- sr-users - kamailio.org

Kamailio:Contact validation during INVITE
by Prashant Desai 25 Nov '15

25 Nov '15

Hi, We observe that the following kamailio API is trying to find out user registration by comparing source IP and port with stored contact, if it fails, then it compares based on incoming SIP msg contact uri. is_registered(struct sip_msg* _m, udomain_t* _d){ getContactP(_m,_d); } pcontact_t * getContactP(struct sip_msg* _m, udomain_t* _d) {); //Getting pcontact from source IP and port if (ul.get_pcontact_by_src(_d, &received_host, _m->rcv.src_port, _m->rcv.proto, &c) == 1) { LM_DBG("No entry in usrloc for %.*s:%i (Proto %i) found!\n", received_host.len, received_host.s, _m->rcv.src_port, _m->rcv.proto); } if (c == NULL) { //Getting pcontact from contact uri. if (ul.get_pcontact(_d, &ct->uri, &c) == 0) { } } We have use case where INVITE message is from same source IP and port but having unregistered contact.With this scenario,is_registered() will return success. We would like to know reason behind comparing the source IP and port to fetch the pcontact? Regards, Prashant

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Releasing Kamailio v4.3.4
by Daniel-Constantin Mierla 25 Nov '15

25 Nov '15

Hello, I am considering to release Kamailio v4.3.4 next week on Wed, Nov 25 or the following day. As usual, announce any issues that you are aware and not reported to sr-dev or bug tracker. Also, if some patch was not backported and you think it has to be, write to sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 1

