- sr-users - kamailio.org

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

Kamailio 4.3 and Debian 8.2 systemctl
by Grant Bagdasarian 16 Nov '15

16 Nov '15

Hello, I've just installed Kamailio 4.3 on Debian 8.2, but starting kamailio does not work since Debian requires systemctl to be setup. I've tried the below config but it didn't work (https://github.com/kamailio/kamailio/issues/294#issuecomment-138590939) usr/local/src/kamailio# systemctl status kamailio.service â kamailio.service - Kamailio SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled) Active: failed (Result: resources) systemd[1]: [/etc/systemd/system/kamailio.service:14] Not an absolute path, ignoring: $PIDFILE systemd[1]: Failed to load environment files: No such file or directory systemd[1]: kamailio.service failed to run 'start' task: No such file or directory systemd[1]: Failed to start Kamailio SIP Server. systemd[1]: Unit kamailio.service entered failed state. [Unit] Description=Kamailio (OpenSER) - the Open Source SIP Server After=syslog.target network.target [Service] Type=forking Environment='PIDFILE=/var/run/kamailio/kamailio.pid' Environment='CFGFILE=/etc/kamailio/kamailio.cfg' Environment='SHM_MEMORY=64' Environment='PKG_MEMORY=8' Environment='USER=kamailio' Environment='GROUP=kamailio' EnvironmentFile=/etc/default/kamailio PIDFile=$PIDFILE # ExecStart requires a full absolute path ExecStart=/usr/sbin/kamailio -P $PIDFILE -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP ExecStopPost=/bin/rm -f $PIDFILE Restart=on-abort [Install] WantedBy=multi-user.target Does anyone have a working kamailio.service file? Regards, Grant

4 3

TLS module not initialized in 4.3.3, worked in 4.1.5
by Sebastian Damm 16 Nov '15

16 Nov '15

Hello, we just updated one kamailio server from 4.1.5 to 4.3.3, and although the config file is correct and kamailio starts up, it doesn't initialize TLS and says " tls support enabled, but no tls engine available (forgot to load the tls module?)" In the log I see: Old shutdown (last lines): Nov 13 11:44:38 lasola /usr/sbin/kamailio[15890]: DEBUG: <core> [mem/shm_mem.c:235]: shm_mem_destroy(): destroying the shared memory lock Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:271]: tcp_read_data(): error reading: Connection reset by peer (104) Nov 13 11:44:41 lasola /usr/sbin/kamailio[14818]: ERROR: <core> [tcp_read.c:1296]: tcp_read_req(): ERROR: tcp_read_req: error reading New startup (first lines): Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [daemonize.c:583]: set_core_dump(): core dump limits set to 18446744073709551615 Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2475]: main(): tls support enabled, but no tls engine available (forgot to load the tls module?) Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: WARNING: <core> [main.c:2476]: main(): disabling tls... Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [async_task.c:88]: async_task_init(): start initializing asynk task framework Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): xmlrpc Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules//sl.so] Nov 13 11:44:42 lasola /usr/sbin/kamailio[16113]: DEBUG: <core> [sr_module.c:959]: init_mod(): sl In our config file we have the following lines for TLS (pretty late, after all other module loading and after most parameters): #!ifdef ENABLETLS loadmodule "tls.so" modparam("tls", "private_key", "/etc/ssl/private/my.kamailio-key.pem") modparam("tls", "certificate", "/etc/ssl/certs/my.kamailio.crt") #!ifdef TLS_CA_CHAIN # Maybe we want to use a chain to the CA modparam("tls", "ca_list", "/etc/ssl/certs/my.ca-bundle.crt") #!endif enable_tls=1 listen=tls:1.2.3.4:5061 #!endif After starting up, kamailio listens on port 5060, but not on port 5061. In version 4.1.1, this config worked without a problem. Has anybody seen this before? the tls module is there and available, it doesn't say anything about "cannot load module", and it is only a warning message. I'm also wondering, why this message is the first after starting the server. From config I would expect that sl, tm and all the other modules should be initialized before tls. Best Regards, Sebastian

2 9

Detecting problems of kamailio under very high load
by aft 16 Nov '15

16 Nov '15

We have multiple kamailio servers with 4 cpu cores and 16G RAM. We use kamailio+rtpproxy as a outbound sip proxy. Usually there are many thousands of concurrent sip sessions of occurring there. Periodically sometimes it just stops serving request and spits out 5** replies. At that point we usually reload the kamailio daemon and things return back to normal. As we have a lot of servers, manually doing this is a pain in the neck. So We installed a homer/sipcapture on a separate server . And from there, we periodically scan for 500 msgs for a given time interval. If there are any, we then proceed to reload kamailio. I know its hell of an inefficient system to monitor. So I'm wondering, if there can be any more rational way to detect problems pre-emptively. any fifo/mi command which we could run to find out if the kamailio instance is hung up? Cheers Arif

3 2

Loading mohqueue stops kamailio from starting up
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello all, I installed kamailio on debian from kamailio's apt repository. Configured it with kamdbctl, started it up properly, no issues. I configured mysql also, worked properly. Next i loaded up "mohqueue" and everything went to hell :) This is my config: loadmodule "mohqueue.so" # ----- moh-queue ----- modparam ("mohqueue", "db_url", "mysql://user:password@localhost/kamailio") modparam ("mohqueue", "db_qtable", "mohqueue") modparam ("mohqueue", "db_ctable", "mohqcalls") modparam ("mohqueue", "mohdir", "/var/kamailio/MOH") and my db table for mohqueue is: +----+------------+-------------------------+--------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------------+-------------------------+--------------------+---------+-------+ | 1 | test_queue | sip:1234@172.16.163.130 | /var/kamailio/MOH/ | test | 9 | +----+------------+-------------------------+--------------------+---------+-------+ when starting, i get: ... 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(5069) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(5069) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(5069) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(5069) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Anyone can help? Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 2

(no subject)
by David Villasmil Govea 14 Nov '15

14 Nov '15

Hello guys, I'm trying to startup 4.3 and i'm getting the following: 0(4420) DEBUG: <core> [db_row.c:117]: db_allocate_row(): allocate 192 bytes for row values at 0x7fe6bf6b5a48 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [1] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [ sip:1234@172.16.163.130] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [/var/kamailio/MOH/] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test.8] 0(4420) DEBUG: <core> [db_val.c:118]: db_str2val(): converting STRING [test_queue] 0(4420) DEBUG: <core> [db_val.c:74]: db_str2val(): converting INT [9] 0(4420) ERROR: mohqueue [mohq_db.c:504]: update_mohq_lst(): Queue,Field (test_queue,mohdir): Unable to find MOH files (/var/kamailio/MOH//test.8.*)! 0(4420) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <bind_sl> in module sl [/usr/lib/x86_64-linux-gnu/kamailio/modules/sl.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_tm> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_newtran> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_tcp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay_to_udp> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_relay> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_forward_nonack> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <t_release> in module tm [/usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so] 0(4420) DEBUG: <core> [sr_module.c:689]: find_mod_export_record(): find_export_record: found <load_rr> in module rr [/usr/lib/x86_64-linux-gnu/kamailio/modules/rr.so] 0(4420) DEBUG: <core> [sr_module.c:695]: find_mod_export_record(): find_export_record: <rtpproxy_answer> not found 0(4420) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(4420) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules Any help is appreciated, Thanks! -- DVG -- Imagination is more important than knowledge Albert Einstein

2 1

Kamailio queues
by David Villasmil Govea 13 Nov '15

13 Nov '15

Hello Guys, I'm new to this list. I'm trying to setup a queue with 4.3.3 (x86_64/linux) and I see a couple of things i'm wondering about: - kamdbctl created the table "mohqueues" and the associated value in the "version" tables. Whereas kamailio uses "mohqueue" (no extra "s" at the end). - after fixing that, it finds the table, but when searching for the moh file, which i set as "/var/kamailio/MOH/test.8" on the mohqueue table, but kamailio can't find it: 0(61540) ERROR: mohqueue [mohq.c:390]: mod_init(): Unable to load rtpproxy_answer 0(61540) ERROR: <core> [sr_module.c:962]: init_mod(): Error while initializing module mohqueue (/usr/lib/x86_64-linux-gnu/kamailio/modules/mohqueue.so) ERROR: error while initializing modules ls -l /var/kamailio/MOH/test.8 -rw-rw-rw- 1 root root 166038 Jun 10 2009 /var/kamailio/MOH/test.8 Can anyone lend a hand? Thanks! David -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Re: [SR-Users] Flaky & unstable RTP connections
by Infinicalls Infinicalls 13 Nov '15

13 Nov '15

Hi, I have started RTP with # ./rtpproxy -F -l localhost -s udp:*:7890 and when I issue #netstat -pln | grep rtpp I get, udp 0 0 0.0.0.0:7890 0.0.0.0:* 8759/rtpproxy I am using another similar instance of RTP also. Still my connection seems to be flaky. This doesn't occur always. I am not using dispatcher here. Simple rtpproxy with udp:rtp1 and udp:rtp2 in kamailio.cfg file. Is there anyway to tell if RTP is healthy other than making calls and finding out? Can somebody test it from your end? Or is this problem client specific? I am using microlite and sipper on windows. phone.infinicalls.com u:test;p:test123 u:trial;p:trial123 5060 regards Ganesh Kumar

1 0

Support-for-AKAv2-MD5-authentication-in-kamailio
by Anil Kumar 13 Nov '15

13 Nov '15

Hi, I want to enable AKAv2-MD5 type authentication using a plain kamailio. Is there any way I can achieve it ? Please do needful. Thanks Anil

2 1

Kamailio with Siremis
by Ryan Holbein 12 Nov '15

12 Nov '15

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?

2 5

issue with uac_req_send to websocket peer
by Vasiliy Ganchev 12 Nov '15

12 Nov '15

Hi folk! Have a scenario: UA-A -> REGISTER -> kamailio kamailio make auth, and save(location), after this, before exit, I want to send to UA-A a custom NOTIFY with specific event (this should be done without SUBSCRIBE/PUBLISH/NOTIFY logic) .... $uac_req(method) = "NOTIFY"; $uac_req(ruri) = $sel(contact.uri); $uac_req(furi) = "sip:service@" + $rd; $uac_req(turi) = $sel(contact.uri); $uac_req(hdrs) = ......... here custom header is added xlog("L_ERR", "debuginfo, NOTIFY to ru $uac_req(ruri), fu $uac_req(furi); uac_req_send(); ............ when UA-A is connected via UDP/TCP/TLS - works OK when UA-A connected via WS/WSS", have errors: Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: <script>: debuginfo, NOTIFY to ru sip:251@10.100.1.218:47114;transport=ws, fu sip:service@10.100.1.85, Nov 10 14:59:29 /usr/sbin/kamailio[6260]: WARNING: <core> [msg_translator.c:2799]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Nov 10 14:59:29 /usr/sbin/kamailio[6260]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message I see no problem with ws connect (calls to/from this ws peer works OK), immediately after registration: root@local:/home/admin# kamctl fifo ws.dump 2: wss:127.0.0.1:40239 -> wss:127.0.0.1:5061 (state: OPEN, last used 5s ago, sub-protocol: sip) The difference in UDP/TCP/TLS from WS/WSS - that first connects directly to Kamailio to port 5060/5061, but second connects to custom ports, where listen special daemon and proxy sip to kamailio 5060/5061 via localhost I tried also to add in script $uac_req(sock) = "tls:127.0.0.1:5061"; but the result is the same version: kamailio 4.2.5 (i386/linux) any suggestion? cheers -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-uac-req-send-to-websocke… Sent from the Users mailing list archive at Nabble.com.

2 2

Unable to find dialog
by Ryan Brindley 12 Nov '15

12 Nov '15

Hey community, Our current switch is being overloaded and when it does we're getting these 2 messages spammed in our logs: WARNING: dialog [dlg_handlers.c:1127]: dlg_onroute(): unable to find dialog for ACK with route param '35d.d2f1' [3411:7981] ERROR: tm [t_reply.c:1341]: t_should_relay_response(): ERROR: t_should_relay_response: status rewrite by UAS: stored: 503, received: 100 We're obviously working on not having the switch get so overloaded, but I'm also curious to understand why it can't find the dialogs. Is it a memory issue? Networking/dropping packets issue? I also imagine it's not as simple as a single overall answer either, but nevertheless I figured I'd reach out to the community and see if anyone has an understanding of what is happening and would be willing to take a few mins to explain. Thanks in advance, Ryan Brindley Software Development Officer Stratics Networks, Inc. 1.866.635.6918 x108

2 1

Active-Passive Auto deployment of Kamailio
by Gholamreza Sabery 11 Nov '15

11 Nov '15

Hello. Recently I started a very simple open source project for auto deploying an Active-Passive Kamailio cluster using Corosync and Pacemaker. For auto deployment I used Ansible. I thought it can be useful for some users. Link of project's repository: https://github.com/ghrst/Kamailio-HA

2 2

Kamailio can not read to_tag at BYE request
by Yuriy Gorlichenko 11 Nov '15

11 Nov '15

Hello. I have some one provider that sends me BYE request with to_tag that kamailio can not parse IP my.pro.vi.der.5060 > my.ser.vi.ce.5060: UDP, length 406 E.......x...S.0X_..L......8.BYE sip:0987654321@my.ser.vi.ce SIP/2.0 Via: SIP/2.0/UDP my.pro.vi.der:5060;branch=z9hG4bKsj7bv820c0s18k8a15q1sd000ag33.1 Call-ID: 00137fb21f65f19a3094c38b30d80f5c CSeq: 1479 BYE From: <sip:1234567890@my.pro.vi.der>;tag=SDi9r0899-5ci36vojn0 To: <sip:0987654321@my.pro.vi.der>;tag=as6635bbe1 Reason: Q.850;cause=16;text="Normal call clearing" Max-Forwards: 69 Content-Length: 0 So I guess that problem in provider (may be ";" with another code) But may be we have some instruments to get tag (without parsing strings in a config file or something like that )

2 1

Removing transport=TLS from the contact header
by Rizwan Khan 10 Nov '15

10 Nov '15

I want to remove the transport=TLS from the contact header. How can I do that inside Kamailio? The flow is an under: Client --------> Kamailio -------------> FS TLS UDP Client <-------- Kamailio <------------- FS TLS UDP FS gets a request on udp but it complains about TLS profile as the contact header has the transport=TLS. Since its receing traffic on UDP, if I only get rid of the value from the Contact Header, it would be great. Thanks, Rizwan Khan

4 15

Header field Contact not changed on kamailio 4.2
by Зуев Михаил 09 Nov '15

09 Nov '15

Hi all, I need to modify the host part of a contact header. I'm trying something like: route { ..... if (remove_hf("Contact")){ if (insert_hf("Contact: rn", "Contact")) xlog("Contact modified"); } if (!t_relay("YYY.YYY.YYY.YYY","5060")) { sl_reply_error(); } ...... } When i look SIP DUMP on YYY.YYY.YYY.YYY side i see older Contact field. I try subst_hf with regexp, subst and simply remove Contact, all this does not work. What i am doing wrong? -- С уважением Зуев Михаил

3 2

Examples
by Ryan Holbein 09 Nov '15

09 Nov '15

Hello, I have everything setup and installed... Does anyone have a good link or could tell me the steps of how to connect my trunks to phone provider and then another one would be how to route the calls to the internal PBX system. Thank you

3 4

kamailio enabling reply to options
by amar Smart Telecom 09 Nov '15

09 Nov '15

Dear All, Could you pls advise: How can we enable 'options' in Kamailio. Other server is sending "options" but our Kamailio is not responding. Thanks.

