- sr-users - kamailio.org

dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

5 6

Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

2 1

Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael

4 5

Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel.

3 10

How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication Schemes? can't find it in hss_db, kamailio and not in scscf. Thank you, Moshe.

1 0

call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** 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 block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** 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_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_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* 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 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### 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 */ # mpath="/usr/lib/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.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 "acc.so" loadmodule "counters.so" loadmodule "dispatcher.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" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!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 --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(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) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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")) { 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); route(DISPATCH); # user location service } # Wrapper for relaying requests 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; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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($ua =~ "friendly-scanner|sipcli|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } 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 processing 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; } # IP authorization and user authentication 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; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection 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 and signaling updates for NAT traversal 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; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif 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(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!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) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } 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 # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($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 reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context 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_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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 } -- *Regards:* Gaurav Kumar

2 2

How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no message event. Please review my config file at following URL: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4 Developers, I also want to know if there is any way to log *text message* in kamailio server.

3 3

Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamailio-minimal-proxy.cfg* <https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…> and I changed the local domain alias with SIP IP -* 10.50.185.2*. When I try to connect to SIP via proxy using *linphone* (UDP) I got below errors in* /var/log/syslog* and shows *connection in progress* every time. Nov 23 15:15:49 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:53 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:57 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:01 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:05 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Can you help me what I should do? -- *Thanks & Regards,* *Ashwini Kumar | Technology |* www.industrybuying.com | Ghitorni, Delhi - 110030 | India | È*+91-8802137782* | * ashwinikumar(a)industrybuying.com |

2 1

Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) 'dispatcher' Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1811]: db_redis_query(): no columns given to build query keys, falling back to full table scan Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:119]: db_new_result(): allocate 56 bytes for result set at 0x7f0be4273898 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:156]: db_allocate_columns(): allocate 40 bytes for result names at 0x7f0be4273938 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:167]: db_allocate_columns(): allocate 20 bytes for result types at 0x7f0be42739c8 Nov 22 21:15:35 test COPS[26661]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query Given the msg... "no columns given to build query keys, falling back to full table scan", I assume I'm missing a keys modparam.. but I don't know how to build it, and the docs (to me) seem confusing? https://kamailio.org/docs/modules/devel/modules/db_redis.html#db_redis.p.ke… Can anyone help me understand how to build the keys modparam for dispatcher table? Or if the reason for this is different, what do I need to do to avoid the full table scan and the warning in the log? Thanks, Joel.