Build error on Solaris 11 with Solaris Studio
by Spencer Thomason 25 Nov '15

25 Nov '15

Hello, I realize its probably uncharted waters but I was hoping someone could point me in the right direction… I attempting to build Kamailio on a UltraSPARC T2 box running Solaris 11.3 with Sun cc and I’m not getting very far: $ CC=cc gmake all CC (cc) [kamailio] timer_proc.o cc: Warning: -xarch=v9 is deprecated, use -m64 to create 64-bit programs "timer_funcs.h", line 111: warning: argument mismatch "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "timer_funcs.h", line 111: syntax error before or at: __func__ "cfg/../atomic/atomic_unknown.h", line 59: #warning: no native memory barrier implementations, falling back to slow lock based workarround "cfg/../lock_alloc.h", line 72: warning: argument mismatch "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "cfg/../lock_alloc.h", line 72: syntax error before or at: __func__ "ut.h", line 236: warning: argument mismatch "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 236: syntax error before or at: __func__ "ut.h", line 240: warning: argument mismatch Any help would be greatly appreciated! The contents of config.mak are below: # this file is autogenerated by make cfg MAIN_NAME= kamailio CFG_NAME= kamailio SCR_NAME= sip-router FLAVOUR= kamailio INSTALL_FLAVOUR= kamailio SRC_NAME= sip-router RELEASE= 4.3.3 OS= solaris ARCH= sparc64 C_DEFS= -DNAME='"kamailio"' -DVERSION='"4.3.3"' -DARCH='"sparc64"' -DOS='solaris_' -DOS_QUOTED='"solaris"' -DCOMPILER='"cc 5.1"' -D__CPU_sparc64 -D__OS_solaris -DSER_VER=4003003 -DCFG_DIR='"/usr/local/etc/kamailio/"' -DRUN_DIR='"/var/run/kamailio/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DUSE_DNS_CACHE -DUSE_DNS_FAILOVER -DUSE_DST_BLACKLIST -DUSE_NAPTR -DWITH_XAVP -DDBG_QM_MALLOC -DMEM_JOIN_FREE -DUSE_TLS -DTLS_HOOKS -DUSE_CORE_STATS -DSTATISTICS -DMALLOC_STATS -DWITH_AS_SUPPORT -DUSE_SCTP -DSPARC64_MODE -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DUSE_PTHREAD_MUTEX -DHAVE_DEVPOLL -DHAVE_SELECT -DHAVE_FILIO_H DEFS_RM= PROFILE= CC= cc LD= cc MKDEP= cc -xM1 MKTAGS= ctags LDFLAGS= -xO5 C_INCLUDES= CC_MKDEP_OPTS= MOD_LDFLAGS= -G -xO5 LIB_LDFLAGS= -G -xO5 UTILS_LDFLAGS= -xO5 LIB_SONAME= "-h " LD_RPATH= -"-R " LIB_SUFFIX= .so LIB_PREFIX= lib LIBS= -lfast -ldl -lresolv -L/usr/local/lib -lxnet -lnsl -lrt LEX= flex YACC= yacc YACC_FLAGS= -d -b cfg PREFIX= /usr/local LOCALBASE= /usr/local TAR= gtar INSTALL= ginstall INSTALL_CFG= ginstall -m 644 INSTALL_BIN= ginstall -m 755 INSTALL_MODULES= ginstall -m 755 INSTALL_DOC= ginstall -m 644 INSTALL_MAN= ginstall -m 644 INSTALL_LIB= ginstall -m 755 INSTALL_TOUCH= touch INSTALL_SHARE= ginstall -m 644 Q= 1 CC_EXTRA_OPTS= CPU= CFLAGS_RM= CFLAGS= -m64 -g -xO5 -fast -native -xarch=v9 -xCC -xc99 $(CC_EXTRA_OPTS) MOD_CFLAGS= -xcode=pic32 $(CFLAGS) LIB_CFLAGS= -xcode=pic32 $(CFLAGS) UTILS_CFLAGS= $(CFLAGS) BASEDIR= $(DESTDIR) basedir= $(BASEDIR) DESTDIR= LIBDIR= lib64 RUNBASEDIR= $(DESTDIR) runbasedir= $(RUNBASEDIR) PREFIX= $(LOCALBASE) prefix= $(PREFIX) cfg_prefix= $(basedir)$(prefix) cfg_dir= etc/$(MAIN_NAME)/ bin_prefix= $(basedir)$(prefix) bin_dir= sbin/ modules_prefix= $(basedir)$(prefix) modules_dir= $(LIBDIR)/$(MAIN_NAME)/ doc_prefix= $(basedir)$(prefix) doc_dir= doc/$(MAIN_NAME)/ man_prefix= $(basedir)$(prefix) man_dir= man/ ut_prefix= $(basedir)$(prefix) ut_dir= share_prefix= $(basedir)$(prefix) share_dir= share/$(MAIN_NAME)/ lib_prefix= $(basedir)$(prefix) lib_dir= $(LIBDIR)/$(MAIN_NAME)/ data_prefix= $(basedir)$(prefix) data_dir= $(MAIN_NAME)/ cfg_target= $(prefix)/$(cfg_dir) lib_target= $(prefix)/$(lib_dir) data_target= $(prefix)/$(data_dir) run_prefix= $(runbasedir) run_dir= var/run/$(MAIN_NAME)/ run_target= $(run_prefix)/$(run_dir) override makefile_defs:=1 C_DEFS:=$(filter-out $(DEFS_RM) $(extra_defs),$(C_DEFS)) $(extra_defs) CFLAGS:=$(filter-out $(CFLAGS_RM) $(CC_EXTRA_OPTS), $(CFLAGS)) $(CC_EXTRA_OPTS) Thanks! Spencer

1 0

Grab users password from WWW-Auth header
by Alexandru Covalschi 24 Nov '15

24 Nov '15

Hello! My problem is I need to do users authentication through API. So I need to replace if (!www_authenticate("$td", "subscriber")) { www_challenge("$td", "1"); } With if (!my_auth_script()) { www_challenge("$td", "1"); } The main problem is - how can I grab or compare users password? I know nonce, which I understand is MD5 salt. Can I, for example, grab users password from API, then grab the MD5 string and the nonce user sent me, calculate MD5 on base of API password and nonce - and then compare MD5 strings sent by user and calculated? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

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Issue with SNMPstats
by Igor Potjevlesch 24 Nov '15

24 Nov '15

Hello, I have configured the SNMPstats module fine in my machine because I get the SNMP information with the new KAMAILIO MIBS but I don't understand why the value of kamailio dialogs is always 0: KAMAILIO-MIB::kamailioCurNumDialogs.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInProgress.0 = Gauge32: 0 KAMAILIO-MIB::kamailioCurNumDialogsInSetup.0 = Gauge32: 0 KAMAILIO-MIB::kamailioTotalNumFailedDialogSetups.0 = Counter32: 0 However I can see the transactions: KAMAILIO-SIP-COMMON-MIB::kamailioSIPSummaryTotalTransactions.0 = Counter32: 54797 Should I configure an specific parametr of the SNMPstats module? Regards,