2 1

Re: [SR-Users] Branch & Forking
by Daniel-Constantin Mierla 09 Nov '15

09 Nov '15

Hello, the aggregation of replies for a parallel fork is done automatically and negative responses are absorbed. Can you send a ngrep trace of a call where you found that the negative response of a branch is not absorbed in case a branch answered? Cheers, Daniel On 05/11/15 16:41, Marino Mileti wrote: > I'm playing with branch and i'm using this simple route (copied from kamailio website) > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > t_relay(); > break; > } > > A B & C rings at the same time. > > If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. > > Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) > > Marino Maria Mileti > marino.mileti(a)alice.it > > > --- > Questa e-mail è stata controllata per individuare virus con Avast antivirus. > https://www.avast.com/antivirus > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- 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

Outgoing SIP messages
by Varghese Paul 09 Nov '15

09 Nov '15

Hi all, Is it possible to log the outgoing SIP messages relayed from kamailio ? Regards Varghese Paul

3 3

Question about siremis
by Bruno Salzano 09 Nov '15

09 Nov '15

Hello, i hope not to be OT. However I noticed that latest release is 4.2.0 and that this version is compatible with kamailio 4.2.x. Is it compatible also with 4.3.x? Another question is related to Siremis development: this great webapplication is developed using PHPOpenBiz Framework. From several days the PHPOpenBiz website is down. So will Siremis development continue on PHPOpenBiz? Is there some release schedule? Best Regards Bruno

2 1

Preprocessor define puzzle
by Alex Balashov 09 Nov '15

09 Nov '15

Hi, I've got: # define WITH_VENDOR_ROUTING_LNPOIP at the top of my config. This is intended to mean that this constant is commented out and disabled ('# define' vs. '#!define'), and tests correctly in a given request route. Later: route[STAGE_3] { ... #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED STAGE 3\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED STAGE 3\n"); #!endif route(OUTBOUND_VENDOR_SELECT); } This tests correctly: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED STAGE 3 However, in route[OUTBOUND_VENDOR_SELECT]: #!ifdef WITH_VENDOR_ROUTING_LNPOIP xlog("L_INFO", "LNPOIP ENABLED\n"); #!else xlog("L_INFO", "LNPOIP NOT ENABLED\n"); #!endif This prints: Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP ENABLED Nov 8 21:17:54 centosity6 /usr/local/sbin/kamailio[12296]: INFO: LNPOIP NOT ENABLED ... which I must admit is quite puzzling. Looks like it's both defined and not defined. Running 4.3:16be09c here. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Kamailio 4.2.3 generates core dump when using sql_xquery in event_route htable:mod-init
by Jan Hazenberg 06 Nov '15

06 Nov '15

Daniel, I'm currently running a kamailio 4.2.3 with htable enabled and i'm trying to load data into a hash table using the "event_route[htable:mod-init]". When i try to execute a sql_xquery kamailio generates a core dump on startup. Example event_route: event_route[htable:mod-init] { sql_xquery("kamailio", "SELECT id,number FROM blacklist_number", "blacklist_number"); } Log output when starting kamailio: 0(16905) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(16905) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(16905) INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 0(16905) INFO: auth [auth_mod.c:350]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.allow 0(16905) INFO: permissions [permissions.c:608]: mod_init(): default allow file (//etc/kamailio/permissions.allow) not found => empty rule set 0(16905) INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: //etc/kamailio/permissions.deny 0(16905) INFO: permissions [permissions.c:617]: mod_init(): default deny file (//etc/kamailio/permissions.deny) not found => empty rule set 0(16905) INFO: pike [pike.c:115]: pike_init(): PIKE - initializing 0(16905) INFO: pike [ip_tree.c:88]: init_lock_set(): probing 256 set size 0(16905) INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 124928 0(16905) INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 249856 0(16905) INFO: rtpproxy [rtpproxy.c:1593]: rtpp_test(): rtp proxy <udp:10.1.8.20:22000> found, support for it enabled 0(16905) NOTICE: <script>: Startup event route! Segmentation fault (core dumped) Could this be a bug? Or is there a better way to load data from a custom table into a hashtable on startup. Thanks, Jan Hazenberg

2 2

running kamilio on mac
by Stefan Mancunianet 06 Nov '15

06 Nov '15

Hi, I’ve installed kamilio on mac, but when trying to run it i get the following error message: [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Can sctp flag be disabled ? It’s not required for my needs, and from what I found on the web, it’s very tricky to use sctp on Mac. Thanks

3 4

Webpage Display
by Ryan Holbein 06 Nov '15

06 Nov '15

Hello, I have installed: CentOS 6 Kamailio<http://www.kamailio.org/> v4.2 Siremis v4.2 I went thought the siremis install just fine but when you get to the last page and click launch siremis the very next screen is displayed wrong. Unable to login and display any images.

3 6

segmentation fault when mongodb & tls enabled. ( kamailio 4.3.2)
by Jijo 06 Nov '15

06 Nov '15

I'm observing a segmentation fault when mongodb & tls enabled. It doesn't happen when one of the module is disabled. OS: centos 6.7 kamailio 4.3.2 mongo-c-driver version 1.1.10. It happens immediately after i register a subscriber. The location table is updated with the new data and in parallel core is generated as well. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 233 ATOMIC_FUNC_CMPXCHG(cmpxchg, "cmpxchgl %2, %1", int , int) Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 0x00007fd3adf65e69 in atomic_cmpxchg_int (var=0x56444e3055445030, old=0, new_v=1) at ../../atomic/atomic_x86.h:233 #1 0x00007fd3adf65eb6 in futex_get (lock=0x56444e3055445030) at ../../futexlock.h:99 #2 0x00007fd3adf66468 in prv_lock_tree_branch (b=192 '\300') at ip_tree.c:47 #3 0x00007fd3adf664e5 in lock_tree_branch (b=192 '\300') at ip_tree.c:66 #4 0x00007fd3adf6dda6 in swap_routine (ticks=123206032, param=0x0) at pike_funcs.c:293 #5 0x00000000005fef8e in compat_old_handler (ti=1971296512, tl=0x7fd32ccea430, data=0x7fd32ccea430) at timer.c:996 #6 0x00000000005ff94c in slow_timer_main () at timer.c:1130 #7 0x00000000004a8676 in main_loop () at main.c:1628 #8 0x00000000004ae38e in main (argc=11, argv=0x7fff1d97ec28) at main.c:2533 (gdb) On Clean up after the core, mongo db does segmentation fault as well Loaded symbols for /lib64/libnss_files-2.12.so Reading symbols from /lib64/libnss_dns-2.12.so...Reading symbols from /usr/lib/debug/lib64/libnss_dns-2.12.so.debug...done. done. Loaded symbols for /lib64/libnss_dns-2.12.so Core was generated by `/usr/local/kamailio_proxy/sbin/kamailio -f /usr/local/kamailio_proxy/etc/kamail'. Program terminated with signal 11, Segmentation fault. #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 28 if ((mutex->__data.__kind & PTHREAD_MUTEX_ROBUST_NORMAL_NP) == 0 Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-15.el6_6.2.x86_64 cyrus-sasl-plain-2.1.23-15.el6_6.2.x86_64 db4-4.7.25-19.el6_6.x86_64 keyutils-libs-1.4-5.el6.x86_64 krb5-libs-1.10.3-42.el6.x86_64 libcom_err-1.41.12-22.el6.x86_64 libcurl-7.19.7-46.el6.x86_64 libidn-1.18-2.el6.x86_64 libselinux-2.0.94-5.8.el6.x86_64 libssh2-1.4.2-1.el6_6.1.x86_64 libunistring-0.9.3-5.el6.x86_64 libxml2-2.7.6-20.el6.x86_64 mysql-libs-5.1.73-5.el6_6.x86_64 nspr-4.10.8-1.el6_6.x86_64 nss-3.18.0-5.3.el6_6.x86_64 nss-util-3.18.0-1.el6_6.x86_64 openldap-2.4.40-5.el6.x86_64 openssl-1.0.1e-42.el6.x86_64 zlib-1.2.3-29.el6.x86_64 (gdb) bt #0 __pthread_mutex_destroy (mutex=0x7fd32cce7be8) at pthread_mutex_destroy.c:28 #1 0x00007fd32c86f9e6 in _mongoc_ssl_thread_cleanup () at src/mongoc/mongoc-ssl.c:555 #2 _mongoc_ssl_cleanup () at src/mongoc/mongoc-ssl.c:106 #3 0x00007fd32c8606a9 in _mongoc_do_cleanup () at src/mongoc/mongoc-init.c:127 #4 0x000000344480cdb3 in pthread_once () at ../nptl/sysdeps/unix/sysv/linux/x86_64/pthread_once.S:104 #5 0x00007fd32c8523df in __do_global_dtors_aux () from /usr/lib64/libmongoc-1.0.so.0.0.0 #6 0x0000000000000000 in ?? () (gdb) mongo-c-driver version 1.1.10 version: kamailio 4.3.2 (x86_64/linux) b5980b-dirty 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, F_MALLOC, DBG_F_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: b5980b -dirty compiled on 13:35:24 Sep 28 2015 with gcc 4.

3 11

UacRegInfo flags
by DanB 06 Nov '15

06 Nov '15

Hey Guys, I have been digging around but could not find some proper documentation on flags which are coming back in the UacRegInfo rpc query. Any chance I can find out from the flags what is the status of that registration(whether password failed or user online)? Thanks in advance for any kind of tip! DanB

2 2

Webpage Display
by Ryan Holbein 05 Nov '15

05 Nov '15

Can any help me with this .. please go to 65.61.9.54 and look at webpage how do I fix it?

1 0

q-value on unsupported clients
by Daniel W. Graham 05 Nov '15

05 Nov '15

I have a need to use q-value for serial forking but my clients don't support it. I need to specify unique q-value on all ports on the client device. I explored using the User-Agent header and appending q-value, but this wouldn't help on multi-port clients. Some things I have been thinking about are 'abusing' the authorization user or from user to somehow append q-value but this opens the door to other issues, and a database trigger however there would be no way to match this up to a record in another table as the usernames would be the same. Any thoughts/ideas or methods others have used in situations like this? -Dan

4 7

Branch & Forking
by Marino Mileti 05 Nov '15

05 Nov '15

I'm playing with branch and i'm using this simple route (copied from kamailio website) route { seturi("sip:a@example.com"); append_branch("sip:b@example.com"); append_branch("sip:c@example.com"); t_relay(); break; } A B & C rings at the same time. If B or C aren't registered, and the caller send a CANCEL, it receives also a 404 messsage because for example B wasn't online. Is it possible that Kamailio "absorb" this 404? I don't want that my caller receive it :) Marino Maria Mileti marino.mileti(a)alice.it --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Adding proprietary event in kamailio
by Priyaranjan Nayak 05 Nov '15

05 Nov '15

Hi All, If I send a SUBSCRIBE message with "Event: presence" then I am getting 200 OK. When I am sending some proprietary event(i.e. Event: as-feature-event ) then I am getting 489 Bad event. Could you please help me how can I add proprietary event in the kamailio-4.2.5 to handle the SUBSCRIBE message with proprietary event ? Thanks Priyaranjan

2 6

pv_printf buffer for spec value
by Sven Neuhaus 05 Nov '15

05 Nov '15

Hello, I'm getting these errors since the last couple of changes I made to the Kamailio 4.3.x configuration: ERROR: <core> [pvapi.c:1361]: pv_printf(): no more space for spec value ERROR: <core> [pvapi.c:1370]: pv_printf(): buffer overflow -- increase the buffer size... I'm not even using pv_printf() so this error is a bit surprising. The first thing I tried was increasing the buffer from its default size of 1024 to 4096 (pv_buffer_size=4096) but the error messages persist. What other functions use this buffer? Regards, -Sven Neuhaus

3 10

RTP Proxy to integrate with kamailio
by Safdar Khan 05 Nov '15

05 Nov '15

Hi friends, Its been two days and still finding a way to install rtp proxy and integrate it with kamailio. I have checked running status of rtp proxy by *netstat* command in this way... *#netstat -pln | grep rttpproxy* and i got the output like that(i assume its working): *udp 0 0 127.0.0.1:7722 <http://127.0.0.1:7722> 0.0.0.0.*:* Well here is my setup of kamailio and rtp proxy... 1.I am using two computers one for running kamailio and another one for rtp proxy. 2.Both are running seperately. 3.Two android mobile phones to test the service(App using imsdroid). I am little bit confused with ip settings in kamailio configuration file in the part where *rtpproxy_sock* is given.For now i have given the *ip:port of *the computer on which rtp proxy is running separately. And i am getting the error message in *syslog* like this *can't send command to a RTP proxy* *proxy <udp:192.168.5.97:7722 <http://192.168.5.97:7722>> does not respond, > disable it* *can't get version of the RTP proxy* Actually i want to record the calls. Any kind of help to configure kamailio with rtp proxy is appreciated.

4 4

listening on multiple port
by Thibault Gueslin 04 Nov '15

04 Nov '15

I would like to set up kamailio in order to listen to multiple port. I have set : port=5060 port=5228 But it always listen on the port indicated on last line: Listening on udp: 127.0.0.1:5228 udp: x.x.x.x:5228 tcp: 127.0.0.1:5228 tcp: x.x.x.x:5228 Aliases: I don't have any "listen" in my config file. Is it supposed to work like that ?

4 7

[CFP] FOSDEM 2016, RTC devroom, speakers, volunteers neeeded
by Daniel Pocock 04 Nov '15

04 Nov '15

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2016 takes place 30-31 January 2016 in Brussels, Belgium. http://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (including the Saturday night dinner), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 30 January 2016. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Speaking opportunities ---------------------- Note: if you used Pentabarf before, please use the same account/username Main track: the deadline for main track presentations is midnight today, 30 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit Real-Time Communications dev-room: deadline 27 November Please also use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM16/ Other dev-rooms: some speakers may find their topic is in the scope of more than one dev-room. It is permitted to apply to more than one dev-room but please be kind enough to tell us if you do this. You can find the full list of dev-rooms at https://www.fosdem.org/2016/schedule/tracks/ Lightning talks: deadline 27 November The lightning talks are an excellent opportunity to introduce a wider audience to your project. Given that dev-rooms are becoming increasingly busy, all speakers are encouraged to consider applying for a lightning talk as well as a slot in the dev-room. On the "General" tab, please look for the "Track" option and choose "Lightning Talks". https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. For any questions, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 30 January - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists FOSDEM is made possible by volunteers and if you have time to contribute, please feel free to get involved through https://volunteers.fosdem.org/ Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 28 and 29 January 2016 http://wiki.xmpp.org/web/Summit_19 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially on 29 January. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 29 January On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: http://planet.jabber.org ralphm(a)ik.nu http://planet.sip5060.net daniel(a)pocock.pro http://planet.opentelecoms.org daniel(a)pocock.pro Please also link to the Planet sites from your own blog or web site. Contact ======= For discussion and queries, please join the free-rtc mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Daniel Pocock <daniel(a)pocock.pro> Ralph Meijer <ralphm(a)ik.nu> Saúl Ibarra Corretgé <saul(a)ag-projects.com> Iain R. Learmonth <irl(a)debian.org>

3 3

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Ok, good to know it was solved. For the records, if you want to have different number of max contact per user, then you can rely on the xavp configured to registrar xavp_cfg parameter, e.g.,: modparam("registrar", "xavp_cfg", "reg") $xavp(reg=>max_contacts) = 2; save("location"); Cheers, Daniel On 04/11/15 12:23, Marino Mileti wrote: > > Thank you Daniel!! > > I've removed the 0x04 flag and now works great!! > > > > Cheers, > > Marino > > > > On 03/11/15 17:30, Daniel-Constantin Mierla wrote: > > > > Hello, > > On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I > set it to 1, all REGISTER overwrite location info. > > > max_contacts will reject new registrations/contacts if there is > already a valid contact in the location table. > > A flag as second parameter to save(...) is the option to overwrite > existing contact with the one from current registration request. > > Cheers, > Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/ > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > > Avast logo <https://www.avast.com/antivirus> > > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > > _______________________________________________ > 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 > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- 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

Re: [SR-Users] Avoid multiple registration
by Daniel-Constantin Mierla 04 Nov '15

04 Nov '15

Hello, On 03/11/15 11:29, Marino Mileti wrote: > > I would like to avoid multiple registration with the same AOR... > > I know that there's a parameter for this (max_contacts) but if I set > it to 1, all REGISTER overwrite location info. > max_contacts will reject new registrations/contacts if there is already a valid contact in the location table. A flag as second parameter to save(...) is the option to overwrite existing contact with the one from current registration request. Cheers, Daniel > > > > Instead i would like to reply with error if a REGISTER arrive with the > same AOR just registered on location server. > > > > Is that possible? Any help is appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo <https://www.avast.com/antivirus> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com <https://www.avast.com/antivirus> > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- 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 1

Need to increase performance to handle more SUBSCRIBE messages
by Priyaranjan Nayak 04 Nov '15

04 Nov '15

Hi All, I am sending 300 REGISTER per second and getting response properly.When I am sending 50 SUBSCRIBE per second the Kamailio is not responding properly. Now We are using kamailio-4.2.5 version. Is there any procedure to increase the performance to handle more SUBSCRIBE in a second ? Thanks, Priyaranjan

2 3

Re: [SR-Users] Avoid multiple registration
by Olle E. Johansson 04 Nov '15

04 Nov '15

> On 03 Nov 2015, at 11:29, Marino Mileti <marino.mileti(a)alice.it> wrote: > > I would like to avoid multiple registration with the same AOR... > I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. > > Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. > > Is that possible? Any help is appreciated J > A simple glance at the documentation page for the REGISTRAR shows me a function called “registred”. http://kamailio.org/docs/modules/4.3.x/modules/registrar.html#registrar.f.r… "The function returns true if the AOR in the URI is registered, false otherwise. “ If you use that one and refuse a registration if true, you will get exactly what you want. Reading available documentation is a good thing (TM) :-) Regards, /Olle

1 0

Serial Forking and Lookup Function
by Daniel W. Graham 04 Nov '15

04 Nov '15

Looking for some clarification / direction on setting up forking. I have configured clients to use q-value and is stored in location table. I see several examples as described in TM module documentation but does not mention use of lookup() function. Per registrar documentation- lookup() The functions extracts username from Request-URI and tries to find all contacts for the username in usrloc. If there are no such contacts, -1 will be returned. If there are such contacts, Request-URI will be overwritten with the contact that has the highest q value and optionally the rest will be appended to the message (depending on append_branches parameter value). Append_branch parameter The parameter controls how lookup function processes multiple contacts. If there are multiple contacts for the given username in usrloc and this parameter is set to 1, Request-URI will be overwritten with the highest-q rated contact and the rest will be appended to sip_msg structure and can be later used by tm for forking. Does lookup() build the destination set and then I can proceed by doing t_load_contacts(), t_next_contacts() and setup failure route? Any direction is appreciated! Thanks, -dan

2 2

Huge TLS performance regression migrating from 3.3.7 to 4.2.6
by Tristan Mahé 03 Nov '15

03 Nov '15

Hi List, I've been reaching a strange thing lately, trying to upgrade to kamailio 4.2.6 ( identical config ): - 35% loss of performance on TLS connections ( Sipp REGISTER scenario, easily reproduced ). - more TCP workers needed to ease the TCP queues ( 32 workers on 3.3.7, at least 64 are needed on 4.2.6 to avoid "queue full" messages in logs ). - kamailio 4.2.6 intermittent processing of requests ( Kamailio stops accepting new tcp connections, then resume a bit later, then stops again,... ) => starts happening when Kamailio is treating more than 20k tcp connections, with between 500-1k req/s. During those periods, trying to connect to kamailio with an openssl s_client -connect will end up in a timeout or with a long delay before the certificate is received ( 03 CONNECTED ). - kamailio 4.2.6 dropping connections, kamailio 3.3.7 keeping those connections: after a while, the number of active connections starts to decrease on kam4, on kam3 it stays constant. I was wondering if I was alone in this situation or if someone else is impacted, has some advices ? It does not seem to be a config issue, as it is the same that was deployed with both versions.