1 1

Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

1 0

Cannot read an indexed xavp from Kemi-Lua
by Michel Pelletier 22 Nov '19

22 Nov '19

Hi, I am having trouble reading an indexed xavp from Kemi-Lua. The xavp contains 3 entries but I can only access the first one. The xavp is the one filled by the dispatcher module after a call to KSR.dispatcher.ds_select(group, 4) with the flags parameter set to 2. I can get the number of entries using KSR.pvx.xavp_child_get("ds_ctx", "cnt") which returns 3. I can also get the contents of the first element using, for example, KSR.pvx.xavp_child_get("ds_dst", "attrs"). But all my attempts at adding an index value using KSR.pvx.xavp_get() or KSR.pvx.xavp_child_get() return the value 'nil'; Below is the output of KSR.pvx.pv_xavp_print(): +++++ start XAVP list: (nil) ((nil)) (level=0) ----- end XAVP list: (nil) (level=0) +++++ start XAVP list: 0x7f94ce01f1f8 ((nil)) (level=0) *** (l:0 - 0x7f94ce01f1f8) XAVP name: ds_ctx XAVP id: 1762531321 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01f150> +++++ start XAVP list: 0x7f94ce01f150 (0x7f94ce01f218) (level=1) *** (l:1 - 0x7f94ce01f150) XAVP name: cnt XAVP id: 7279890 XAVP value type: 1 XAVP value (int): 3 *** (l:1 - 0x7f94ce01f150) end ----- end XAVP list: 0x7f94ce01f150 (level=1) *** (l:0 - 0x7f94ce01f1f8) end *** (l:0 - 0x7f94ce01eaf8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ea30> +++++ start XAVP list: 0x7f94ce01ea30 (0x7f94ce01eb18) (level=1) *** (l:1 - 0x7f94ce01ea30) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey3;maxload=200 *** (l:1 - 0x7f94ce01ea30) end *** (l:1 - 0x7f94ce01e988) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01e988) end *** (l:1 - 0x7f94ce01e8c8) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01e8c8) end ----- end XAVP list: 0x7f94ce01ea30 (level=1) *** (l:0 - 0x7f94ce01eaf8) end *** (l:0 - 0x7f94ce01edd0) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ed08> +++++ start XAVP list: 0x7f94ce01ed08 (0x7f94ce01edf0) (level=1) *** (l:1 - 0x7f94ce01ed08) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey2;maxload=200 *** (l:1 - 0x7f94ce01ed08) end *** (l:1 - 0x7f94ce01ec60) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ec60) end *** (l:1 - 0x7f94ce01eba0) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01eba0) end ----- end XAVP list: 0x7f94ce01ed08 (level=1) *** (l:0 - 0x7f94ce01edd0) end *** (l:0 - 0x7f94ce01f0a8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01efe0> +++++ start XAVP list: 0x7f94ce01efe0 (0x7f94ce01f0c8) (level=1) *** (l:1 - 0x7f94ce01efe0) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey1;maxload=200 *** (l:1 - 0x7f94ce01efe0) end *** (l:1 - 0x7f94ce01ef38) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ef38) end *** (l:1 - 0x7f94ce01ee78) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01ee78) end ----- end XAVP list: 0x7f94ce01efe0 (level=1) *** (l:0 - 0x7f94ce01f0a8) end ----- end XAVP list: 0x7f94ce01f1f8 (level=0) Regards, Michel Pelletier

1 0

kamailio call loop issue B/W kamailio and freeswitch
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi im using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. ibut the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS -- here is my config file data --- #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** 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_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_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* 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 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### 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 */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.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 "acc.so" loadmodule "counters.so" loadmodule "dispatcher.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 --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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(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) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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")) { 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); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests 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 detection 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($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } 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 processing 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; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing 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; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($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 # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($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_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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 } *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi thank you for reply i have check but same issue ,,, is the config file data plz check if you can help thank you ------------------------------------ #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** 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 block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** 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_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_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* 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 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* 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 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### 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 */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.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 "acc.so" loadmodule "counters.so" loadmodule "dispatcher.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 --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- 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 ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- 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(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) modparam("debugger", "log_level_name", "exec") #!endif ####### 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 { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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")) { 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); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests 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 detection 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($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # 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 statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } 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")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } 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 processing 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; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing 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; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($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 # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($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_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { 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 } ------------------------------------------------------------------------- On Fri, Nov 22, 2019 at 4:57 PM Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > hi thank you for reply > i have check but same issue > i have attached my config file ... can you check if there is any issue > > thank you > > On Fri, Nov 22, 2019 at 4:41 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> Hello, >> >> You’re only pasting the dispatcher route. >> >> You need to follow the script logic. An invite comes in, how does it know >> the call is coming from the client and has to be sent to freeSWITCH? >> Conversely, how does it know it’s coming from freeswitch? >> >> Probably using the permissions module with >> >> https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528655 >> >> You can check if the $si is the freeswitch, then send to clientB >> >> On Fri, 22 Nov 2019 at 11:40, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> there is something wrong in your config file, failing to detect the >>> traffic from freeswitch. Load debugger module and enable cfgtrace logging, >>> then see what actions are executed in the configuration file, maybe you can >>> spot what goes wrong there. >>> >>> Otherwise, to get more help from the mailing lists, the full config is >>> needed. >>> >>> Cheers, >>> Daniel >>> On 22.11.19 09:51, Gaurav Bmotra wrote: >>> >>> hi >>> i m using kamailio 5.3 as register server and using dispatcher i m >>> dendign the calls to freeswitch .. >>> >>> but the issue is then freeswitch send back call to kamailio after >>> processing media ... it should go back to client but it again send call to >>> freeswitch ,, which create the loop and hangup the call with (Temporarily >>> Unavailable (code: 480)) >>> >>> >>> *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 >>> >>> *but I am getting this * >>> >>> client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS >>> >>> >>> *Routing Logic *------------------------------- >>> if(!ds_is_from_list()) { >>> >>> route(DISPATCH); >>> } >>> >>> -------- >>> # Dispatch requests >>> >>> >>> >>> route[DISPATCH] { >>> >>> >>> >>> # round robin dispatching on gateways group '1' >>> >>> >>> >>> if(!ds_select_dst("1", "4")) { >>> >>> >>> send_reply("404", "No destination"); >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: >>> $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> # Try next destionations in failure route >>> >>> >>> >>> failure_route[RTF_DISPATCH] { >>> >>> >>> >>> xnotice("RTF_DISPATCH: $rm $rU [$ci]"); >>> >>> >>> >>> if (t_is_canceled()) { >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> # next DST - only for 500 or local timeout >>> >>> >>> if (t_check_status("(^5)") >>> >>> >>> or (t_branch_timeout() and !t_branch_replied())) >>> { >>> >>> xnotice("RTF_DISPATCH: WARNING $du is broken and marked >>> as inactive"); >>> >>> >>> # ds_mark_dst("ip"); >>> >>> >>> >>> if(ds_next_dst()) { >>> >>> >>> >>> #xdbg("--- SCRIPT: retrying to <$ru> via <$du> >>> (attrs: $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> -- >>> thank you >>> >>> >>> >>> >>> >>> >>> >>> *Regards:* >>> Gaurav Kumar >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> > > > -- > > > > > > > > > *Regards:* > Gaurav Kumar > > -- *Regards:* Gaurav Kumar