3 6

MESSAGE is not successfull in Kamailio IMS
by Thierry Luo 24 Nov '15

24 Nov '15

Hi All, I installed kamailio on Ubuntu as IMS pCSCF/iCSCF/sCSCF. REGISTER is fine, but when I tried to send a MESSAGE (SMS) from one user to another, it was always not successful. I captured IP packets on the server, and it seemed that the error was caused on pCSCF. User1 and User2 are both behind their NAT network. Suppose User1's private IP is Private_IP1, and User1's public IP (on router) is Public_IP1, and User2's private and public IPs are Private_IP2 and Public_IP2: User1 pCSCF iCSCF sCSCF user2 MESSAGE ----------------> MESSAGE ---------------------> MESSAGE -------------------> MESSAGE <----------------------------------------------------- MESSAGE ---------------------------------------------------------------------------- ----> Unfortunately, from the captured data on the server, in the last step, the MESSAGE was sent to the Private_IP2 of User2. It should be sent to the Public_IP2. Of course, the datagram with the destination of Private_IP2 cannot reach User2 successfully. Please help analyze why the pCSCF sent the MESSAGE packet to User2's private IP instead of public IP. The configuration file was attached for reference. BR. Thierry Luo

2 6

Simple User Agent change
by Steven Maddox 24 Nov '15

24 Nov '15

Hi, Sorry to ask something probably very simple but I'm difficulty finding the right kinds of resources to learn from which are succinct enough to not overload my head. We have a PBX which we don't control (and hate) which expects only certain makes/models phones to register (it checks for user agents). I have a phone which unfortunately doesn't support tweaking its user agent, so I'd like to pass it through an extremely SIP proxy to have it's user agent tweaked and all other data to/from the PBX transparently let through. I've managed to knock up something (that isn't a proxy) using Asterisk where I made a trunk and a peer... but obviously since this isn't a proxy... things like BLF and transferring calls will never work. If anyone can point me in the direction of a really simple set up to achieve this, please let me know :) -- Steven Maddox

2 2

kazoo module documentation
by Vik Killa 24 Nov '15

24 Nov '15

Hello, I'm testing kazoo module for use with RabbitMQ I noticed there is a required parameter that is not listed in the documentation: http://kamailio.org/docs/modules/devel/modules/kazoo modparam("kazoo", "pua_mode", 0) The above parameter is required but not documented. Can someone please update the documentation? Thank you. /V

2 1

Problem with resuming suspended transactions.
by Dmytro Bogovych 23 Nov '15

23 Nov '15

Greetings. We build specialized softphone for iOS. Push notifications are used to notify user and wake up softphone app on missed calls and messages. We used Daniel's example config script as base for own configuration file. It has the same parts and does not differ too much. I talk about this example: http://www.kamailio.org/events/2014-KamailioWorld/day2/26-Daniel-Constantin… Everything was ok for few monthes - with kamailio 4.1.2. Few weeks ago we upgraded to 4.3.3. Now paused calls cannot be resumed properly. Softphone receives 500 Server error (No error 0/SL) in response This is problem part of script: route[RELAYRESUMED] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if ($var(rr_target) != "") { xlog("L_INFO","Relay to $var(rr_ip):$var(rr_port)\n"); $ru = "sip:" + $rU + "@" + $var(rr_ip) + ":" + $var(rr_port) + ";transport=" + $var(rr_proto); xlog("L_INFO","Current request URI is $ru\n"); if (!t_relay()) xlog("L_INFO","Cannot relay PUSHED invite."); sl_reply_error(); } else { xlog("L_WARN","Target proto is not set\n"); sl_reply_error(); } exit; } It produces log ----- Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: resuming trasaction [23927:345343979] 1985987629 (23927:345343979) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Relay to 95.133.110.17:11791 Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Current request URI is sip:1985987629@95.133.110.17:11791;transport=tls Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: No error (0/SL) Nov 23 14:42:24 ctdsip3 /usr/local/sbin/kamailio[32141]: INFO: <script>: Checking FAILED messages for 1985987629: 0 ----- sl_reply_error() sends 500 with No error (0/SL) description. Later 180 & 200 are sent. But client softphone terminates transaction already (as it handles 500 response code) . There is no "Cannot relay PUSHED invite." message in the log however. I will be happy to heary any idea what can be cause... Thank you!