2 8

Testing RTP Proxy
by Safdar Khan 03 Nov '15

03 Nov '15

Hi friends, I have configured kamailio to work with rtp proxy and i have followed this link http://nil.uniza.sk/sip/kamailio/rtp-proxy-kamailio-32-configuration-debian… However i am not sure that it is working perfectly or not. My Question is.. 1.Is above link ensures that rtp proxy will work with kamailio or there is another way to deal with rtp proxy and kamailio? 2.If the above link is successful integration of rtp proxy and kamailio then how to test it. Thank you

2 1

Re: [SR-Users] Domain register_myself
by DanB 03 Nov '15

03 Nov '15

Hey Daniel, That would make sense also why it worked for me in the past! Many thanks for so fast answer! DanB

1 0

Avoid multiple registration
by Marino Mileti 03 Nov '15

03 Nov '15

I would like to avoid multiple registration with the same AOR... I know that there's a parameter for this (max_contacts) but if I set it to 1, all REGISTER overwrite location info. Instead i would like to reply with error if a REGISTER arrive with the same AOR just registered on location server. Is that possible? Any help is appreciated J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

host for pstn.gw
by Max 03 Nov '15

03 Nov '15

Hi. I'd like to use Kamailio as a SIP proxy routing calls to upstream SIP provider. As far as I've understood the parameters I should set are: pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" The problem is that my upstream provider uses dynamic IP for its server. How can I use hostname instead of ip. >From what I see in route[PSTN] example it seems like it should just work if I use pstn.gw_ip = "sip.myprovider.lol" desc "My upstream SIP host" But if so - why the confusing name? And if not - how do I make it work? Is there pstn.gw_host or smth like that? regards, Max.

3 2

DB Problems after upgrade
by Nahum Nir 02 Nov '15

02 Nov '15

Hi All, I had a working Kamailio with Siremis. I upgraded wanted a new cfg file (big mistake, I know). I defined the DBURL at kamailio.cfg and the server start without errors, users can register but I can not make calls and Siremis missed calls table is not updated. What files and where should I update? Should I run kamdbctl? Thanks, Nir

2 1

Kamailio configuration file rtpproxy_manage not found when starting
by amar Smart Telecom 02 Nov '15

02 Nov '15

Dear All, I was trying to make such that rtpproxy is used for all calls in kamailio, Part of Kamailio.cfg looks like following: ============================= loadmodule "rtpproxy.so" # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") 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"); rtpproxy_manage("co") ; # added amar 20151015 } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } But when I run Kamailio start it gives error: Error looks like following: =================== Oct 15 16:13:42 smart kamailio[18472]: * Not starting Kamailio SIP server: invalid configuration file! Oct 15 16:13:42 smart kamailio[18472]: * Oct 15 16:13:42 smart kamailio[18472]: * 0(18480) ERROR: <core> [cfg.y:3299]: yyparse(): cfg. parser: failed to find command rtpproxy_manage (params 1) What could be the problem?

3 3

Kamailio E164 Normalisation
by Daniel Bryars 02 Nov '15

02 Nov '15

Hello, I have a requirement to normalise a number to an international E164 inside Kamailio. In my scenario I'm receiving a call from "a platform", and then making a call out to the same platform. I'm then adding some services to that call (e.g. recording it) on the way through. The problem is that sometimes the number I receive in the FROM field in the initial invite from "A platform" has 44 at the front, but no +. "A Platform" is then rejecting the call back in from myself, because the FROM field isn't formatted correctly. Scenario 1 - WORKS FINE - sometimes we see this. PSTN -> FROM:07700900000 -> A PLATFORM -> FROM:07700900000 -> ME Call Centre <- A PLATFORM <- FROM:07700900000 <- ME Scenario 2 DOES NOT WORK - sometimes we see this. PSTN -> FROM:447700900000 -> A PLATFORM -> FROM:447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:447700900000 <- Aeriandi Scenario 3 WORKS by fixing up the number by adding a + PSTN -> FROM:447700900000 -> A PLATFORM -> FROM: 447700900000 -> Aeriandi Call Centre <- A PLATFORM <- FROM:+447700900000 <- Aeriandi (ADD a PLUS) So I'm investigating if it's possible to fix up the number and add the + in Kamailio. Clearly I could detect if the number starts with 44 and change this to +44, but this seems very brittle (what about other international numbers) - it there a proper way to do this, or a standard module? Unfortunately asking "A platform" to send me the "FROM" in a consistent format is not an option. Many Thanks, Daniel

3 2

kamctl fifo t_uac_dlg TLS
by Ahmed Salem 02 Nov '15

02 Nov '15

Hi, I have a kamailio tls configuration which is working fine for calls & messages In the tls configuration i set these parameters to yes for the server and client verify_certificate = yes require_certificate = yes But when I use "kamctl fifo t_uac_dlg MESSAGE" to send a message I get the following errors, when verify_certificate & require_certificate are set to no the message is sent correctly. Can somebody please help me resolve this issue. tls [tls_server.c:1193]: tls_read_f(): TLS write:error:14090086:SSL routines:SSL3_GET_SERVER_CERTIFICATE:certificate verify failed <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading tls [tls_server.c:1193]: tls_read_f(): TLS accept:error:14094418:SSL routines:SSL3_READ_BYTES:tlsv1 alert unknown ca <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Thanks Ahmed

2 5

Domain register_myself
by DanB 02 Nov '15

02 Nov '15

Hey Guys, I was recently playing with "register_myself" setting of the domain module (which I was the impression that I have used it successfully in the past) and I have the feeling it does not longer behave. I have the following config: """ modparam("domain", "db_url", DBURL) modparam("domain", "register_myself", 1) .... # block foreign domains here if !is_uri_host_local() { sl_send_reply("603","Only local domains accepted"); exit; } """ Based on the register_myself setting I would expect that is_uri_host_local passes the request but it does not although the request is towards the IP address of my server. Is this the intended behavior or am I just unlucky and hit a bug? Ta, DanB

2 1

dmq_usrloc for writing to DB on one machine
by Sebastian Damm 02 Nov '15

02 Nov '15

Hi, I evaluated the dmq_usrloc module, and it works pretty good if there are two identical machines which replicate their location with each other. But what I actually want to do is to send registrations to an extra machine which is just there for writing the location into a central DB (in one table for all the registrars), from where it can be accessed (by a web page for example). So for example, I have three machines, registrar 1 and registrar 2, and receiver. receiver will never interact with customers. It is just there for writing the location into the database. I configured both registrars to send their packets to receiver, and I configured the receiver to send its packets (which it actually never does) to a dns name containing both registrars. But after reading the content of some PING messages and reading something about "autodiscovery" of dmq on the mailing list, I'm not sure whether my scenario will work the way I want it to. Can somebody explain whether I can use dmq_usrloc for this purpose? Or is it only intended to be used for bidirectional replication? Best Regards, Sebastian

2 5

issues with RTP on different hosts & dispatcher mod
by Infinicalls Infinicalls 02 Nov '15

02 Nov '15

Hi, I whenever I try to add rtpproxy to my config file modparam("nathelper", "rtpproxy_sock", "udp:firsthost:12221 udp:secondhost:12221") , I get this error. 0(45810) ERROR: <core> [modparam.c:139]: set_mod_param_regex(): parameter <rtpproxy_sock> of type <1> not found in module <nathelper> 0(45810) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 423, column 82: Can't set module parameter ERROR: bad config file (1 errors) 0(45810) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized I am having with_nat. Secondly, I tried with dispatcher and it works for one way calls. ie. User1 is able to call User2, but vice versa is not possible. Both RTP1 and RTP2 are in different machines and Kamailio sits on a different machine. All have identical network setups. And no NAT. Any idea. Can somebody guide me? thanks. regards Ganesh Kumar

2 1

100% CPU usage after NTP date update
by Stefan Kohlhauser 02 Nov '15

02 Nov '15

Hello everyone! Setup: I am currently using a Kamailio 4.2.3 on an embedded device (low CPU and RAM). The device is not able to store the current date, so after a reboot its 1. Jan 1970 until NTP updates the date. Problem: If the date is updated after the Kamailio has started (a leap of 45 years) the Kamailio uses up all the CPU and renders the device useless for a very long time. Notes: There are no incomming requests at the time. (But the Kamailio is not responsive during that time anyway.) Adjusting the internal Kamailio-time seems to take longer when the date update leap is bigger. My questions: 1) Can I prevent the Kamailio from using up the entire CPU after a date update? Or maybe do the internal ticks-update more gracefully? 2) For my better understanding: What happens within the Kamailio after adjust_ticks has updated the internal date? Thanks in advance for your help! Best regards, Stefan

2 4

Multi-Array with AVPs
by Jonathan Hunter 02 Nov '15

02 Nov '15

Hi Guys, Hope all are well? sorry to post but Im just trying to get something working where Im trying to convert a php function script into the kamailio.cfg. Im getting there but wonder how I can group results of sets of avps with another avp. Say I have these variables; $(avp(NTgrp)) = 'SB100_000361@1.2.3.4|SB100_000213@1.2.3.5|SB100_000225@1.2.3.6|SB100_000361@1.2.3.7|SB100_000213@1.2.3.8';$(avp(Prio)) = '1|1|1|2|2';$(avp(Wgt)) = '10000|10000|80000|30000|70000';$(avp(CInc)) = '0|0|1|0|1';$(avp(WTot))= '100000|100000|100000|100000|100000'; I then run a while script to split up their variables via the delimiter | in user; $var(j)=0;while($var(j)<$var(count)) {$(avp(NTgrpP))=$(avp(NTgrp){s.select,$var(j),|});$(avp(PrioP))=$(avp(Prio){s.select,$var(j),|});$(avp(WgtP))=$(avp(Wgt){s.select,$var(j),|});$(avp(CIncP))=$(avp(CInc){s.select,$var(j),|});$(avp(WTotP))=$(avp(WTot){s.select,$var(j),|});xlog("L_INFO"," Now Each time we have $(avp(NTgrpP)) $(avp(PrioP)) $(avp(WgtP)) $(avp(CIncP)) $(avp(WTotP))\n");$var(j)=$var(j)+1;} This then results in;INFO: <script>: Now Each time we have SB100_000223(a)1.2.3.4 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000225(a)1.2.3.5 1 10000 0 100000INFO: <script>: Now Each time we have SB100_000386(a)1.2.3.6 1 80000 0 100000INFO: <script>: Now Each time we have SB100_000213(a)1.2.3.7 2 30000 0 100000INFO: <script>: Now Each time we have SB100_000361(a)1.2.3.8 2 70000 0 100000 Now what Id like to do is group them via priorities so I can do some further functional checks on them when in same priority group, as you can see there are 3 results at priority 1 and 2 at priority 2. Is there away I can create something of a multi-array using the AVP $(avp(PrioP)) I just created so I can group by priority? As in php I would do something like; $multi_arr[$prio_arr[$p]][] = array( "ntgrp" => $ntgrp_arr[$p], "prio" => $prio_arr[$p], "wght" => $wght_arr[$p], "cinc" => $cinc_arr[$p], "wtot" => $wtot_arr[$p], ); } Any help would be great. Many thanks Jon

3 6

Q: about CRL list (TLS)
by Vladimer Gabunia 02 Nov '15

02 Nov '15

hello all. we compiled kamailio with TLS Support. but have next problem when using CRL Lits. Our Certificate issuing scheme is follow: Offline Root CA -> Enterprise SubCA -> Server and Phone Certificate CRL list is signed by SubCA. option "require client certificate is enables (1) " When we enable CRL list, phones are not registered. CA file is offline RootCA certificate in pem format. We think that the reason is that СRL was signed by Subca or incorrect CRL format. CRL is converted from MS CRL to PEM. (What is the format for the CRL) maybe someone have experiance with similar scenarios? thanks in advance!! ________________________________ [gh.ge] ვლადიმერ გაბუნია IT სამსახურის უფროსი ტელ: (+995) 32 2505222 +8183 მობ: (995) 577 095333 შპს "ჯეო ჰოსპიტალს" სათავო ოფისი თბილისი 0160, ვაჟა-ფშაველას გამზ. № 16; http://www.gh.ge <http://gh.ge>

2 5

Problems with dispatcher module | feature disabled
by Alessio Casco 02 Nov '15

02 Nov '15

Hello! We noticed today an issue with the kamailio loadbalancing. Basically kamailio is not considering if an host is up or not, backends are set always as "FLAGS: AP" even if the host is down. I see this when the proxy starts : ERROR: dispatcher [dispatcher.c:768]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled These the parameters I have set: # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;code=404") modparam("dispatcher", "ds_ping_from", "sip:proxy@sip.nexmo.com") modparam("dispatcher", "ds_ping_interval", 20) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_hash_size", 10) modparam("dispatcher", "ds_hash_expire", 3600) modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 10) modparam("dispatcher", "use_default", 0) modparam("dispatcher", "attrs_pvname", "$avp(carrierattr)") modparam("dispatcher", "setid_pvname", "$avp(carriergroup)") modparam("dispatcher", "flags", 2) I use 4.3.3, I think this issue started after upgrading from 4.3.1 but I can't say for sure. Can someone please give us some advices on this? Thanks Alessio

2 5

Which option uses for check certificate revocation in kamailio 1.5.x ?
by Evgeniy Ivanov 02 Nov '15

02 Nov '15

Hello! I use kamailio 1.5.4-tls (i386/linux). I configured the tls options: disable_tls = 0 listen=tls:192.168.50.60:5061 tls_certificate="/etc/kamailio/tls/crt.pem" tls_private_key="/etc/kamailio/tls/key.pem" tls_ca_list="/etc/kamailio/tls/ca.pem" tls_verify_client=on tls_require_client_certificate=on I need to check certificate revocation, but the option tls_crl does not work. Which option uses for check certificate revocation in kamailio 1.5.x ?

3 6

Display name
by Loïc Boissy 30 Oct '15

30 Oct '15

Hello, I would like to know if kamailio is able to store contact "display name" in its database? Thank you for your help. Best regards. Loïc

2 2

Routing outgoing call from asterisk to registered SIP provider
by Bruno Salzano 29 Oct '15

29 Oct '15

Hello, i've DID provider that require to register to it's server. I've been able to configure this provider using kamailio and uac and to route all inbound calls from DID provider to internal asterisk media server. All is working great. Now i need help to configure call routing from internal asterisk to the external DID provider. I've done something like: route[LOCATION] { if(is_method("INVITE")) { route(FROMASTERISK); route(TOASTERISK); exit; } [...] } route[FROMASTERISK] { if (ds_is_from_list()) { xlog("L_INFO","[$fU@$si:$sp]{$rm} {$ru} Call from Media-Server Cluster\n"); rtpengine_manage("direction=internal direction=external replace-origin replace-session-connection trust-address"); $ru = "sip:TESTING_NUMBER@EXTERNAL_PROVIDER_IP"; route(RELAY); exit; } } It seems that when i route the call outbound, i get a 407. So i would like to react to this using uac_auth() as described here: http://kamailio.org/docs/modules/4.3.x/modules/uac.html#uac.f.uac_auth() Is this the right way? Can someone help me, giving hints and some working example? Using kamailio 4.3.3 with rtpengine 4.1.0 Best regards, Bruno -- The life is short.. live at your best!

