtpproxy, and the call fails. Is there an approach I can take with Kamailio and rtpengine to allow this scenario to succeed? Regards, Jeff

9 years, 6 months

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis

by cpcnetworking

Thx, exactly what I was looking for!

9 years, 6 months

tcp_read_data() errors in syslog

by Petr.Wozniak＠seznam.cz

Hello All, I have installed kamailio with TLS module together with siremis on server and all seems works fine except that sometimes appear in syslog file the following errors: /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection timed out (110) /usr/sbin/kamailio[3266]: ERROR: <core> [tcp_read.c:1281]: tcp_read_req(): ERROR: tcp_read_req: error reading /usr/sbin/kamailio[3268]: ERROR: <core> [tcp_read.c:289]: tcp_read_data(): error reading: Connection reset by peer (104) Some part of syslog please you find enclosed. I don't know where I should search for the cause of these errors and how to debug and remove them. I don't know if these errors are caused by incorrect kamailio configuration or are caused on client's side. Thank you for your opinions and help in advance Regards, Petr

9 years, 6 months

Adding Dynamic Routing Module management to Siremis

by cpcnetworking

I'm using Kamailio 4.15 with Siremis 4.1.0 both are great BTW! I'd like to add management of the Kamailio Dynamic Routing Module to Siremis web manager Any pointers on how to do this; I'm not familiar with OpenBiz Thx in advance

9 years, 6 months

multiple endpoints call

by Marino Mileti

Take a look at : modparam("registrar", "max_contacts", x) to have up to "x" contacts for AOR and in your REGISTRAR route if your save("location") has some extra parameters (for example the 0x04 used to restrict one single record in loaction table ----Messaggio originale---- Da: ovoshlook(a)gmail.com Data: 6-ott-2014 22.02 A: "Kamailio (SER) - Users Mailing List"<sr-users(a)lists.sip-router.org> Ogg: [SR-Users] multiple endpoints call Hello. I have multiple endpoints registered at my kamailio with one account (for example user1 registerd from norhway, USA, and Russia at one time), so when I call from user2 to user1 I want to ring all endpoints registered by user1 account. Now I can ring only one, first entry at location table. What setting may I use for this? Thanks

9 years, 6 months

Re: [SR-Users] Adding Dynamic Routing Module management to Siremis

by cpcnetworking

ok thanks

9 years, 6 months

tcp_read_data() errors in syslog

by Petr.Wozniak＠seznam.cz

9 years, 6 months

Presence PUBLISH error

by Marino Mileti

Hi guys, with Kamailio 4.1.6 standard configuration and dbtext as db module, when server receives a publish message on the log there's: ERROR: db_text [dbt_base.c:391]: dbt_insert(): incompatible types v[5] - c[7]! ERROR: presence [presentity.c:402]: update_presentity(): inserting new record in database ERROR: presence [publish.c:464]: handle_publish(): when updating presentity and the server reply with 500 Server Internal Error. My conf is standard I've only used DBURL to point to my dbtext files: I've enable presence and presence_xml module # ----- presence params ----- modparam("presence", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence", "db_table_lock_type", 0) modparam("presence_xml", "db_url", text:///opt/kamailio-orig/dbtext) modparam("presence_xml", "force_active", 1) Any idea? Marino

9 years, 6 months

Problem when I want install kamailio on CENTOS 7 x64

by ANTHONY HESNAUX

Hello all, I want to install kamailio on CENTOS 7 x64 but I have this problem when I execute "kamdbctl create » Can you help me ?? Thank YOU Anthony Console display: [root@localhost kamailio-4.0.1]# /usr/local/sbin/kamdbctl create MySQL password for root: INFO: test server charset Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. INFO: creating database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: granting privileges to database kamailio ... Warning: Using a password on the command line interface can be insecure. INFO: creating standard tables into kamailio ... Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. Warning: Using a password on the command line interface can be insecure. ERROR 1101 (42000) at line 2: BLOB/TEXT column 'body' can't have a default value ERROR: Creating core tables failed at msilo! [root@localhost kamailio-4.0.1]#

9 years, 6 months

BYE issue - Kamailio +RTP proxy

by balu

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@76.42.89.25:5060 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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;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@76.42.89.25:5060 SIP/2.0 Via: SIP/2.0/UDP 94.50.203.32:8764 ;branch=z9hG4bKf7c3.3c8b946c31cdd32f930549d06acfbe9a.0 Max-Forwards: 69 From: sip:test@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12140 ACK Content-Length: 0 INVITE sip:18792356789@76.42.89.25:5060 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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25 Contact: <sip:test@192.168.2.11:58429;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@76.42.89.25:5060", 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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:18792356789@76.42.89.25;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@76.42.89.25;tag=bP9Zm5tvFPFiwrGTOXxp6mAZY1L9cBGE To: sip:18792356789@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz CSeq: 12141 ACK Content-Length: 0 BYE sip:test@192.168.2.11:58429;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@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;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@76.42.89.25;tag=19093814023234994340686221 Call-ID: wjTFhNWrPQu9AUldOZNBSN6peHfp2myz To: sip:test@76.42.89.25;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:kamailiorw@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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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 }

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sr-users October 2014