2 3

Kamailio-SEMS Integration
by Prashant Desai 23 Nov '15

23 Nov '15

Hi, We are working on kamailio-4.2.3 and we require B2BUA functionality to be achieved.From kamailio official website we got to know,we can integrate kamailio with SEMS to achieve B2BUA functionality. We have few queries regarding this integration. 1.SEMS-1.6.0(latest version available) is been used,is this version suits kamailio-4.2.3? 2.SEMS-1.6.0 does not support unix based communication with kamailio-4.2.3,as SEMS-1.6.0 doesn't have unix socket interface plugin. Since we need unix socket based communication with SEMS.can we port the unix socket interface plugin to SEMS-1.6.0?If no,please suggest us better approaches. Regards, Prashant.

2 2

Kamailio, rtpproxy, websocket and 488 error
by Alfredo Pastor 23 Nov '15

23 Nov '15

Hello, I was doing some tests with Elastix MT that use version 4.1.6 of Kamailio and by default uses rtpproxy, I had the problem of error 488 and unable to make calls from websockets (Incompatible SDP). I could see that Peter Dunkley in a video explains that the problem is solved by rewriting the SDP, but he uses the mediaproxy not rtpproxy. What are your recommendations? I hope you can help me. Chreers, Alfredo.

1 0

rtpengine via-branch=extra question
by Juha Heinanen 22 Nov '15

22 Nov '15

rtpengine readme tells: via-branch=... - Include the “branch” value of one of the “Via” headers in the request to the RTP proxy. Possible values are: “1” - use the first “Via” header; “2” - use the second “Via” header; “auto” - use the first “Via” header if this is a request, or the second one if this is a reply; “extra” - don't take the value from a header, but instead use the value of the “extra_id_pv” variable. This can be used to create one media session per branch on the RTP proxy. When sending a subsequent “delete” command to the RTP proxy, you can then stop just the session for a specific branch when passing the flag '1' or '2' in the “rtpengine_delete”, or stop all sessions for a call when not passing one of those two flags there. if via-branch=extra is used in offers of branched, should it also be used in answers and deletes? -- juha

2 6

Functionality of rtpproxy
by kai.ohnacker＠cbc.de 20 Nov '15

20 Nov '15

Hello, I have installed a kamailio server with rtpproxy. The server is located between 2 networks and bridge these networks. My question: Does handle rtpproxy only 1 call at the same time? Because my configuration works, but only for one call from the first to the second network. The second call is not working. But if I make internal or external call, I can do more than one call. Do I need the multiple rtpproxy usage configuration? I thought not, because I don't use different IPs for the second call... Cheers, Kai

2 1

403 Forbidden for INVITE in kamailio for ims call
by Priyaranjan Nayak 20 Nov '15

20 Nov '15

Hi All, I have installed kamailio and using as a i-cscf,s-cscf and p-cscf server. I have taken FHoSS(OpenImscore) for hss server Now I am able to do register and unregister. When I am sending INVITE for call, the calling-party and called-party were registerd but I am getting "403 Forbidden - Dialog not found on S-CSCF or Terminating user not suitable for unregistered services". Could you please tell me what is the root cause of above error message ? Best regards, Priyaranjan

1 0

RFC: rtpproxy and a=nortpproxy:yes
by Daniel-Constantin Mierla 19 Nov '15

19 Nov '15

Hello, I am looking to get some feedback on some issues I noticed more and more lately. Apparently some SIP devices (media servers, phones, ...) are keeping the "a=nortpproxy:yes" line in SDP when replying to an INVITE that contains such line. [Alice] ------> [Kamailio+RTPProxy] ------> [Bob] The 200ok response from Bob has "a=nortpproxy:yes" in SDP. By default, that line in SDP makes the rtpproxy not to engage itself anymore in rtp relaying, and as a result things like no audio or one way audio happens. Anyone else encountering such situations? If yes, what are the devices with such behaviour? So far I noticed with some FreeSwitch and Snom -- none of them I can control, so there might be a specific configuration of those devices, not something by default there. The solution is to set: modparam("rtpproxy", "nortpproxy_str", "") and use flag 'r' for rtpproxy_manage() if the IP in SDP is not a private address. I already updated the default config for master to use flag 'r' if the SDP media IP is not private, wondering if nortpproxy_str should be set to empty in kamailio.cfg (or made empty as default in config). Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