4 6

gcc warning for t_cancel
by Yasin CANER 27 Oct '15

27 Oct '15

3 3

Routing between 2 domains
by kai.ohnacker＠cbc.de 27 Oct '15

27 Oct '15

Hello community, I work with Kamailio 4.2.0 which is placed between two networks (with two interfaces) and RTPproxy in bridge mode. The task from Kamailio is to handle the calls from internal networks to internal networks only on the internal Interface without bridging. For external networks vice versa. If the call comes from external to internal, then the RTPproxy should work in bridge mode and vice versa. I adapt the example with IPv4 and IPv6 routing, but this is not working for internal to external calls and vice versa (http://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6). Only external calls and only internal calls works. I think the cause of this behavior is the routing which I implemented to the RTPmanage part in the Kamailio config. The error message is given by the internal device. (call from external to internal) If I make a call from internal to external, the RTP stream is only in one direction. Internal Interface IP: 203.207.111.58 External Interface IP: 193.16.163.58 The RTPproxy config is: CONTROL_SOCK=udp:127.0.0.1:9000 EXTRA_OPTS="-l 203.207.111.58/193.16.163.58 -d WARN:LOG_LOCAL1" Kamailio.cfg: #!define INT_IP 10.96.0.0/14 #!define EXT_IP 193.0.0.0/8 # 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"); # Start Test routing if((src_ip==INT_IP && dst_ip==EXT_IP)){ rtpproxy_manage("cowie"); } if((src_ip==EXT_IP && dst_ip==INT_IP)){ rtpproxy_manage("cowei"); } 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; } Can someone help me with the right routing? Do you need some more debugging information? Cheers, Kai

2 4

sqlite errors
by nadie 27 Oct '15

27 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, sorry for the question. Kamailio 4.3.3, debian Jessie: Kamailio with sqlite all work correctly but syslog tells: Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 17) Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 50) Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: attempt to write a readonly database Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [ucontact.c:683]: db_insert_ucontact(): inserting contact in db failed Oct 26 19:36:01 voip /usr/sbin/kamailio[2643]: ERROR: usrloc [urecord.c:418]: wb_timer(): inserting contact into database failed (aor: 59) kamailio.cfg #!KAMAILIO #!define WITH_SQLITE #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS ..... #!ifdef WITH_SQLITE loadmodule "db_sqlite.so" #!endif ...... #!ifdef WITH_SQLITE modparam("auth_db", "db_url", "sqlite:///home/kami/app/kamailio.sqlite") #!define DBURL "sqlite:///home/kami/app/kamailio.sqlite" #!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 Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJWLnRoAAoJEG5DH1ilMeee+P8P/A3/YiDqiA2VsZR3GvTtAYgh 5XjGBI9MptT2IuLPe3FMnuOtKPGBkKIlcUpW8EmAvV37wVooHl0ZyF4/Ap+izgMm G8PeoOhgQ0jaY+j+p/KVK3OpN3uQprH/F+H0o5ljHi2R9oYE/FFVv8/A+mGcAztD KEmqSq9HNqcCn+4p+qB5yMM+nmi9vCGWFlPiMIInQ7YOIWLCkJJCjkeGgjKgQzJw tAYRJkoytmpTbUxTw7bwPYniQee7ATfXaMhslLU0bIGIleFjuElcC32NoBs6KZKW jYAWjMMwuVtyMxjkLq29q46/Me/D7PIbIBuxxoPUho9OKCjEN3ob2K3GRvJf6d5j p0uNYCNY6nsftdO3UzZE1+xz9NqXvpmO+oOcWw/J9N+5/vMEOorwwvmoD5JX4LSn 7jqXkeGvGoqiwPkquLEYfTWB0hOZJyDRHJWej9WBTPF063seLvaX5OFRXjuvOIHd D6fmG4vxMTB0VpANxQQuf73HA0hlYZ5WzvjoNT8cc259uWY8OPiuLcIuOfHtBphk xlttHbTnozCvCJVP4W0OLDZHwPR3AuOdBhtjOM+kIsUwzMVlVA5/Drqai/EDsX4/ tzVPHKNneBU98jPtNTmFQqWnT2sGnZeNYBO/HdcZYNGFsJ993WCQvVAZZ/0OwL+J hV3EXhaP/KqeZRtHjwFN =x7Zq -----END PGP SIGNATURE-----

2 1

Kamailio start up fails with strange IO Error on RHEL 7.1
by M S 26 Oct '15

26 Oct '15

Hi, I have been trying to setup Kamailio v4.3.3 rev. e275bc on RHEL 7.1 64bit. It has been very difficult to resolve all dependencies initially due to RHEL's extremely limited repositories and libraries support, but nevertheless I am finally able to successfully compile kamailio with all essential modules on it. However, now kamailio fails to start up and gives this strange error, -- 36(3335) CRITICAL: <core> [io_wait.c:526]: init_io_wait(): could not alloc epoll array 36(3335) CRITICAL: <core> [tcp_main.c:4523]: tcp_main_loop(): exiting... 0(3250) ALERT: <core> [main.c:725]: handle_sigs(): child process 3335 exited normally, status=255 0(3250) INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD -- I have never seem this error before nor i can find any info related to it anywhere on internet. So, i doubt it is due to any missing dependency. Any idea what may be the cause of this? Thank you.

2 6

Unable to Update Headers in failure route which were build in request route
by Sunil More 26 Oct '15

26 Oct '15

Hi All, I am trying to have a failover route where if a initial relay fails , i can forward the same request to a second provider. I am using a certain set of headers that help to authenticate over my providers. I am trying to update those headers. However update and removal both fail. The headers are still present. Please help me understand if i am wrong. Here's the code. ..... route(GET_CREDENTIALS); } remove_hf_re("X-*"); append_hf("X-Src-IP: Trusted\r\n", "Call-ID"); append_hf("X-Identity: $hdr(X-Calling-User)\r\n", "Call-ID"); append_hf("X-Auth-ID: $avp(siptrunk_username)\r\n", "Call-ID"); $avp(realm) = $fd; $rd = $avp(siptrunk_domain); $var(duri) = "sip:"+$rU+"@"+$avp(siptrunk_proxy); $du = $var(duri); t_on_failure("UAC_AUTH"); if(!t_relay()) { sl_reply_error(); exit; } } failure_route[UAC_AUTH] { if(t_check_status("(4|5)[0|8][0|2|3|4|6|8]")){ xlog("L_INFO","failover testing"); revert_uri(); route(FAILOVER); } if (t_check_status("40[17]")) { xlog("got challenged \n"); uac_auth(); xlog("L_INFO", "Return code is $retcode"); $rd = $avp(siptrunk_proxy); t_on_failure("UAC_AUTH"); t_relay(); } } route[FAILOVER]{ xlog("L_INFO","doing failover now.. "); if($avp(failover_routing_id)!=$null) { $avp(outbound_routing_id)=$avp(failover_routing_id); xlog("L_INFO","Failover carrier credentials are.."); route(GET_CREDENTIALS); #not adding headers now remove_hf_re("X-*"); append_hf("X-Src-IP: Trusted\r\n", "Call-ID"); append_hf("X-Identity: $hdr(X-Calling-User)\r\n", "Call-ID"); append_hf("X-Auth-ID: $avp(siptrunk_username)\r\n", "Call-ID"); $avp(realm) = $fd; $rd = $avp(siptrunk_proxy); $var(duri) = "sip:"+$rU+"@"+$avp(siptrunk_proxy); $du = $var(duri); xlog("L_INFO","Destination Uri from failover is : $du"); t_on_failure("UAC_AUTH2"); if(!t_relay()) { sl_reply_error(); exit; } exit; } } failure_route[UAC_AUTH2] { if(t_check_status("(4|5)[0|8][0|2|3|4|6|8]")){ xlog("L_INFO","Failover failed. cancelled call "); exit; } if (t_check_status("40[17]")) { xlog("got challenged \n"); uac_auth(); xlog("L_INFO", "Return code is $retcode"); $rd = $avp(siptrunk_proxy); t_relay(); } exit; } Thanking You, Sunil More

3 2

issue with sending "delete" to rtpengine for cancelled branch
by Vasiliy Ganchev 26 Oct '15

26 Oct '15

Hi folk! Have an issue, and do not see the obvious way how to solve it, maybe i'm missing something. The call scenario: UA-A (hard phone, no rtpengine) call ->> UA-B (has 2 registration: hard phone (no rtpengine) and WebRTC based web phone (here the rtpengine is called, in branch_route)) When Kamailio fork call to UA-B, for the branch with WebRTC UA, I do rtpengine_manage (rtpengine receive command "offer" and allocate 4 ports). If UA-B answer via hard phone, Kamailio Cancel branch to WebRTC phone, and here I need to do rtpengine_manage (to send "delete" command to rtpengine). But as I see Kamailio CANCEL all nonanswered branches automatically (I see no place where to call rtpengine_delete) Any suggestion how to cope with such issue? cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/issue-with-sending-delete-to-rtpeng… Sent from the Users mailing list archive at Nabble.com.

1 1

kamctl rtpproxy nh_show_rtpp not working
by amar Smart Telecom 25 Oct '15

25 Oct '15

Dear all, I have installed Kamailio and Rtpproxy. Following is the process (ps) info: "kamailio 648 1 0 Oct08 ? 00:00:05 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.58.25" But when I give command: " kamctl mi nh_show_rtpp" it returns following: "500 command 'nh_show_rtpp' not available" What could be the problem? Thanks, Amar.

3 5

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://vivamous.com/youth.php?lkb> brettlist(a)nemeroff.com

2 1

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://stillbontechnology.com/smiling.php?h98ha> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://janatyachar.org/woman.php?mq> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://moverdubai.net/having.php?tky> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://lapeste.org/land.php?eiz> brettlist(a)nemeroff.com

1 0

important message
by brettlist＠nemeroff.com 22 Oct '15

22 Oct '15

Hello! New message, please read <http://igoasia.org/wish.php?9w> brettlist(a)nemeroff.com

1 0

msilo dump with non 2xx answer
by Péter Barabás 22 Oct '15

22 Oct '15

Hi, I set instant message store&forward in kamailio using MSILO module. It works fine in many cases. We have special message handing (encryption in "session"), i do not go into detail since it is not necessarily to solve the problem. So there are situations, when the offline receiver should get a MESSAGE and will acknowledge with 4xx status code, like 493: the original message is stored in silo table and when the receiver will be online, it will be dumped and he will send 493. Question: how can I delete the message from silo table with MSILO (since it is not 2xx) and how can i forward this ack message to the original sender instead kamailio? Thanks. Dr. Péter Barabás Director of Software Development Arenim Technologies AB.

3 4

Kamailio and FreeSwitch - ClueCon Weekly Conference Call
by Daniel-Constantin Mierla 21 Oct '15

21 Oct '15

On Wednesday, October 21, 2015, at 17:00GMT (12:00CT, 18:00 London, 19:00 Berlin), the Cluecon weekly conference call will focus on Kamailio and FreeSwitch. I will be joining the call, answering the questions about Kamailio and its options to integrate with FreeSwitch. Expect the FreeSwitch core developers to be around to handle the questions about their project. Participation is open for anyone, you can dial in for audio or video sessions using a SIP phone or webrtc capable browser: - sip:888@conference.freeswitch.org - https://cantina.freeswitch.org/vc More dial in options (e.g., PSTN) are presented at: - https://freeswitch.org/confluence/display/FREESWITCH/ClueCon+Weekly+Confere… Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

MSILO module
by Safdar Khan 21 Oct '15

21 Oct '15

How can i store messages in silo table not only offline also online.Demo code will be appreciated.waiting for response.

2 3

htable never
by Juha Heinanen 21 Oct '15

21 Oct '15

i noticed that entries that i had added to htable at sip proxy start in event_route [htable:mod-init], e.g., $sht(htable=>app_srv::sbc::<null>) = "sip:SBC:SEMS_PORT"; had disappeared from htable after some (don't know exactly how long) time. i have not specified any autoexpire for that table: modparam("htable", "htable", "htable=>size=HTABLE_SIZE;dbtable=htable;") after restart of sip proxy they reappeared in htable like this: # sip-proxy_ctl htable.get htable 'app_srv::sbc::<null>' { item: { name: app_srv::sbc::<null> value: sip:127.0.0.1:5090 flags: 2 expire: NEVER } } what could explain the disappearance? -- juha

3 3

Implementation of RFC 5393
by Guillaume 21 Oct '15

21 Oct '15

Hi guys, What do you think about the RFC 5393 on loop detection and amplification attack protection? The RFC is short and still a proposed standard but don't you think it could be useful to prevent loop and amplification attack? Because even if the max-forward field reduces the loop to ~70 hosts (in most cases) with some techniques we could fork the message up to 2^70 messages (as described in the RFC) to crash the servers. Basically the server has to do 2 things: * check if it is not already in the via of the message * the previous check is not enough as a B2BUA could have replace the via headers, so the RFC introduces a new field called max-breadth to limit the forking. I have not seen a lot of implementation of this RFC on the free SIP software and I think it could be a good way to improve kamailio making a module for it (the easier way to implement this feature I think). In fact I'm in a research internship about VoIP security and I have time to develop such a module for kamailio if you think it's a good idea (I'm looking for some security improvements in free software solutions so if you have other idea don't hesitate to tell me). Cheers, Tetram

3 4

pua_dialoginfo publish sent without recieved etag (intermittent)
by Asgaroth 21 Oct '15

21 Oct '15

Hi All, I've come accross an intermittent issue where an initial publish is sent to our presence server, proxy recieves the subsequent 200 with etag, but the following publish sent does not contain the sip-if-match header of the recieved 200, which ends up causing trouble with BLF on our test devices, it also has the added side affect of not being cleared out of the presentity table, these entries need to be manually removed. This does not occur all the time, I can use the same phone and make several calls before this issue occurs. It is not limited to a device type, we've tried with panasonic, polycom, cisco, zoiper, yealink and linksys devices, all exhibit the same sporadic behaviour. I've attached an example sip trace of the publish messages where this happens, you can see the second publish doesnt have the sip-if-match header whose content should be that of the sip-etag in the previous 200 to the initial publish. I'm not sure if there is something I am misunderstanding here, a configuration issue, or if this is indeed a bug. Both the proxy and the presence server are running kamailio v4.3.3. Any pointers/suggestions/guidance would be appreciated. If you need any additional info, please dont hesitate to ask. Thanks

2 2

Sending ReINVITE from Kamailio
by Grant Bagdasarian 21 Oct '15

21 Oct '15

Hello, Is it possible to have Kamailio send a ReINVITE every X minutes to determine if a session is still active? I know it's a proxy and doesn't have B2BUA capabilities, but there was a module which allowed Kamailio to generate SIP messages, but I can't find it anymore. If Kamailio is not the place to do this, which component in the voip network should be responsible for this? Are there perhaps other ways to poll for session state in Kamailio? Regards, Grant

3 4

Re: [SR-Users] MySQL Connect Errors
by John Billings 20 Oct '15

20 Oct '15

Both boxes are running Debian 8. I’m still looking at the mysql logs, but I haven’t found anything relevant, yet. > On Oct 14, 2015, at 5:00 AM, sr-users-request(a)lists.sip-router.org wrote: > > Hello, > > do you have centos+selinux? Because I heard about a lot of limits with > the combination of the two. > > Have you checked the mysqld syslog messages, anything relevant there? > > Cheers, > Daniel >

3 2

Far end NAT traversal
by NITESH BANSAL 20 Oct '15

20 Oct '15

Hello, I would really appreciate if someone can point me to some documentation for Far end nat traversal in Kamailio. Thanks,Nitesh

3 2

Dialog timeout if record route have both strict and loose route
by Arun Kumar 20 Oct '15

20 Oct '15

Hi dialog timeout is occurred if 200 ok having mixed loose and strict routing entries in record route header conf file attached INVITE sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Contact: <sip:123455555@176.204.26.157:63939;ob> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 INVITE Route: <sip:11.11.11.11:345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: zeo-iphone Opcode: 19815 Content-Type: application/sdp Content-Length: 320 v=0 o=- 3653999880 3653999880 IN IP4 10.220.83.191 s=pjmedia b=AS:25 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 18 96 c=IN IP4 10.220.83.191 b=TIAS:8000 a=rtcp:4001 IN IP4 10.123.11.223 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 407 Proxy Authentication required Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK3d4f.6f8dff1ec0806cb23f56d847d5fe3934.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 INVITE Proxy-Authenticate: Digest nonce="6b24dc6996de20028c78dbeaf13edc55",realm="22.22.22.22",opaque="",stale=FALSE,algorithm=MD5 Content-Length: 0 ACK sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjvAKLQD72G88kpqC2IJwbmTcXUlHGULcW Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14602 ACK Route: <sip:11.11.11.11:345;lr> Content-Length: 0 INVITE sip:123456789@22.22.22.22:5060 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22 Contact: <sip:123455555@176.204.26.157:63939;ob> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Route: <sip:11.11.11.11:345;lr> Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: zeo-iphone Opcode: 19815 Proxy-Authorization: Digest username="123455555", realm="22.22.22.22", nonce="6b24dc6996de20028c78dbeaf13edc55", uri=" sip:123456789@22.22.22.22:5060", response="554a1da255084f7a95d3b6e7a0c48933", algorithm=MD5 Content-Type: application/sdp Content-Length: 320 v=0 o=- 3653999880 3653999880 IN IP4 10.220.83.191 s=pjmedia b=AS:25 t=0 0 a=X-nat:0 m=audio 4000 RTP/AVP 18 96 c=IN IP4 10.220.83.191 b=TIAS:8000 a=rtcp:4001 IN IP4 34.220.83.191 a=sendrecv a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 100 Trying Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Content-Length: 0 SIP/2.0 183 Session progress Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 CSeq: 14603 INVITE Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh Contact: <sip:22.22.22.22:5060> Content-Type: application/sdp Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345;lr> Content-Length: 298 v=0 o=- 3653999880 3653999880 IN IP4 50.56.154.204 s=zeodialer c=IN IP4 50.33.154.204 b=AS:25 t=0 0 m=audio 53128 RTP/AVP 18 96 c=IN IP4 50.33.154.204 b=TIAS:8000 a=rtpmap:18 G729/8000 a=sendrecv a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=X-adapter:some value SIP/2.0 183 Session Progress Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE P-Early-Media: sendrecv Contact: <sip:22.22.22.22:5060> Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 Content-Type: application/sdp From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu Content-Length: 212 v=0 o=- 1426744035 1426744035 IN IP4 50.56.154.204 s=Nexge c=IN IP4 50.22.154.204 t=0 0 m=audio 53128 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=ptime:20 SIP/2.0 200 OK Record-Route: <sip:22.22.22.22:5060;lr>,<sip:11.11.11.11:345> Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,UPDATE Contact: <sip:22.22.22.22:5060> Via: SIP/2.0/UDP 11.11.11.11:345;branch=z9hG4bK4d4f.b793c28bbb6684da0ad60d4320be2d6b.0,SIP/2.0/UDP 176.204.26.157:63939 ;rport=63939;branch=z9hG4bKPjmWCyQQWaKBZC0fGKfS2LCXCIklN-UIHI;received=176.204.26.157 To: <sip:123456789@22.22.22.22>;tag=GB-1445011078679 Content-Type: application/sdp From: <sip:123455555@22.22.22.22>;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu Content-Length: 212 v=0 o=- 1426744035 1426744035 IN IP4 50.56.154.204 s=Nexge c=IN IP4 50.33.44.204 t=0 0 m=audio 53128 RTP/AVP 18 96 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:96 telephone-event/8000 a=ptime:20 ACK sip:11.11.11.11:345 SIP/2.0 Via: SIP/2.0/UDP 176.204.26.157:63939 ;rport;branch=z9hG4bKPjldbprXw3ybGOxoeKWwvtdxLmNp8jXsOX Max-Forwards: 70 From: sip:123455555@22.22.22.22;tag=E9YwhssbjdD63KZpXep-UCmOxshWrcwu To: sip:123456789@22.22.22.22;tag=GB-1445011078679 Call-ID: VlaYDWuM5COSfEIqydwAxFM76TLv7Syh CSeq: 14603 ACK Route: <sip:22.22.22.22:5060;lr> Route: <sip:22.22.22.22:5060> Content-Length: 0