1 0

disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi i m using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. but the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS *Routing Logic*------------------------------- if(!ds_is_from_list()) { route(DISPATCH); } -------- # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } -- thank you *Regards:* Gaurav Kumar

3 2

How to retrieve local socket that will be used to send a message from the branch route
by Patrick Wakano 22 Nov '19

22 Nov '19

Hello list, Hope you are all doing well! I am trying to figure out a way to retrieve the local socket the t_relay() has decided to use in a mhomed=1 env. Is there a way to do that? I couldn't find any variable other than the $snd ones that maybe have this info, but I am not using onsend_route anyway..... I am after this value because this is what I can use to figure out if I need to engage RTPEngine with parameters "external, internal", "external, external", "internal, external" or "internal, internal". In my case, my server is in between two networks, and calls can traverse or can be routed back to the same interface depending on what was decide by the LCR module. Signalling is fine, but engaging the RTPEngine must be done with the correct parameters to get the correct IPs in the SDP, other else I get one way audio in the call..... Any help is much appreciated! Thank you! Kind regards, Patrick Wakano

4 7

Adding SIP Gateway Endpoint in Kamailio
by Sujit Roy 21 Nov '19

21 Nov '19

Hello Here is my call flow scenario. SIP Gateway (A) Kamailio (K) Asterisk (AST) SIP Gateway (B) Now i want to send calls from A -> B by using Asterisk as media server. Kamailio shall be used to authenticate A and allow A to send calls to B. What are the configurations i need to make in Kamailio and Asterisk ? Thanks in advance. -- Regards =================== Sujit Roy

3 6

How to access different uris in a header
by Duarte Rocha 21 Nov '19

21 Nov '19

Greetings, In order to identify Redirecting Number and Original Called Number on a given request i look for the top( Redirecting Number) and bottom (Original Called) Diversion headers using hdr(Diversion)[0] and hdr(Diversion)[-1]. However, i have a client that sends me all the Diversions URIS in the same header. Does kamailio has a similar tool for accessing the different URIS in the header? Best Regards,

2 1

CentOS Repo for Kamailio 5.3 - Strange Python23 Dependency?
by Zach Nielsen 21 Nov '19