1 0

rtpproxy database structure for kamailio 4.2.3
by Jan Hazenberg 19 Nov '15

19 Nov '15

Hi all, I'm trying to load the rtpproxy url's from database but get a error when starting kamailio (4.2.3): ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Unknown column 'set_name' in 'field list' Looks like the field set_name is missing in the database structure, but i can't find it in the docs. If i add the field kamailio starts, but what is the field set_name used for. Seems it needs to be a int value: ERROR: rtpproxy [rtpproxy.c:528]: get_rtpp_set(): Invalid set name 'test' - must be integer Thanks, Jan

2 2

Fw: Kamailio with Siremis
by Ryan Holbein 19 Nov '15

19 Nov '15

________________________________ From: Ryan Holbein <rtholbein(a)hotmail.com> Sent: Wednesday, November 18, 2015 8:07 AM To: miconda(a)gmail.com Subject: Re: [SR-Users] Kamailio with Siremis Hello, This is what I have setup so far I know it is not much but looking for help this is my very first Proxy. If you look at my Attachments I am trying to do IP auth. I have 2 Asterisk Boxs with 192.168.89.2 and .3.. The Proxy has 2 nics the outside and the 192.168.89.0/24 ... How do I setup the LCR rule if you look at the LCR Attachment? Also is my trusted permissions setup correct(Those IPs you see are correct)? If you are willing to help Thank you in Adv. ________________________________ From: Daniel-Constantin Mierla <miconda(a)gmail.com> Sent: Thursday, November 12, 2015 12:05 PM To: Ryan Holbein; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis If you have a single Asterisk box, then you can simply forward using kamailio.cfg only. Look at route[PSTN] in default configuration file for an example. If you have more Asterisk boxes, then depends on what you need/want to do: - for load balancing look at dispatcher module - it has a complete sample config in the docs - for least cost routing, then lcr module is indeed an option Cheers, Daniel On 12/11/15 17:56, Ryan Holbein wrote: Now I assume in LCR I will add the Asterisk boxs so it can route? Does that sound correct? Anyone able to provide an example? Thank you ________________________________ From: sr-users <sr-users-bounces(a)lists.sip-router.org><mailto:sr-users-bounces@lists.sip-router.org> on behalf of Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Sent: Thursday, November 12, 2015 3:01 AM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio with Siremis Hello, look at permissions module. Default kamailio can do IP auth by enabling WITH_AUTH and WITH_IPAUTH and adding the trusted IP in database table 'address' with group id 1. Siremis has a panel for managing the records in address table. Cheers, Daniel On 11/11/15 14:10, Ryan Holbein wrote: Hello, I have setup with my phone provider an IP endpoint to do IP Auth. How would I go about setting that up in Kamailio or even better can some tell me how to do that in Siremis? _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda<http://twitter.com/#%21/miconda> - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com Kamailio Advanced Training, Nov 30-Dec 2, Berlin - http://asipto.com/kat

2 2

ndb_mongodb, db_mongodb and replicasets
by Kelvin Chua 19 Nov '15

19 Nov '15

anybody ever tried using a mongodb url with replicasets and parameters to work with kamailio? Kelvin Chua

2 11

Content-Type
by Safdar Khan 19 Nov '15

19 Nov '15

How to check content-type of ongoing various type of media streams in kamailio?