2 3

kamctl rtpproxy nh_show_rtpp not working
by amar Smart Telecom 19 Oct '15

19 Oct '15

Dear all, I have installed Kamailio and Rtpproxy. Following is the process (ps) info: "kamailio 648 1 0 Oct08 ? 00:00:05 /usr/bin/rtpproxy -s udp:127.0.0.1 7722 -u kamailio kamailio -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.58.25" But when I give command: " kamctl mi nh_show_rtpp" it returns following: "500 command 'nh_show_rtpp' not available" What could be the problem? Thanks, Amar.

2 1

RtpProxy socket timeout not working
by Arun Kumar 19 Oct '15

19 Oct '15

Hi Rtp Session timeout notification is not working , we tried with kamailio 4.2 with rtpproxy 2.0 / 1.3 beta/1.2 releases , i think kamailio rtpproxy is not submitting session timeout socket to rtpproxy kamailio 4.2.0 ----------------------------------- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "timeout_socket", "xmlrpc:http://127.0.0.1:5060/RPC") ............... rtpproxy_manage(); ................ RtpProxy ------------------------- RTPProxy 2.0 RTPProxy 1.3 Beta with patch given by *Daniel* RTPProxy 1.2 rtpproxy -F -s udp:127.0.0.1:7722 -d DBUG -f -i -T 10 RTPRoxy Log ---------------------------------------------------- DBUG:get_command: received command "4242_9 Uc0,8,101 2479e562440c023d 192.168.1.103 6074 5410ef73;1" INFO:handle_command: new session 2479e562440c023d, tag 5410ef73;1 requested, type strong INFO:handle_command: new session on a port 45984 created, tag 5410ef73;1 INFO:handle_command: pre-filling caller's address with 192.168.1.103:6074 DBUG:doreply: sending reply "4242_9 45984 " DBUG:get_command: received command "4242_10 Lc0,101 2479e562440c023d 192.168.1.102 4002 5410ef73;1 ADqH-A-mLLN0th2gGoopYMK1cy.-ozDf;1" INFO:handle_command: lookup on ports 45984/36756, session timer restarted INFO:handle_command: pre-filling callee's address with 192.168.1.102:4002 DBUG:doreply: sending reply "4242_10 36756 " INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4002 (RTP) INFO:rxmit_packets: callee's address latched in: 192.168.1.102:4003 (RTCP) INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6075 (RTCP) INFO:rxmit_packets: caller's address latched in: 192.168.1.103:6074 (RTP) INFO:process_rtp: session timeout INFO:remove_session: RTP stats: 328 in from callee, 5 in from caller, 333 relayed, 0 dropped INFO:remove_session: RTCP stats: 3 in from callee, 4 in from caller, 7 relayed, 0 dropped INFO:remove_session: session on ports 45984/36756 is cleaned up Kamailio RTP log ----------------------------------------------- 3(4242) DEBUG: rtpproxy [rtpproxy_funcs.c:148]: check_content_type(): type <application/sdp> found valid 3(4242) DEBUG: rtpproxy [rtpproxy.c:2704]: force_rtp_proxy(): proxy reply: 45984 for more log please see attachment

2 1

Kamctl cannot add user to pgsql
by Pavlo Karbovnyk 19 Oct '15

19 Oct '15

Good Day All. I run into following problem: (It may be helpful for other people how have similar problem) kamctl add 1000 password database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 <--- showing counter=0 check_db_alias: alias counter= <--- showing counter= nothing INFO: user '1000' already exists as alias ============================================================ how to fix it : in kamctl locate a function check_db_alias() { ..... CNT=`$DBROCMD "$QUERY" | $EGREP -v ERROR | $LAST_LINE` ..... } replace with CNT command with following: check_db_alias() { ..... CNT=`$DBROCMD "$QUERY" "$DBRAWPARAMS" | $EGREP -v ERROR | $LAST_LINE` ..... } ============================================================ after this kamctl add 1000 password database engine 'PGSQL' loaded Control engine 'FIFO' loaded is_user: user counter=0 check_db_alias: alias counter=0 pgsql_query: /usr/bin/psql -A -q -t -P fieldsep=' ' -h localhost -U kamailio kamailio -c 'insert into subscriber (username, domain,ha1,ha1b,password) values ('1000','127.0.0.1','6fda0594de6ea94b03a12fdd998758ae','31e118ac5b09f07 1858740d6125d901d','');' new user '1000' added Regards Pavlo.

2 1

Weight balancing with dispatcher module
by Ben Bliss 19 Oct '15

19 Oct '15

Hi, I am having issues getting the dispatcher to work using weighting. I have been digging through old posts to the list, and tried all that I have found, but am still having no luck at all. So, information... version: kamailio 4.2.4 (amd64/freebsd) be62bd flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, select, kqueue. id: be62bd Dispatcher configuration... modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 3) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") Dialplan config.. route[DISPATCH] { if(!ds_select_dst("1", "9")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); return; I am loading the values from a database, which are being loaded fine, and appear to be being formatted correctly aswell.. +----+-------+-------------------------+-------+----------+------------+----------------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+-------------------------+-------+----------+------------+----------------------+ | 1 | 1 | sipx.x.x.106:5060 | 8 | 1 | weight=100 | | | 2 | 1 | sip:x.x.x.107:5060 | 8 | 1 | weight=0 | | +----+-------+-------------------------+-------+----------+------------+----------------------+ The IP's are public, so i've masked them... The following is from kamcmd.. { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:x.x.x.107:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: weight=0 DUID: MAXLOAD: 0 WEIGHT: 0 } } DEST: { URI: sip:x.x.x106:5060 FLAGS: AP PRIORITY: 1 ATTRS: { BODY: weight=100 DUID: MAXLOAD: 0 WEIGHT: 100 } } } } } } However, when any call is placed, it will always goto the server ending 107 (with weight 0, which means, as far as I understand, it should be ignored). I've tried with ; after the weights, this made no change either. When set to round robin, or via priority, it appears to work fine. Any ideas? -- Kind Regards, --------- Ben

2 3

kamailio cfg file with user db authentication and rtpproxy
by amar Smart Telecom 19 Oct '15

19 Oct '15

Dear all, Can you pls mail me a working Kamailio cfg file which has: "Sip user db authentication" and "rtpproxy for all calls" Thank you, Amar.

3 2

Link between presence module and subscriber table
by Loïc Boissy 19 Oct '15

19 Oct '15

Hello, l would like to send a SIP message (NOTIFY for example) to a specific user when a new account is created in the kamailio database. Is it possible to do that? Best regards Loïc

2 1

Re: [SR-Users] call-limit peer
by Angel Elena 19 Oct '15

19 Oct '15

Ok, now I check dlg_manage settings. Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ -------------------------------- -----Mensaje original----- De: ycaner <yasin.caner(a)netgsm.com.tr> Enviado: Lun 19-10-2015 08:22 Asunto: Re: [SR-Users] call-limit peer Para: sr-users(a)lists.sip-router.org; > Hello; > with dialog module using dont forget set dlg_manage or setting flags. > or you can do it with pike limit for only INVITE methods. > > > > > -- > View this message in context: > http://sip-router.1086192.n5.nabble.com/call-limit-peer-tp142403p142407.html > Sent from the Users mailing list archive at Nabble.com. > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >

1 0

call-limit peer
by Angel Elena 19 Oct '15

19 Oct '15

Hi everybody. I have a kamailio 4.0 running with rtpproxy and debian 7.8, and I need limit calls by IP. My kamailio.cfg: # ----------------------------------------- #------------------------------------------ loadmodule "dialog.so" modparam("dialog", "dlg_flag", 4) modparam("dialog", "enable_stats",1) modparam("dialog", "hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) In my route outside, I have: get_profile_size("callquota", "$avp(s:tag)", "$var(SIZE)"); xlog ("L_INFO", ".... $avp(s:tag) ..... CALLQUOTA .... $var(SIZE)..... \n "); On $avp(s:tag) is the client code-peer/ip. When I view the logs... the result: Oct 18 22:18:03 sipproxy /usr/local/sbin/kamailio[19724]: INFO: <script>: .... 4012 ..... CALLQUOTA .... 0..... Always, the result is '0'. Is correct my configuration ? thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

2 1

Re: [SR-Users] Changing a user's password in the Kamailio routing logic
by Surendra.P 19 Oct '15

19 Oct '15

Hello mike, you can achieve it by doing DB update query using sqlops module. If you want to change the user's password dynamically under a certain condition, just invoke a update query to the DB using sqlops. Hope it solves your issue. Regards Surendra Hi all, I was wondering if there was a way to change a user's password from within the routing logic section of kamailio.cfg (i.e. like do what 'kamctl password' does, but in the routing logic)? Thank, Mike ******

1 0

Re: [SR-Users] Auth module Issue:UA getting challanged thrice when load
by Surendra.P 19 Oct '15

19 Oct '15

Hello Daniel, Thanks for your reply. I did some debugging, I found that I am getting error like this, earlier this log was in DBG mode, changed to error and then I ran the load.. "auth:pre_auth: Invalid nonce value received (ret 2)" Problem is that this issue getting simulated every time. I am using auth module version of kamailio 4.1.5 , since I checked with latest kamailio version 4.3.2 , I didn't find any differences in this module. Please find my finding below Auth/nonce.c:check_nonce() b_nonce2_len=calc_bin_nonce_md5(&b_nonce2, cfg, secret1, secret2, msg); if (!memcmp(&b_nonce.n.md5_1[0], &b_nonce2.n.md5_1[0], 16)) // here where condition is getting failed and it is returning 2. Regards Surendra Hello, are there any error messages in the syslog? Maybe setting debug level for auth modules to 3 and looking at logs will help to reveal what is happening. Be sure syslog is in async mode, otherwise it will slow down processing. Cheers, Daniel On 16/10/15 08:42, Surendra.P wrote: > > Hello Daniel, > > > > I have a concern regarding auth module, I am running > 2000 TPS for 1000000 subscribers, using SIPP. What I observed when > you run small load like 500 TPS, everything get passed, but for 2000 TPS > > With same subscriber.csv file some of the subscribers getting > challenged. May I this why auth module is behaving like this.. below > are the parameters which I configured in auth module > > > > modparam("auth", "nonce_expire", 3600), rest all parameters default > values. > > > > Regards > > Surendra >

1 0

Changing a user's password in the Kamailio routing logic
by Michael McVady 16 Oct '15

16 Oct '15

Hi all, I was wondering if there was a way to change a user's password from within the routing logic section of kamailio.cfg (i.e. like do what 'kamctl password' does, but in the routing logic)? Thank, Mike

1 0

Multiple crashes of Kamailio 4.2.1
by Igor Potjevlesch 16 Oct '15

16 Oct '15

Hello, 3 crashes occurred today (we never seen crashes on this server before) during a period of 20 minutes. I had a look to the coredumps. Here is a quick overview of the first entries of "bt full": core.27671 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27656 : #0 0x00007f557700a048 in free_cell (dead_cell=0x7f5560abbce0) at h_table.c:175 core.27685 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27689 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.27691 : #0 0x00007f5577048d65 in run_trans_callbacks_internal (cb_lst=0x7f5560abbd50, type=2, trans=0x7f5560abbce0, params=0x7fff3712acb0) at t_hooks.c:286 core.31977 : #0 0x00007f038e66fdee in free_cell (dead_cell=0x7f037811ebe0) at h_table.c:157 core.31994 : #0 0x00007f038e6ab65a in t_forward_nonack (t=0x7f037811ebe0, p_msg=0x7f038fdca438, proxy=0x0, proto=0) at t_fwd.c:1762 core.3700 : #0 0x000000000061697a in fm_insert_free (qm=0x7f19861fa000, frag=0x7f19864d53e8) at mem/f_malloc.c:245 core.3727 : #0 0x00000000006167a4 in fm_extract_free (qm=0x7f19861fa000, frag=0x7f19864f4cc8) at mem/f_malloc.c:208 I don't know how to proceed now. Can you help me to highlight the issue here? Thank you. Regards, Igor.

3 13

Re: [SR-Users] Auth module Issue:UA getting challanged thrice when load is high
by Daniel-Constantin Mierla 16 Oct '15

16 Oct '15

Hello, are there any error messages in the syslog? Maybe setting debug level for auth modules to 3 and looking at logs will help to reveal what is happening. Be sure syslog is in async mode, otherwise it will slow down processing. Cheers, Daniel On 16/10/15 08:42, Surendra.P wrote: > > Hello Daniel, > > > > I have a concern regarding auth module, I am running > 2000 TPS for 1000000 subscribers, using SIPP. What I observed when > you run small load like 500 TPS, everything get passed, but for 2000 TPS > > With same subscriber.csv file some of the subscribers getting > challenged. May I this why auth module is behaving like this.. below > are the parameters which I configured in auth module > > > > modparam("auth", "nonce_expire", 3600), rest all parameters default > values. > > > > Regards > > Surendra > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

routing based on TLS auth
by Max 16 Oct '15

16 Oct '15

Hi. I've got bunch of sip gateways to use and bunch of asterisk instances connecting to my Kamailio over TLS (own CA used). I'd like to match source with destination based on TLS authentication: * all the calls from asterisk A (TLS cert A) allowed via provider A * all the calls from asterisk B (TLS cert B) allowed via provider B ... * all the incoming calls from provider A forwarded to asterisk A (TLS cert A) * all the incoming calls from provider B forwarded to asterisk B (TLS cert B) ... * calls not coming from provider X or via TLS dropped. Providers are regular VoIP gateways (login:password, connection via SIP). I'm pretty sure this is possible with Kamailio but I'd appreciate any examples and pointers to modules I should use to implement this. Or detailed explanation of why this is a bad idea. Also, I'm not sure where it's better to place user database - should it be separate on per-asterisk basis? Central in Kamailio? No user db needed at all? cheers, Max.