21 Nov '19

Hi all, Working with some of our Centos-based Kamailio boxes lately has produced a strange result when using the provided yum repos (download.opensuse.org). When trying to yum update the system or install some additional modules, we're getting feedback from yum indicating that it's expecting /usr/bin/python23 --> Finished Dependency Resolution Error: Package: kamailio-5.3.1-3.1.x86_64 (home_kamailio_v5.3.x-rpms) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest Tried my best to do my due diligence here to make sure there was nothing up with our local env that was causing this, but this issue persists across multiple systems, including a new (no frills) one that we just build out. Can I get some confirmation if this is a "me" issue, or if maybe there's an issue with what made it into the repo? Also, just a general question... what's the typical way of you folks deploying Kamailio from a sysadmin perspective? Do you generally compile from source, or do any of you use the provided repositories to manage it? Thanks! Zach

5 6

kamailio Dispatcher error
by Gaurav Bmotra 20 Nov '19

20 Nov '19

hi i want to use dispatcher Module and following is the config.. --------------------------------------- loadmodule "dispatcher.so" -------------- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "ds_hash_size", 2) -------------------------------------------------------------- but when i hit $ kamailio out put: ERROR: bad config file (4 errors) ------------------------------------------------------ logs : kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Wed 2019-11-20 15:16:34 IST; 16s ago Process: 25747 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255/EXCEPTION) Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 5. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Start request repeated too quickly. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Failed with result 'exit-code'. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. ------------------------------------------------------------- plz help me -- Thank you *Regards:* Gaurav Kumar

4 6

Dynamic routing and REGISTER
by Igor Olhovskiy 20 Nov '19

20 Nov '19

Hi! I’m trying to get `drouting` module working for REGISTER. But seems it’s not passing one of internal checks, cause in REGISTER R-URI does not contains username and I’m getting errors drouting [drouting.c:567]: build_ruri(): stripping 0 makes username <> null drouting [drouting.c:970]: do_routing(): failed to build ruri So, as I got, this module is mainly for calls (means INVITE) routing, rather than to use it like one more dispatcher, or it’s also a way to handle this part here? Thanks, Igor

3 5

Query regarding Kamailio Sip Server for IMS supporting VOLTE
by Shubhendu Das 20 Nov '19

20 Nov '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

1 0

How to intercept an INVITE after (ts_append)
by Oleg Podguyko 20 Nov '19

20 Nov '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using rtpgngine module I proxy media stream via rtpengine(rtpengine_manage()). I change SDP. SiP2SIP calls work fine. Now I would like to have possibility to "wake up" sip-clients(at the android and ios phones) which are usually not on-line via push notification. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { ts_store(); # if it a new transuction ,it will be store for some time. } } switch ($var(rc)) { case 1: route(RELAY); break; case -1: break; case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu"); # if transuction for this sip-client is exist. The invite will be send. exit; } If the client is not registered at the kamailio during the call. Then such calls are stored through the "ts_store" function for some time. And if during this time the client send register, kamailio will be sent INVITE to it. But I have the problem. Such calls after "wake up"go through a special branch route that I don't know how to intercept. As a result, I can't change SDP(using rtpengite_manage commands) and rtp stream do not proxy via rtpengine… and call failed. /usr/sbin/kamailio[9957]: exec: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=428 a=26 n=ts_append /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tsilo [ts_append.c:72]: ts_append(): transaction 36343:92182831 found for 777777777, going to append branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:255]: lookup_helper(): contact for [777777777] found by address /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:456]: lookup_helper(): instance is /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:77]: t_append_branches(): transaction 36343:92182831 in status 0 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:99]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (1) outgoing branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:77777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/msg_translator.c:2933]: create_via_hf(): id added: <;i=1>, rcv proto=2 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:777777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba] with ruid [uloc-5dd2faa0-26e5-1] /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:163]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (0) outgoing branches after clear_branches() /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (9957 (22)), reusing fd /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... -- Oleg Podguyko

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