2 1

siptrace segfault - attack?
by Marian Piater 18 Nov '15

18 Nov '15

Hello, we had segfault today. Kamailio has been running for few months without problems. Nov 18 10:51:20 sbc kernel: [11326028.926502] kamailio[14452]: segfault at 30 ip 00007f20f7f3838a sp 00007ffef3ab7b10 error 4 in siptrace.so[7f20f7f23000+27000] GDB: Reading symbols from kamailio...done. [New LWP 14452] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kam'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 1041 sto.totag = get_to(msg)->tag_value; (gdb) bt #0 0x00007f20f7f3838a in sip_trace (msg=0x7f2100f96750, dst=0x0, dir=0x0) at siptrace.c:1041 #1 0x0000000000457ca9 in do_action (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1053 #2 0x0000000000463cb9 in run_actions (h=0x7ffef3ab8320, a=0x7f2100b8a928, msg=0x7f2100f96750) at action.c:1548 #3 0x00000000004643bf in run_top_route (a=0x7f2100b8a928, msg=0x7f2100f96750, c=0x0) at action.c:1634 #4 0x0000000000573ea7 in receive_msg ( buf=0x9c9400 <buf> "OPTIONS sip:100@XXX.XXX.XX.XX SIP/2.0\r\nv: SIP/2.0/UDP 69.64.39.119:5060;branch=z9hG4bK-82135822;rport\r\nContent-Length: 0\r\nf: \"MisterX\"<sip:100@1.1.1.1>;tag=61326665333131663133633401333733313630343335"..., len=357, rcv_info=0x7ffef3ab85b0) at receive.c:196 #5 0x0000000000493ff5 in udp_rcv_loop () at udp_server.c:495 #6 0x000000000051fdd7 in main_loop () at main.c:1573 #7 0x0000000000525b6b in main (argc=13, argv=0x7ffef3ab8998) at main.c:2533 (gdb) quit Our IP is hidden, but there is 69.64.39.119, which is foreign address and I think, it was attack. Unfortunatelly I don't have SIP packet details, but you can see 1.1.1.1 or strange tag or content-length=0 in received message. I looked into siptrace.c and there is function sip_trace_prepare where is get_from(msg) checked, but no get_to(msg). This function is run from main sip_trace function. So I think, we need also check get_to(msg) function. I just disabled siptrace module, but we need it. Thank you. Marian