2 1

Auth module Issue:UA getting challanged thrice when load is high
by Surendra.P 16 Oct '15

16 Oct '15

Hello Daniel, I have a concern regarding auth module, I am running 2000 TPS for 1000000 subscribers, using SIPP. What I observed when you run small load like 500 TPS, everything get passed, but for 2000 TPS With same subscriber.csv file some of the subscribers getting challenged. May I this why auth module is behaving like this.. below are the parameters which I configured in auth module modparam("auth", "nonce_expire", 3600), rest all parameters default values. Regards Surendra

1 0

Strange TLS problem
by Pete Kelly 14 Oct '15

14 Oct '15

Hi I am trying to use the [client] directives in tls.cfg for the first time. For my configuration I need to connect to 2 servers, both via TLS and both using different certs. I have defined an entry in tls.cfg like [client:1.2.3.4:5061] [client:5.6.7.8:5071] and I also have an entry in dispatcher tables for each server using the URI format sip:1.2.3.4:5061;transport=tls With this configuration, I expect dispatcher to send OPTIONS to each server, and Kamailio to resolve the host in dispatcher to the correct client section of tls.cfg and negotiate a connection. However this is not happening. Dispatcher module is trying to send OPTIONS as expected, and it is using TLS as expected but it is not using the correct [client] section from tls.cfg. Instead it is always falling back to [client:default], and if that is not present it then tries to look for some internal Kamailio defaults. Has anyone come across this issue before? Debug logs don't yield anything useful, however my suspicion is that Kamailio may be trying to match "sip:1.2.3.4:5061;transport=tls" from dispatcher module to "1.2.3.4:5061" from tls.cfg. Any help or advice would be appreciated Regards Pete

2 2

setflag and hash table variables
by Asgaroth 14 Oct '15

14 Oct '15

I originally sent this to sr-dev in error, I am resending to sr-users instead, thanks Hi All, Quick question, is it possible to use variables in the setflag function, I am trying to set a flag that is stored in a hash table using the setflag function, but I keep getting syntax errors. I am using kamailio 4.3.3. Here's what I am trying to do: htable core_setid defines a dispatcher setid key and its value is associated with a flag as follows: id:key:key_type:value_type:value:expires 1:102:0:1:7:0 so, simple hash table type with an integer key of value 7 and no expiry In dispatcher I have the following modparams set, so that ds_is_from_list() will set them if found: modparam("dispatcher", "setid_pvname", "$var(ds_setid)") modparam("dispatcher", "attrs_pvname", "$var(ds_setid_attrs)") Then in my routing logic I have the following code block: if ( ds_is_from_list() ) { if ( $sht(core_setid=>$var(ds_setid)) != $null ) { setflag( $sht(core_setid=>$var(ds_setid)) ); return; } else { send_reply("403", "No flag set for source"); exit; } } When I test this using kamailio -c -f <filename> I get the following error: 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: syntax error 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: missing '(' or ')'? 0(20629) : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 21-52: bad command: missing ';'? 0(20629) : <core> [cfg.y:3435]: yyerror_at(): parse error in config file /shared/kamailio/etc/routing_logic.cfg, line 175, column 53: bad command ERROR: bad config file (4 errors) If I change the setflag line to setflag(7), then I dont get the error. I was wondering if its possible to have hashtable variables as parameters to the setflag function, or am I going about this in the wron way. Any thoughts would be appreciated. Thanks

2 3

debian jessie and dependencies
by nadie 14 Oct '15

14 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello. If you install mariadb instead mysql Oracle, and then, you want to install kamailio with kamailio-mysql-modules: No se satisfacen las dependencias de los siguientes paquetes: mariadb-client-core-10.0 : Entra en conflicto: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Rompe: virtual-mysql-client-core que es un paquete virtual. mysql-client-5.5 : Rompe: virtual-mysql-client que es un paquete virtual. Rompe: virtual-mysql-client-core que es un paquete virtual. mariadb-server-core-10.0 : Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. mariadb-client-10.0 : Rompe: mysql-client pero se va a instalar 5.5.44-0+deb8u1. Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Rompe: virtual-mysql-client que es un paquete virtual. mariadb-server-10.0 : Rompe: mysql-client-5.5 pero se va a instalar 5.5.44-0+deb8u1. Las acciones siguientes resolverán estas dependencias Eliminar los paquetes siguientes: 1) mariadb-client-10.0 2) mariadb-client-core-10.0 3) mariadb-server 4) mariadb-server-10.0 5) mariadb-server-core-10.0 Thanks -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJWGzvRAAoJEG5DH1ilMeeeLl4P/iLDNNIhW73Sducag6IuJaM6 VWeKhCpvkEGxUUiAfL5Byktme058W6d1y7KDUyBzzMLZysNBxm4kJAX4fZjOQBpO 6e1np/qYXC/1A54hAKrjynPIMFpofCG3W6I2wCGp9nKHhq/ail++SW7aFqht6nus yF9h/0mJjv8wNkZuzf0BGuZdTJZUqVBbAOoWNNZHVfqQ20IjbAL2rgg/ahXLt8tv BoYVBnCQAAActmiA2ETF13R0ioJY5gLUorn2IW/sq/TQyXiABaskmxY68d/z2svc DmIB3JaCpOOMeccU4Ay2A6zmo+HuN5LcMuPu0b+sk3aHSWn6Wl4JWGozjdneiNpp +ksNTJYhJm6dCvhsj4RIw+ALf0/mDF14YV+n0z6OiaG5V1pEt7Z7nGyVBY3uDHR3 Xm9oYkj3CKYW5nOgpimHir9f+1qD457AWMRkHyIJrOXbth0lSodUg+KbAw1mUci5 D1E5bf47M7+CbmAi0t7uOSKh2J+oQq4pW3SzZYQbrnWQnIkwpmFy72Oqp7GBiSD7 rxxKUt0xBIVjnzGQzCgQdLfeYJt3sZ8V9cHvnEe7Tte2hLRLdnqskOhmMkGxNavJ ZKA7McTjWFyXJNJAJAowtBLHbarIxo5oREAma1nJ42xLv9KyIhXpVao472lVJZLz V9kMzZksd4yhvAHH2eyi =oZca -----END PGP SIGNATURE-----

2 4

Kamailio vs RTPEngine Performance
by M S 13 Oct '15

13 Oct '15

Hi, I was asked some interesting questions yesterday by a rather techy client related to Kamailio and RTPEngine performance. Q1: For Kamailio we have say 8 child processes, which means there are 8 threads per listen address to process sip messages, i.e. for example 8 INVITE messages at any instance of time can be processed by Kamailio. If say, one INVITE takes 200ms to be processed and send out to destination, then we can estimate processing power of kamailio to be 40 INVITE requests per second per listen address, right? Q2: If so, then assuming each INVITE requires to engage RTPEngine. How can we ensure that RTPEngine would be capable to handle these 40 INVITE requests? The only relevant parameter I see in RTPEngine daemon script is "NUM_THREADS" (default value 5). What does this represent? Does this analogs to children as in Kamailio? Q3: How does RTPEngine manages SRTP <=> RTP translation (e.g. if one endpoint is WebRTC and other is traditional SIP)? My understanding is that RTPEngine has a kernel module (assuming kernel module is installed) which also manages this conversion besides forwarding the media packets. The Linux kernel already has encoders / decoders for nearly all encryption algorithms which are utilized by RTPEngine for doing the conversion in kernel space, right? if not then how it is done? Q4: Continuing to Q2, If these NUM_THREADS process actual media packets (RTP or SRTP) then are these synchronous or asynchronous? I think these are asynchronous, just want to confirm (otherwise RTPEngine won't process more then e.g. 5 calls at a time). So, assuming asynchronous how many packets can be queue to each thread? This would help estimating RTPEngine throughput using various codecs in calls. (e.g. assuming G.711 codec we have 50pps, if each thread queue size is 1000 then each thread can process 1000/50=20 concurrent calls and whole RTPEngine would process 20x5=100 concurrent calls). Thank you.

1 0

Capturing Kamailio retransmissions in routing script
by Olle E. Johansson 13 Oct '15

13 Oct '15

Hi! Is there a simple way to log outbound retransmissions by tm in the routing script? I would like to be able to see retransmissions. /O

2 1

evapi tcp parsing??
by Jayesh Nambiar 13 Oct '15

13 Oct '15

Hello, I'm exploring the evapi module for my kamailio to interface with an external node.js app for third party stuff like AAA, billing engine tasks, notifications and so on. I followed and took some ideas from the rtjson and evapi tutorial found here( http://kb.asipto.com/kamailio:k43-async-sip-routing-nodejs) to build the node.js app consuming events. When I stress tested the scenario using SIPp and tried sending a lot of events at 300-350cps from Kamailio, I noticed that at times the client is receiving 2-3 events in a single message together although I do event_sync_relay once per SIP message received and have netstrings enabled. I believe this is a typical behavior of TCP and needs to be handled by the client using some kind of Netstring handler. Please correct me if I'm wrong. And hence I'd like to know what particularly needs to be taken care of while writing a client that is listening for events on raw tcp socket and how does kamailio handle this situation while receiving messages over TCP socket?? Does kamailio recognize the end of netstring properly on evapi:message-received and give exactly one message to take care of on every "message-received" event or should that be handled in the script somewhere !! I also referred cgrates client over evapi example which is written in GO, but I couldnt find them handling TCP streams clearly either. I'd really appreciate some expert suggestion here to make an informed decision on using the evapi module for a large scale solution. Thanks, - Jayesh

3 26

How to get runtime configuration?
by Magnus Nordström 13 Oct '15

13 Oct '15

Hi, Is there any command to get the runtime configuration from kamailio? I tried "kamctl monitor" but there was not a lot of information. For example I need information about what tcp_conn_wq_max is set to. Best regards, Magnus Nordström

3 2

Re: [SR-Users] Lost 180 Ringing Response
by John Billings 13 Oct '15

13 Oct '15

> Hello, > > look at: > > http://kamailio.org/docs/modules/stable/modules/tm.html#tm.p.cancel_b_method > > Maybe one or another value for that parameter is suiting better your use > case. Thank you for the suggestion. This looked promising, but the cancel_b_method parameter is already at the default value of one. I also tried changing the value to two and got a similar result. This may sound crazy, but it almost seems like the problem happens only when kamailio is processing unrelated INVITES at the same time.

2 1

Re: [SR-Users] MySQL Connect Errors
by John Billings 13 Oct '15

13 Oct '15

2 1

Persistence of enable/disable for uacreg
by DanB 13 Oct '15

13 Oct '15

Hey Guys, I was wondering how does enable/disable flag for uac registration gets persistence, since I could not spot it in the mysql tables. Is that just temporary in-memory and on next restart/refresh does not longer take effect? Thanks, DanB

2 2

rfc: event route when kamailio starts
by Daniel-Constantin Mierla 12 Oct '15

12 Oct '15

Hello, based on a recent discussion with Alex Balashov (at Kazoocon this week), I pushed a patch that enables the execution of event_route[core:worker-one-init] when kamailio is started. It would be more like an alternative for event_route[htable:mod-init], for the cases when htable is not used, but also should ensure that all the modules are initialized. Moreover, because most of the modules close connection in mod init, I though of doing it when child one (udp receiver) initialization for all modules is finished, which means that all is ready for runtime, still this is before receiving any packet. On the other hand, forking is done by main process, so other children can start listening for sip. More or less what I want to ask here is whether the forking of the other children should wait until the event_route[core:worker-one-init] is finished (e.g., what is done in this event route is important before any sip packet is routed) or just leave it like it is. Worth to say that some modules do special processing in child init of worker one, like usrloc module loads the records from database to memory, which means that overall start of kamailio could be slower. Also, if there are time consuming actions in the event_route[core:worker-one-init], getting to the stage of routing sip packets will take longer. What would you find important, fork the other processes so routing sip will happen as soon as possible, or block until execution ofevent_route[core:worker-one-init] is finished. I guess that making it cusomizable in some way would be a solution, but knowing what use cases would be important may help in reducing the complexity to get it enabled. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

3 2

Dialog keep alive options are with error parsing headers
by José Seabra 12 Oct '15

12 Oct '15

Hello there, I'm using dialog module to send keep alive options to callee and caller in order to detect if dialog still up, but when kamailio generates the OPTION and parse it, gives the following errors: [root@c4-compute02-srv1 kamailio]# 37(1587) DEBUG: dialog [dlg_hash.c:614]: dlg_lookup(): ref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_hash.c:616]: dlg_lookup(): dialog id=2210 found on entry 2350 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to caller 37(1587) DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=< sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [dns_cache.c:3293]: dns_srv_sip_resolve(): (62.28.197.25, 0, 0), ip, ret=0 37(1587) DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 14692 37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 *37(1587) ERROR: <core> [parser/msg_parser.c:96]: get_hdr_field(): ERROR: get_hdr_field: bad header* 37(1587) DEBUG: <core> [parser/msg_parser.c:280]: get_hdr_field(): get_hdr_field: error exit *37(1587) INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [VC* *]* 37(1587) DEBUG: topoh [topoh_mod.c:232]: th_prepare_msg(): parsing headers failed37(1587) DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: 37(1587) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <OPTIONS> 37(1587) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: < sip:unknown@62.28.197.25:5060> 37(1587) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> 37(1587) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK4693.2af96e95000000000000000000000000.0>; state=16 37(1587) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 37(1587) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 37(1587) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=90d40279ee5745d8851aa989e6fadfc2 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [64]; uri=[sip:unknown@62.28.197.6] 37(1587) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:unknown@62.28.197.6>] 37(1587) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <0> <OPTIONS> 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=e34d43e164ad4b7392dc9c6e1058a983 37(1587) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 37(1587) DEBUG: dialog [dlg_cseq.c:215]: dlg_cseq_msg_sent(): traking cseq updates 37(1587) DEBUG: dialog [dlg_hash.c:671]: internal_get_dlg(): ref dlg 0x7f89134c1d40 with 1 -> 4 37(1587) DEBUG: dialog [dlg_hash.c:674]: internal_get_dlg(): dialog callid='13208317017c42128fbf00e1685ee1b3' found on entry 2350, dir=2 37(1587) DEBUG: dialog [dlg_cseq.c:233]: dlg_cseq_msg_sent(): request not going downstream (2) 37(1587) DEBUG: dialog [dlg_hash.c:846]: dlg_unref(): unref dlg 0x7f89134c1d40 with 1 -> 3 37(1587) DEBUG: dialog [dlg_req_within.c:452]: dlg_send_ka(): keep-alive sent to caller 37(1587) DEBUG: dialog [dlg_req_within.c:427]: dlg_send_ka(): sending OPTIONS to callee I'm using the following dialog configuration: modparam("dialog","db_url", DBURL) modparam("dialog","db_mode", 0) modparam("dialog","enable_stats", 1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","profiles_with_value","concurrentcalls") modparam("dialog", "detect_spirals", 1) modparam("dialog","dlg_match_mode", 2) modparam("dialog", "initial_cbs_inscript", 0) modparam("dialog", "default_timeout", 3600) modparam("dialog", "send_bye", 1) modparam("dialog", "timeout_avp", "$avp(s:timeout_dlg)") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) I'm also using topoh module. Anyone can help me to identify if I'm doing the proper configuration? Thank you for your support Best Regards José Seabra

3 9

Startup Route in Kamailio
by Sunil More 12 Oct '15

12 Oct '15

Hi All, I am new to Kamailio world. I am trying to initialize some details at Kamailio startup, this requires some database queries from a few tables. I have read about the htable event, however sql queries are not working in this route. I have come upon following errors. Please suggest a solution for this. Route used: event_route[htable:mod-init] { xlog("L_INFO", "********Initailising the startup route ******\n"); sql_xquery("cb","SELECT id, media_server_port, name FROM domains","ra"); } error : Oct 8 13:28:32 P172 /usr/local/kamailio/sbin/kamailio[28735]: INFO: <script>: ********Initailising the startup route ****** Oct 8 13:28:32 P172 kamailio: ERROR: <core> [daemonize.c:303]: daemonize(): Main process exited before writing to pipe Thanking You, Sunil More Ph : 9503338275

2 1

Familiar topic about Kamailio with Asterisk behind NAT
by Dirk Teurlings - SIGNET B.V. 09 Oct '15

09 Oct '15

Hi, I found some existing topics on this but failed to get a solutions out of them. We're running into some issues with client devices connecting to our private addresses. The way it is setup now: CLIENTS <-> (NAT) <-> INTERNET <-> KAMAILIO(4.2.5) with RTPPROXY(v1) <-> PRIVATE LAN <-> ASTERISK (v1.8) Our Kamaialio and Asterisk are in a private address range, but Kamailio also has a public interface. Most of the clients (about 95%) work well with this setup, but a couple don't. We have one case now where the CLIENT tries to connect to the private address of ASTERISK. And of course, that doesn't work. I'm kind of stuck as to where I need to fix this. I tried using the externaddr option in Asterisk to solve it on that end. But that didn't help anything. The NAT options in Kamailio are not really suited for this, as they tend to fix client NAT problems. Any pointer or help would be greatly appriciated. Cheers, Dirk

8 16

[pua] Question to DB Mode 0
by Maciej Marczyński 09 Oct '15

09 Oct '15

Hi Guys, I have a question regarding pua module parameter 'db mode'. I run Kamailio (4.3.2) with sqlite and I tried to set the parameter to 0 but even then I received: > ERROR: db_sqlite [dbase.c:489]: db_sqlite_commit(): sqlite commit failed: NOT NULL constraint failed: pua.call_id > ERROR: db_sqlite [dbase.c:138]: db_sqlite_cleanup_query(): finalize failed: NOT NULL constraint failed: pua.call_id > ERROR: pua [pua.c:1179]: db_update(): while inserting in db table pua > > As far as I checked the documentation, it says that '0' means: > high speed memory based storage Does it still writes to DB anyway (on timer or whatever?) or just keep it in memory? Is there any way to disable writing to DB? The second thing is the reason of the ERRORs above. Why does pua think that my call_id is null? If you need any more information please let me know. Best regards and have a good weekend, Maciej

3 7

Siremis debug window
by Infinicalls Infinicalls 08 Oct '15

08 Oct '15

Hi, I have installed Siremis and modified it for public user registration. I have added a new user. When I try to log in it opens up a new (debug) window and prints "1". It doesn't allow me to log in. But I can log in as admin;admin. And siremis/user/register doesn't load properly. It redirects to the login page again. Kindly let me know how do I solve this. regards Ganesh Kumar

2 8

MySQL Connect Errors
by John Billings 08 Oct '15

08 Oct '15

We are using kamailio 4.2.0 with auth_db enabled and connected to a MySQL server. Occasionally, in the middle of the night when there is no activity we get the following errors out of the blue (the last three lines are repeated indefinitely). According to the MySQL site <http://dev.mysql.com/doc/refman/5.0/en/error-lost-connection.html>, this can be caused by a connect timeout that is too short (We verified that Aborted_connects is steadily rising). The kamailio server is right next to the database server so there is no network latency. We increased the db_mysql timeout_interval parameter, but we still see MySQL’s Aborted_connects counter incrementing. So it’s like kamailio is still timing out on connection attempts. Ideas? Thanks. Oct 1 02:34:49 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:34:49 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:34:55 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:01 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'reading authorization packet', system error: 11 Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:07 sip-proxy1 /usr/sbin/kamailio[9260]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Host ‘X.X.X.X' is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts' Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: auth_db [authorize.c:175]: get_ha1(): failed to query database Oct 1 02:35:11 sip-proxy1 /usr/sbin/kamailio[9264]: ERROR: db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Host 'X.X.X.X' is blocked because of many connection errors; unblock with 'mysqladmin flush-hosts' ...