1 1

Stop msrp relay from kamailio to freeswitch
by Safdar Khan 18 Nov '15

18 Nov '15

Hi folks I have integrated kamailio and freeswitch. The basic register and authentication is happening through kamailio and other media services through freeswitch like file sharing (using msrp). My question is how can I stop file sharing through msrp via freeswitch since everything going from kamailio to freeswitch to client. I just wanted to share file only through kamailio. I am attaching my cfg file below... #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_FREESWITCH #!define WITH_PRESENCE #!define WITH_TLS #!define WITH_MSRP #!define WITH_XHTTP #!define WITH_MSILO #!define WITH_NAT #!substdef "!MY_IP_ADDR!192.168.5.99!g" #!substdef "!MY_DOMAIN!<SERVER_IP>!g" #!substdef "!MY_WS_PORT!9080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!define WITH_WEBSOCKETS # Kamailio (OpenSER) SIP Server v4.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - 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 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 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:127.0.0.1:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 ####### 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 #!ifdef WITH_FREESWITCH freeswitch.bindip = "192.168.5.99" desc "FreeSWITCH IP Address" freeswitch.bindport = "5090" desc "FreeSWITCH Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/local/kamailio-4.3/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so" #!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 #!ifdef WITH_MSRP loadmodule "msrp.so" #!endif loadmodule "rtimer.so" loadmodule "sqlops.so" loadmodule "cfgutils.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" ##loadmodule "nathelper.so" #!endif #!ifdef WITH_MSILO loadmodule "msilo.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 1) # 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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- #modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.5.99:5060") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "NULL") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio-4.3/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/usr/local/kamailio-4.3/etc/kamailio/privkey.pem") modparam("tls", "certificate", "/usr/local/kamailio-4.3/etc/kamailio/kamailio1_cert.pem") modparam("tls", "ca_list", "/usr/local/kamailio-4.3/etc/kamailio/calist.pem") modparam("tls", "verify_certificate", 1) modparam("tls", "require_certificate", 1) modparam("tls", "crl", "/usr/local/kamailio-4.3/etc/kamailio/my_crl.pem") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "msrp=>size=8;autoexpire=MSRP_MAX_EXPIRES;") #!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 #!ifdef WITH_MSRP # ----- msrp params ----- modparam("msrp", "sipmsg", 1) modparam("msrp", "cmap_size", 8) modparam("msrp", "use_path_addr", "msrp://192.168.5.99:5060") #!endif modparam("rtimer", "timer", "name=tst;interval=300;mode=1;") modparam("rtimer", "exec", "timer=tst;route=8") modparam("sqlops","sqlcon","ca=>mysql://kamailio:kamailiorw@localhost/kamailio") #!ifdef WITH_AUTH # ----- auth params ----- #modparam("auth", "nonce_count", 1) #modparam("auth", "qop", "") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif #!ifdef WITH_MSILO # ------ msilo params ------ modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@192.68.5.98") modparam("msilo","contact_hdr","Contact: <sip:registrar@192.168.5.99>\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","") modparam("msilo", "check_time", 200000) modparam("msilo", "expire_time", 259200) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { 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); #!ifdef WITH_FREESWITCH # save callee ID $avp(callee) = $rU; route(FSDISPATCH); #!endif # user location service route(LOCATION); route(RELAY); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE_MESSAGE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } if($ua =~ "friendly-scanner") { sl_send_reply("200", "OK"); exit; } #!endif 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("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. 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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); ## #!ifdef WITH_MSILO ### m_dump(); ### #!endif exit; } } # USER location service route[LOCATION] { xlog("going inside LOCATION\n"); #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_MSILO xlog("going inside ifdefMSILO\n"); #if(is_method("MESSAGE")) # { xlog("going inside MSILO\n"); route(MSILO); # exit; #} #!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; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif #!ifdef WITH_MSILO if(is_method("MESSAGE")){ return; } #!ifdef WITH_FREESWITCH if(route(FSINBOUND)) return; #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[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 singaling 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; rtpproxy_manage("co"); 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($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } #!ifdef WITH_FREESWITCH # FreeSWITCH routing blocks route[FSINBOUND] { if($si== $sel(cfg_get.freeswitch.bindip) && $sp==$sel(cfg_get.freeswitch.bindport)) return 1; return -1; } route[FSDISPATCH] { if(!is_method("INVITE")) return; if(route(FSINBOUND)) return; # dial number selection switch($rU) { case /"^41$": # 41 - voicebox menu # allow only authenticated users if($au==$null) { sl_send_reply("403", "Not allowed"); exit; } $rU = "vm-" + $au; break; case /"^441[0-9][0-9]$": # starting with 44 folowed by 1XY - direct call to voice box strip(2); route(FSVBOX); break; case /"^433[01][0-9][0-9]$": # starting with 433 folowed by (0|1)XY - conference strip(2); break; case /"^45[0-9]+$": strip(2); break; default: # offline - send to voicebox if (!registered("location")) { route(FSVBOX); exit; } # online - do bridging prefix("kb-"); if(is_method("INVITE")) { # in case of failure - re-route to FreeSWITCH VoiceMail t_on_failure("FAIL_FSVBOX"); } } route(FSRELAY); exit; } route[FSVBOX] { if(!($rU=~"^1[0-9][0-9]+$")) return; prefix("vb-"); route(FSRELAY); } # Send to FreeSWITCH route[FSRELAY] { $du = "sip:" + $sel(cfg_get.freeswitch.bindip) + ":" + $sel(cfg_get.freeswitch.bindport); route(RELAY); exit; } #!endif # manage outgoing branches # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_MSILO #manage failure message failure_route[MANAGE_FAILURE_MESSAGE] { if (!method=="MESSAGE") { exit; } log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { t_reply("202", "Accepted"); }else{ t_reply("503", "Service Unavailable"); } } #!endif # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_FREESWITCH failure_route[FAIL_FSVBOX] { #!ifdef WITH_NAT if (is_method("INVITE") && (isbflagset(FLB_NATB) || isflagset(FLT_NATS))) { unforce_rtp_proxy(); } #!endif if (t_is_canceled()) { exit; } if (t_check_status("486|408")) { # re-route to FreeSWITCH VoiceMail $rU = $avp(callee); route(FSVBOX); } } #!endif route[8] { sql_query("ca", "insert into statistics (time_stamp,random,shm_used_size,shm_real_used_size, shm_max_used_size,shm_free_used_size,ul_users,ul_contacts) values ($Ts, $RANDOM,$stat(used_size),$stat(real_used_size),$stat(max_used_size), $stat(free_size),$stat(location-users),$stat(location-contacts))", "ra"); } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0", "$msrp(method)")) { if(auth_get_www_authenticate("myrealm", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } msrp_cmap_save(); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } if (msrp_cmap_lookup()) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_relay(); } else { msrp_reply("481", "Session-does-not-exist"); } } else { msrp_reply("501", "Request-method-not-understood"); } } #--------------websockets starts here------------- #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; exit; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "http://communicator.MY_DOMAIN" # && $hdr(Origin) != "https://communicator.MY_DOMAIN") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif #--------------websockets ends here------------- #---------------msilo starts here------------------ route[MSILO]{ #!ifdef WITH_MSILO if(!is_method("MESSAGE")) return; ##..........added...... if(lookup("location")) { if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; }; if(!lookup("location")) { $var(rc)=$rc; if (! t_newtran()) { sl_reply_error(); exit; }; switch ($var(rc)) { case -1: case -3: if (m_store("$ru")) { if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; case -2: t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; } }; t_on_failure("MANAGE_FAILURE_MESSAGE"); t_relay(); exit; #!endif return; } #------------------msilo ends here----------------

1 0

Kamailo crashes with auth_radius module
by Volkan Oransoy 18 Nov '15

18 Nov '15

Hi, I try to enable auth_radius module with my kamailo box. But it crashes on the first REGISTER request. I have enabled it with this two lines. loadmodule "auth_radius.so" modparam("auth_radius", "radius_config", "/etc/radiusclient/radiusclient.conf") This is the authentication part of my script. if (is_method("REGISTER")) { if (!radius_www_authorize("$td")) { switch ($rc) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; } And in the /etc/radiusclient/dictionary file, I included sip and kamailio dictionaries. $INCLUDE dictionary.sip $INCLUDE dictionary.kamailio Service succesfully starts but it crashes on the first REGISTER request. Nov 18 11:30:27 sip kernel: [159645.360105] traps: kamailio[6941] general protection ip:7f89b1660920 sp:7ffc8d3bac78 error:0 in libc-2.19.so [7f89b15cb000+19f000] Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:728]: handle_sigs(): child process 6941 exited by a signal 11 Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: ALERT: <core> [main.c:731]: handle_sigs(): core was generated Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD Nov 18 11:30:27 sip /usr/sbin/kamailio[6961]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6959]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6951]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6936]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6948]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6944]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6937]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6953]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6945]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6938]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6955]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6943]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6942]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6957]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6939]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6940]: INFO: <core> [main.c:794]: sig_usr(): signal 15 received Nov 18 11:30:27 sip /usr/sbin/kamailio[6934]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Thanks,

2 3

Authorization selects
by Alexandru Covalschi 17 Nov '15

17 Nov '15

Hello everyone! I need to extract values from authentication header, but 408. @authorization["string"] 409. @authorization["string"].username 410. @authorization["string"].username.user 411. @authorization["string"].username.domain 412. @authorization["string"].realm 413. @authorization["string"].nonce 414. @authorization["string"].uri 415. @authorization["string"].cnonce 416. @authorization["string"].nc 417. @authorization["string"].response 418. @authorization["string"].opaque 419. @authorization["string"].algorithm 420. @authorization["string"].qop these values doesn't seem to work for me. Used https://github.com/kamailio/kamailio/blob/master/doc/select_list/select_cor… doc. $var(z) = @authorization["string"].realm; xlog("Realm $var(z)"); Using that simple check I get nothing in $var(z) version: kamailio 4.3.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.9.2 Can anyone check that please? Or maybe syntax is wrong and I need newer selects doc? -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 3

Kamailio IMS - VM
by Bruno Cruz 17 Nov '15

17 Nov '15

Hello everyone, My name is Bruno, I'm a student from Portugal studying IMS at the moment. I've been following tutorials online, trying to get Kamailio to work as an IMS core - I'm pretty much a newbie in this field. Unfortunantly I've been getting several errors and I can't seem to properly configure everything. I was wondering if there is a All-in-One VM that can be downloaded, with Kamailio ready to be used as an IMS core? Thanks in advance. Best regards, Bruno C.

4 8

General question about mohqueue
by David Villasmil Govea 16 Nov '15

16 Nov '15

Hello list, I've been given a task to setup a predictive dialer with for 8000+ agents taking accepted calls in a queue. I've done this for a much smaller setup in freeSWITCH but i recently saw that kamailio has a mohqueue (which i already setup as a PoC) that might work for this. My question is whether you think it's a good idea to go with this on kamailio or is freeSWITCH better suited for this kind of application? The setup would be to generate outgoing calls based on a list (predictive dialer) and send answered calls to a queue to be answered by an agent. Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

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