2 1

TM not ending transactions
by Daniel Tryba 08 Oct '15

08 Oct '15

Running 4.3.x (both .1 and .2) I have seen it happening twice sofar that dialogs (the module) will not end, resulting in to calls failing due to call limits based on active dialogs. Accounting logged a BYE for all these calls. kamcmd tm.stats shows an increasing number of current/waiting transactions (as grphed in attachment). Has anyone seen this problem?

3 7

Questions about LCR
by Scott C. Fertig 08 Oct '15

08 Oct '15

Hello, Sorry to bother and if I sound like a newb thats because I am kind of new (but im trying =) ) I am having some issues getting started using LCR and was wondering if anyone could point me in the right direction and/or let me know what I am doing wrong. I began setting up the other day using kamailio (4.4.0-dev5) from git (maybe I picked the wrong version?) and am struggling to get lcr to work properly. So far I have been able to get the module to load, added the route to the config and have added some LCR values. Inbound calling is working perfectly to a registered device which is great so far. This is my config for the LCR: http://pastebin.com/LUF0Pgr1 I have the following values in my database for LCR: http://pastebin.com/bDpQug0J I was not sure if dumping all the text in the email would be appropriate if needed please let me know to resend with the configs in the email. Whats strange to me is that even with the xlog I do not get any messages to syslog at all, I have defined WITH_DEBUG, debug level at 3, and log facility set to LOG_LOCAL0 (Sorry if I'm bringing up multiple issues) Anyways back on point, I'm wondering if this is related to the avp's listed, I am not sure what the avps referenced are supposed to represent, when I try to dump all the avps I get a empty list, and it is querying the usr_preference table for these. Is there something I need to add within this table, or is there perhaps something wrong with my configuration? [root@localhost kamailio]# kamctl avp list database engine 'MYSQL' loaded Control engine 'FIFO' loaded Query: SELECT uuid,username,domain,attribute,type,value FROM usr_preferences ; Dumping AVPs Thank you!

2 2

Lost 180 Ringing Response
by John Billings 07 Oct '15

07 Oct '15

Hi Everyone, We are using Kamailio 4.2.0 with TLS. It’s setup so multiple phones can register to the same account and they all ring together. It always works great unless one of the phones is on a slower Internet connection. Occasionally that phone will start ringing after another phone has already picked up the call. The problem is that the INVITE to the slow phone is not followed by a CANCEL. So the phone keeps on ringing for a long time. When there is no problem, the logs on the phone show: Receive: INVITE Send: 180 Ringing Received: CANCEL Send: 200 Ok When the phone keeps ringing: Delay: up to several seconds Receive: INVITE Send: 180 Ringing The problem never seems to happen if I wait until all of the phones have received the INVITE. So is the late 180 response not getting handled by the right session? Any ideas on how to track this down?

2 1

Re: [SR-Users] uac_req_send and reply timeout
by Paweł Sternal 07 Oct '15

07 Oct '15

On 07.10.2015 16:41, Daniel-Constantin Mierla wrote: > Hello, > > in the event route for uac_req_send() is practically during handling the > reply for what was sent before. Also, the uac_req_send() sends something > out of the processed sip message, so typically functions won't work > because they will be applied to the processed sip message. > > Maybe try setting the avp with the timeout value (see tm parameters), > might work, I haven't checked -- report the results, would be good to know. Unfortunately solution with fr_timer_avp also doesn't work. Only set globally fr_timer works. Well... there is always a solution. Thanks anyway. > Cheers, > Daniel Regards, Paweł

2 1

uac_req_send and reply timeout
by Paweł Sternal 07 Oct '15

07 Oct '15

Hi, It is posibility to set timeout for uac_req_send with setting "evroute" to 1. Yes, I found a solution without this set when using event_route[tm:local-request] [1] but with "evroute" option this don't work ofcorse. regards Pawel [1] http://lists.sip-router.org/pipermail/sr-users/2011-October/070695.html

2 1

SIREMIS Help
by Vojimir Raovic 07 Oct '15

07 Oct '15

Hi, Just wanted to see how to get SIREMIS user manaual? Regards, Voyimir [image: me]*Vojimir Raovic*, Principal Engineer & Co-founder *Office:* +61 8 9200 3500 | *Web:* www.norwoodsystems.com | *Twitter: * @norwoodsystems <http://twitter.com/norwoodsystems> *Direct:* +61 8 9200 3503 | *Mobile:* +61 450 019 775 [image: me][image: me] [image: me] NOTICE: If you are not an authorised recipient of this e-mail, please contact Norwood Systems immediately by return e-mail or by telephone on +61 8 9200 3500. In this case, you should not read, print, re-transmit, store or act in reliance on this e-mail or any attachments, and should destroy all copies of them. This e-mail and any attachments are confidential and may contain legally privileged information and/or copyright material of Norwood Systems or third parties. You should only re-transmit, distribute or commercialise the material if you are authorised to do so. Any legal privilege attaching to the contents of this e-mail or any attachments is expressly reserved and is not waived by any transmission to an unauthorised recipient. This notice should not be removed. *Please consider the environment before printing this e-mail.*

2 1

Recall: Kamailio B2BUA or Dual router ?
by Mark Henriksen 07 Oct '15

07 Oct '15

Mark Henriksen would like to recall the message, "[SR-Users] Kamailio B2BUA or Dual router ?".

1 0

Kamailio B2BUA or Dual router ?
by Mark Henriksen 07 Oct '15

07 Oct '15

Hello All I have been looking for answers and trying really hard to figure out if I can use Kamailio as a Dual Router (Inbound and Outbound using 2 IPs on 1 Eth "Amazon AWS") I can not seem to find information is this is possible using Kamailio or if am just plain "stupid" and can not read the Wiki well enough J Hope you can help //Mark

2 2

Kamailio as a LB with Asterisk and consul
by Sylvain Boily 06 Oct '15

06 Oct '15

Hello, I did a new small project with Asterisk to register it on consul (http://consul.io) to use a discovery service tools. The project is here : https://github.com/sboily/asterisk-consul-module This last day, i'm working to use kamailio as a loadbalancer for Asterisk and autoscale it. You can find example on my docker-compose.yml, contribs directory and README. It's not perfect, because i'm clearly not a kamailio expert, but it works. So, i'm looking about feeling to this dev, and i would expect some help from kamailio community to configure kamailio as a LB better ;) I will be at astricon, so if people want to share with me about this project, contact me. Have a good day Sylvain

2 2

creating debian package from source
by Thibault Gueslin 06 Oct '15

06 Oct '15

I have difficulties to install the debian package of kamailio I have build from source Creation works fine. (make cfg, configuration of proper debian in pkg/debian/deb , make deb) But kamailio tries to find module here : /usr/lib/x86_64-linux-gnu/kamailio/modules/ and modules are installed in /usr/lib64/ Of course, it works fine by specifying -L with actual path. I have tried also to download the source of debian package with same result. I have missed some steps. Thibault

1 0

Http authentification with JSON-RPC command
by Loïc Boissy 06 Oct '15

06 Oct '15

Hello, I would like to authorize JSON-RPC commands for a specific user. Is it possible to enable HTTP authentification on the xHTTP module? Thanks for your help. Best regards Loic BOISSY

3 3

Re: [SR-Users] diameter auth with round trip
by JB 06 Oct '15

06 Oct '15

Thanks for the suggestion about auth_diameter, do you think there would be an interest in the community if we were to do this modification on ims_auth or auth_diameter ? could it be integrated to trunk ? We do not want to publish our own fork of ims_auth or auth_diameter with that small change and maintain the code repository if it can be avoided, so it would be nice if we can put it in the trunk and forget about it. Plus it is more open source oriented to publish to main branch i feel :) JB 2015-10-03 23:53 GMT+02:00 JB <jbf.nospam(a)gmail.com>: > Hello all, we are working on a SIP solution using Kamailio. > > We want to secure our base of user credentials even in case of attack on > the SIP server, and for that reason we plan to use diameter authentication > as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt > > Paragraph 6.2 describes a mode where the HSS answer with code > > DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. > > > This does NOT corresponds to what is done on Kamailio module ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) > > ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) > > Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? > > Thank you > > JB > >

2 1

diameter auth with round trip
by JB 06 Oct '15

06 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio module ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

2 1

Prepend values to PSTN gateway?
by Infinicalls Infinicalls 05 Oct '15

05 Oct '15

Hi, I am able to successfully connect to a PSTN gw from Kamailio. Now, he says I need to prepend a value (eg.123123) before the number(ENUM) and then only terminate the calls. Where do I add this? Can somebody help me? thanks Ganesh Kumar

3 3

Interoperability issues between browsers and mobile phones to place a call.
by Safdar Khan 05 Oct '15

05 Oct '15

Hi Daniel sir I have enabled web-sockets module into my kamailio sip server.I am trying to make call from web browsers and mobile phones. To make a call from web browser i am using sipml5 and as a mobile app i am using imsdroid. However i have done some search on net and i found nothing relevant. Any suggestions and help?.Waiting your response eagerly. Thank you

2 3

defaults missing from tcp ops module readme
by Juha Heinanen 05 Oct '15

05 Oct '15

after seeing some commits on tcpops module, i decided to check its readme and noticed that nothing is mentioned about the default values of parameters. it might be useful to tell that, e.g., if tcp_keepalive_enable is not set, then tcp_keepalive core parameter value tells what the status is. -- juha

2 2

problem with dispatcher load balancing
by Ding Ma 05 Oct '15

05 Oct '15

I'm trying to set up kamailio dispatcher to distribute calls to 2 asterisk servers. So far, the failover case seems ok, but I cannot get the dispatcher to distribute load. All calls are going to the last destination entry in the dispatcher table even if I have set the maxload attributes. I'm using algorithm 10 for load distribution. The number of calls sent to one asterisk is well above the maxload. Hope someone more experienced in dispatcher can review my config and give me some recommendations. Thanks in advance. Here is the content of dispatcher table in postgresql db. select * from dispatcher; id | setid | destination | flags | priority | attrs | description ----+-------+--------------------+-------+----------+---------------------------+------------- 1 | 1 | sip:10.0.1.31:5061 | 0 | 0 | duid=asterisk1;maxload=25 | Asterisk1 2 | 1 | sip:10.0.1.33:5061 | 0 | 0 | duid=asterisk2;maxload=25 | Asterisk2 (2 rows) Here are the dispatcher sections in kamailio.cfg. # ----- dispatcher params ----- #!ifdef WITH_DISPATCHER modparam("dispatcher", "db_url", DBASTURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "force_dst", 1). # If flag 2 is set, then failover support is enabled. modparam("dispatcher", "flags", 3) # the last address in destination set is used as a final option to send the request to modparam("dispatcher", "use_default", 1) # load balancing fail over modparam("dispatcher", "dst_avp", "$avp(dsdst)") modparam("dispatcher", "grp_avp", "$avp(dsgrp)") modparam("dispatcher", "cnt_avp", "$avp(dscnt)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:dispatcher@localhost") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshhold", 10) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) #!endif route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing #!ifdef WITH_DISPATCHER if(is_method("BYE|CANCEL") && ds_is_from_list("1", "3")) ds_load_update(); #!endif ...... } } route[FROMASTERISK] { #!ifdef WITH_DISPATCHER if(ds_is_from_list("1", "3")) { xlog("L_DBG","$rm from $fU@$si:$sp: Call from Asterisk cluster\n"); return 1; } return -1; #!else if ($si==$sel(cfg_get.asterisk.bindip)) { return 1; } return -1; #!endif } onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); #!ifdef WITH_DISPATCHER if(is_method("INVITE") && ds_is_from_list("1", "3")) { if(status=~"2[0-9][0-9]") { ds_load_update(); } else if(status=~"[3-7][0-9][0-9]") { ds_load_unset(); } } #!endif if(status=~"[12][0-9][0-9]") route(NATMANAGE); } #!ifdef WITH_DISPATCHER failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { # mark the destination Inactive and Probing ds_mark_dst("IP"); # select the new destination if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } else { # last available node failed to reply, no other destinations available send_reply("404", "No destination"); exit; } } } #!endif route[TOASTERISK] { #!ifdef WITH_DISPATCHER # ds_mark_dst("IP"); # Call load distribution if(!ds_select_dst("1", "10")) { sl_send_reply("500", "Service Unavailable"); xlog("L_INFO","$rm from $fU@$si:$sp: No destinations available for $rd\n"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); #!else $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); #!endif route(RELAY); exit; } # Dispatcher detects a destination goes down event_route[dispatcher:dst-down] { xlog("L_ERR", "Destination down: $rm $ru ($du)\n"); } # Dispatcher detects a destination comes up event_route[dispatcher:dst-up] { xlog("L_ERR", "Destination up: $rm $ru\n"); }

2 10

Changed the bindport of Kamailio but still listening to older port
by amjad ali 04 Oct '15

04 Oct '15

Hi, I am scratching my head for one month to get this sorted and I have spent so many hours but couldn't figure out. Please help ! I have changed the bind port to 5099 but it is still listening to the previous port even I have rebooted the system. My Code is : kamailio.bindip = "192.168.1.105" desc "Kamailio IP Address"kamailio.bindport = "5099" desc "kamailio port" However, when I restart kamailio service it shows that it is still listening to 5060 (attached the screen shot) Regards,Amjad

2 1

Kamailio is not forwarding the call to asterisk
by amjad ali 04 Oct '15

04 Oct '15

Hi, I have kamailio and asterisk running on same machine. When I make an internal call, it routes the call to other extension. However, if I stop the asterisk service then the call still routes the same way to other extension. Is there a chance that kamailio does the call routing it self. Also, I cannot see any sip extension registered at asterisk when I run a command "Sip show peers" neither there is any activity in Asterisk's log. Regards,Amjad

2 1

Inconsistent string interpolation
by Sven Neuhaus 02 Oct '15

02 Oct '15

Hello, I'm occasionally running into issues with inconsistent string interpolation in the kamailio.cfg. Here's the latest example. Take this line: $var(x) = "sip:+$var(abc)-$var(def)-$si@$Ri:$Rp"; Here, the values of the variables will NOT be interpolated into the string. Instead, I have to use this: $var(x) = "sip:+" + $var(abc) + "-" + $var(def) + "-" + $si + "@" + $Ri + ":" + $Rp; However, when I have an instruction such as: append_hf("P-Asserted-Identity: $var(x)\r\n"); Then string interpolation takes place (i.e. the value of $var(x) is put into the string. This is very confusing and error prone. Is it something that can be fixed? The behaviour should be consistent. -Sven

2 6

Kamailio v4.3.3 Released
by Daniel-Constantin Mierla 02 Oct '15

02 Oct '15

Hello, Kamailio SIP Server v4.3.3 stable release is out. This is a maintenance release of the latest stable branch, 4.3, that includes fixes since release of v4.3.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.3.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.3.3. For more details about version 4.3.3 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/10/kamailio-v4-3-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

[ims_auth] compute authenticate field in diameter server, not kamailio
by JB 02 Oct '15

02 Oct '15

Hello all, we are working on a SIP solution using Kamailio. We want to secure our base of user credentials even in case of attack on the SIP server, and for that reason we plan to use diameter authentication as described in RFC http://www.rfc-base.org/txt/rfc-4740.txt Paragraph 6.2 describes a mode where the HSS answer with code DIAMETER_MULTI_ROUND_AUTH ,and then validate user credential after a second round trip. This does NOT corresponds to what is done on Kamailio ims_auth, where credentials (actually a hash of the credentials, but its enough to authenticate ) ) are pushed to kamailio, which does the computation of the expected answer (which corresponds to par 6.3 of the RFC 4740) Is there any kamailio module that would allow to use the method with DIAMETER_MULTI_ROUND_AUTH ? Thank you JB

1 0

reSIProcate v1.10.0 released
by Daniel Pocock 02 Oct '15

02 Oct '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA256 The v1.10.0 release has been tagged and made available for download: https://www.resiprocate.org/files/pub/reSIProcate/releases/ and a direct link to the ChangeLog: https://github.com/resiprocate/resiprocate/blob/resiprocate-1.10/ChangeLog Debian, Fedora and Ubuntu packages are in the process of being updated. It is about 20 months since 1.9.0 and there have been many improvements. Highlights of the v1.10.0 release: - - repro SIP proxy now includes: - full presence server - PostgreSQL support - Include directive in the config file - - enhancements to the stack and proxy include: - WebSocket authentication may now use URL parameters instead of cookies, client certs or DIGEST - better Syslog support - TLS improvements (PFS, private key passphrases) - better assertion handling (resip_assert) - WSAPoll for Windows, allowing thousands of TCP connections - removing transports at runtime - added netns (Linux network namespace) support The RTC Quickstart Guide provides an overview of how to get started using repro, Kamailio and other popular SIP, XMPP and TURN products http://rtcquickstart.org Contributors to v1.10.0 include Alan Hawrylyshen, Alexander G. Pronchenkov, Byron Campen, Daniel Pocock, Daniel Tacalau, Dan Petrie, Dario Bozzali, Fedor Brunner, Gregor Jasny, Kevin Green, Reiner Herrmann, Robert Sparks, Scott Godin, Thomas Thorne, Vasil Kolev and Wolfgang Rosenauer. Please also see the hall of fame in Github: https://github.com/resiprocate/resiprocate/graphs/contributors which emphasizes the enormous and sustained contribution Scott Godin has made to this project and this release in particular. We are also very grateful to all those who contributed feedback through the mailing list, testing the beta packages in Debian and all past contributors to previous releases of the reSIProcate project. reSIProcate point releases are now announced on the low-volume resiprocate-announce mailing list, please subscribe if you would like to know when 1.10.1 comes along: http://list.resiprocate.org/mailman/listinfo/resiprocate-announce SHA256 checksums for the files: resiprocate-1.10.0.tar.gz c089686e183b8a739064731fa9e5002ab4d0bd6908cd8f5e648e021d76363c98 resiprocate-1.10.0.zip df4ba3902770390e7a0f9f6b13c5959292feba04785f01b9d044ad31325fa1e0 resiprocate-contrib-1.10.0.tar.gz 2f40c568644cbf1dd8628da0535dadd8d17367860234163d003fe0fbd75ba9a0 resiprocate-contrib-1.10.0.zip ef1c50b7c70b533c7bec1bfb071ca19ff9368a0b1e8a951e430e273b8d413715 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1 iQIcBAEBCAAGBQJWDlraAAoJEOm1uwJp1aqDCxgP/jlJBE0fTSp1FzohA5IRmrmQ rVQcKOB6pI6rWTMEwv8WuBs6sA7z/z94b2jbLYZRo/iwXXb85TbQXqnIQkyw5GAs l+F73B+rPyNYN46CZvjYv/rrscW8rxDPoein/t7uEzfATm1xgAZrIykWhZx42WV2 QUEjZDAXRm2Q2duIBfnxyTmhyFurWgHJhtxLzx6zCVnml/hWt8xslzkH1WYtfXuI 0ltGqNuVlYlhHCi2/BpaxHVdsPELhdiTHpIu88UgBSHI5gKiUBp38dhQttwJr+bt VWF9Ri+KQPbn7d5JGc8mPMeBTlYmJUgFY4s//eIU2pYRcxZ2w1/iim2SHewMpift 51C9Lmp6vSBycpCNPAQ6yQqJdAjIlTlY9eKJ2td+bCUw66Zlx10h3tqY4ka95o0s TfFporu6dpaIu3TSo+qVW8fhXuYmRh86NHLLi3+LCbZ9i6j5E71qor5mkz9x/xLM 7kWbVJKNJFTdbTsVjAKZ7uh4YFfkn3NcYEu+5zwF5Fm/z6ohE36sme12Lo0khjyq kTDhefATpFvZLxu2l0RBmG+NktOj/H2MJvbPVGPyEtHaam1BJaRPv4UMh6bldoR4 vsrt8cL31HhnWGTZS7M7/yg39Ge+JrTAHQD8wpdVh6nBzFTqb2n2WIk2swPICrLA v8lx2bBfol6NYvB0GhAT =R1HL -----END PGP SIGNATURE-----

1 0

Siremis-4.2.0 - error with Alias and login page
by Infinicalls Infinicalls 02 Oct '15

02 Oct '15

Hi, I tried installing siremis-4.2.0 following the step given.The Alias was not working. So, I installed it in the /html folder and the installation was completed. But the login page looked strange. There was no logo displayed and the submit button was not there. Mine is apache 4.2 on Ubuntu LTS 14. Kindly let me know how to solve this problem. thanks Ganesh Kumar

2 2

how to use data received from diameter answer(CCA) in kamailio config script
by SureshTummala 02 Oct '15

02 Oct '15

Hi, I want retrieve the data present in the Credit-Control-Answer response from diameter in kamailio configuration script. Is there any provision in kamailio 4.1.x (or) later? Any suggestions are appreciated. Regards,Suresh Tummala

4 3

FW: Your iptel.org Registration
by Bijal Desai 02 Oct '15

02 Oct '15

When I call on sip:bijaldesai@iptel.org, it is not working. Can you please guide me -----Original Message----- From: registrar(a)iptel.org [mailto:registrar@iptel.org] Sent: Tuesday, September 29, 2015 10:00 PM To: Bijal Desai <Bijal(a)ssminfotech.com> Subject: Your iptel.org Registration Thank you for registering with iptel.org. We are reserving the following SIP address for you: sip:bijaldesai@iptel.org To finalize your registration please check the following URL within 24 hours: http://serweb.iptel.org/user/reg/confirmation.php?nr=473f1e2fc74bfb3e9d28eb… (If you confirm later you will have to re-register.) Here are the details of your account: --------------------------------------------------------------------- Username: bijaldesai(a)iptel.org Password: Angel1987@ Email: bijal(a)ssminfotech.com Full Name: Bijal Desai Sip address: sip:bijaldesai@iptel.org (You can always retrieve these via the "Forgot Password" link on the signup page) -- This message is automatically generated by SerWeb.

2 1

Kamailio-Asterisk real-time integration with Dynamic IP
by Rudy Eschauzier 02 Oct '15

02 Oct '15

All, The Kamailio-Asterisk integration that is described here asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site] is limited to Asterisk servers with a fixed IP address. The reason is that Kamailio identifies and authorizes traffic from Asterisk based on the source IP address. Attached is a patch to kamailio.cfg that adds support for dynamic Asterisk hosts. It introduces the option to define a fully-qualified domain name instead of the IP address to identify the Asterisk server. The host name can be dynamically updated to always point at the (changing) IP address of the Asterisk server. Defining WITH_ASTERISK_FQDN enables the new option. When WITH_ASTERISK_FQDN is set, Kamailio will perform a DNS search on the host name in the asterisk.bindhost variable. It will authorize incoming traffic if it originates from one of the IP addresses the DNS lookup returns (and matches the asterisk.bindport variable as before). Furthermore, the patch introduce the option to automatically set the IP address of the server that Kamailio is running on. This does away with the need to properly define kamailio.bindip on every server that the configuration is running on. This option is enabled by defining WITH_AUTOIP. With the suggested changes I have been able to successfully run an Asterisk server in my home, having a dynamic public IP address and connecting to the Kamailio proxy on a VPS server with a static IP addres. Please let me know if you have any comments or suggestions. Kind regards,Rudy Eschauzier. | | | | | | | | | | | asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [...Table of Contents Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database Architecture Registration Call Initiation Requirements MySQL Insta... | | | | View on kb.asipto.com | Preview by Yahoo | | | | |

2 1

cnxcc module
by Максим Панчоха 02 Oct '15

02 Oct '15

Hello! I'm trying to set max call duration, but after 5 seconds call is continues. Kamailio version: 4.3.2 Here's part of my config file: #!define CC_FLAG 29 ... modparam("cnxcc", "credit_check_period", 1) modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") modparam("dialog", "hash_size", 2048) modparam("dialog", "default_timeout", 3600) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) ... if ((is_method("INVITE")) && (is_request()) && (!has_totag())) { route(CNXCC); } route[CNXCC] { setflag(DLG_FLAG); if (!is_method("INVITE")) return; $var(client) = $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$var(client)", "$var(max_time)")) { xlog("Error setting up credit control"); return; } } event_route[dialog:failed] { xlog("L_INFO", "********END CALL*********$DLG_lifetime*******************"); } event_route[dialog:end] { xlog("L_INFO", "******END CALL************$DLG_lifetime****************"); } event_route[dialog:start] { xlog("L_INFO", "***********************DIALOG STARTED****************************"); } event_route[cnxcc:call-shutdown] { $var(duration) = $TS - $dlg(start_ts); xlog("L_INFO", *****END CALL***BALANCE IS EMPTY************$var(duration)*************"); } Tell me please, where I made a mistake. Thank you for a response.

2 1

ERROR: mediaproxy [mediaproxy.c:1978] #347
by Ngoc 02 Oct '15

02 Oct '15

Hello supporter, I took a look kamailio and get the error: Sep 29 11:28:37 lvps46-163-74-198 /usr/local/sbin/kamailio[1738]: ERROR: mediaproxy [mediaproxy.c:1978]: EngageMediaProxy(): engage_media_proxy requires the dialog module to be loaded and configured. How to resolved this problem, please help me. Regards, --------------------------- Dinh Hong Ngoc Technical Lead Edge-works Software Co.,ltd Hall 5, Quang Trung Software City, Tan Chanh Hiep Ward, Dist. 12, Ho Chi Minh City, Vietnam. Email: ngoc.dinh(a)edge-works.net Phone: +84 8 54371075 | Mobile: +84 909 911 406 Site: http://www.edge-works.net

2 1

receive_fd(): EOF
by jery john 02 Oct '15

02 Oct '15

11(12454) : <core> [mem/q_malloc.c:149]: qm_debug_frag(): BUG: qm_*: fragm. 0x7f9e8fcb2448 (address 0x7f9e8fcb2478) end overwritten(c0c0c000, abcdefed)! 13(12456) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 20 0(12443) ALERT: <core> [main.c:784]: handle_sigs(): child process 12454 exited by a signal 6 0(12443) ALERT: <core> [main.c:787]: handle_sigs(): core was generated 0(12443) INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD 13(12456) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 12(12455) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 9(12452) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 10(12453) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 8(12451) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 4(12447) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 3(12446) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 7(12450) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 6(12449) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 5(12448) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 2(12445) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 1(12444) INFO: <core> [main.c:850]: sig_usr(): signal 15 received 0(12443) INFO: <core> [sctp_core.c:48]: sctp_core_destroy(): SCTP API not initialized using kamailio 4.2 with modified tcp_read.c and tcp_main.c files for buffer manipulation both files attached gdb output Reading symbols from /usr/local/sbin/kamailio...done. warning: exec file is newer than core file. [New LWP 15502] warning: Error reading shared library list entry at 0x79e8c789480056e4 warning: Error reading shared library list entry at 0x482a50b70fd8458b Core was generated by `/usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -P /var/run/ka'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000061f73d in _shm_resize (p=0x61f729 <_shm_resize+541>, s=32767, file=0x3f <error: Cannot access memory at address 0x3f>, func=0x0, line=-1433800800) at mem/shm_mem.c:111 111 DBG("WARNING:vqm_resize: resize(0) called\n");

2 1

blf problem
by huseyin kalyoncu 01 Oct '15

01 Oct '15

Hi I am trying to implement BLF functionality as described in http://kb.asipto.com/kamailio:presence:k43-blf. I am using the same conf with the one in the tutorial except that i am using postgres database instead of mysql. The problem is BLF is not working with this config (presence server and proxy server are in the same kamailio with same IP). But when I separate presence server from proxy server (2 different kamilio boxes with different IPs) BLF working perfect. i can see PUBLISH & NOTIFY messages in the tcpdump output with the working configuration. But i can not see any PUBLISH messages in the tcpdump output with the original config (one with presence and proxy servers are same). My question in the normal working environment (when BLF working perfectly); should i see PUBLISH and NOTIFY messages in the tcpdump output when the presence and proxy servers are the same? thanks Huseyin

2 9

SIP and conntrackd experience?
by Grant Bagdasarian 01 Oct '15

01 Oct '15

Hello, Does anyone have experience with SIP and conntrackd? Our media and signaling is separated. Does this work for VPN connections where SIP is sent over the VPN but the media over public internet? Thanks, Grant

2 5

modifying $fu / $fU (R-URI From)
by Sven Neuhaus 01 Oct '15

01 Oct '15

Hello, I'm having some troubles with modifying $fU in my kamailio.cfg. The following line appears to have no effect: $fU = $(fU{s.substr,2,0}); I would expect it to remove the first two leading characters from $fU. Has anyone else seen this? I have the same issue with modifying $fu. We're not using the uac module. And we're using the latest stable version of Kamailio. Thanks, -Sven

4 4

out of memory Error
by Yasin CANER 01 Oct '15

01 Oct '15

3 15

Crash kamailio 4.3.1
by Thibault Gueslin 30 Sep '15

30 Sep '15

After upgrading to kamailio 4.3.1 and modifying the config, I get this crash very often: ... CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 21 ALERT: <core> [main.c:728]: handle_sigs(): child process 9341 exited by a signal 11 ALERT: <core> [main.c:731]: handle_sigs(): core was not generated INFO: <core> [main.c:743]: handle_sigs(): terminating due to SIGCHLD INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received INFO: <core> [main.c:794]: sig_usr(): signal 15 received ... Is it known ? or may be related to my config ? Thank you Thibault

3 23

db_text and caching/reloading
by Asgaroth 30 Sep '15

30 Sep '15

Hi, I am busy testing the db_text module for some modules who's configuration does not change that often. If I set the db_mode for db_text to 0 (caching), it appears that if I modify the file directly (in this case the address dbtext file) and then issue a "kamctl address reload" the updated contents are *not* reloaded. However, if I set db_mode to 1, then the contents of the db_text file automatically reload when I change it on the file system. Is it possible to run db_text in caching mode but to also force a reload if I manually update the underlying database file? Additionally, in kamctlrc, I had to add the following variable: DBTEXT_PATH="/etc/kamailio/db_text" Whereas the original example for db_text path in kamctlrc is as follows: # database path used by dbtext, db_berkeley or sqlite #DB_PATH="/etc/kamailio/db_text" Is this just a typo in the kamctlrc distribution or is the variable meant to be DB_PATH or DBTEXT_PATH? Thanks

1 3

ACK and BYE are forwarded statelessly (failed to parse Route HF)
by Miroslav Dunaev 30 Sep '15

30 Sep '15

Hello everyone, I have a setup where kamailio acts as UAC to a SIP provider. Kamailio is also configured to pass all incoming calls to an Asterisk. Here is a diagram: Phone<-->Asterisk(10.10.11.232)<--->(PRIV:10.10.11.238)Kamailio(PUB:44.44.44.148)<--->(194.213.29.92)SIP-provider. When a call comes in from the SIP-provider, it gets all the way to the phone. But when the SIP-provider sends BYE or ACK (during hungup event or 200OK/SDP asterisk response.). I get this errors in my kamailio log: siputils [checks.c:100]: has_totag(): totag found <core> [parser/parse_rr.c:49]: do_parse_rr_body(): parse_rr_body(): No body for record-route rr [loose.c:106]: find_first_route(): failed to parse Route HF rr [loose.c:929]: loose_route(): There is no Route HF tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed ..... .... .... siputils [checks.c:100]: has_totag(): totag found tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=2 , global msg id=1 , T on entrance=0xffffffffffffffff tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=10101, isACK=1 tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found tm [t_funcs.c:279]: t_relay_to(): SER: forwarding ACK statelessly here is the link to: kamailio.cfg, ngrep -i any capture, debug_level6 log: http://labmir.com/kamailio/ Looking at the SIP ACK that comes from SIP provider I can see that they use strict routing. (i.e. TO field is set to kamailio's public IP that is next hop to them). Maybe I am wrong to judge that its not loose routing just because of the TO field. Anyways, because of (failed to parse Route HF) error, ACKs and BYEs are forwarded by kamailio statelessly, as a result kamailio uses wrong interface to forward them (SIP_dailog.txt line:233) and therefore asterisk never sees ACKs or BYEs. How can I make kamailio recognize ACKs and BYEs belonging to a dialog and pass them statefully through the right interface? It is worth to mention that I use exactly the same config with a different SIP-provider (which is peered to our PUB IP not registered through UAC like in this case) and it works flawlessly. Thank you, Mir

2 1

Anybody in Beijing interested in Kamailio related project cooperation ?
by Thierry Luo 29 Sep '15

29 Sep '15

很报歉通过这个邮件列表打扰大家。我们是一个创业团队，刚刚拿到投资，正在组建核心团队，现需要一位在OpenIMSCore （或Kamailio/OpenSIPS）方面有经验的C/C++开发工程师加盟，待遇优厚，期权可谈，职位在北京。请有意者速与 luoyongheng(a)nane.cn <mailto:luoyongheng@nane.cn> 联系。谢谢！打扰见谅！罗

1 0

Best practices for troubleshooting deadlocks?
by Alex Balashov 29 Sep '15

29 Sep '15

Hello, Very occasionally, we encounter what appear to be deadlocks in all UDP receiver threads. All Kamailio processes are running, but no SIP messages are being processed. On one of our high-volume installation, this happens extremely infrequently -- maybe once every month or two. On these occasions, the operator restarts the proxy before we get a chance to go in and figure out what's going on. So, I'm trying to provide the operator with a procedure to execute prior to restarting the proxy on these occasions, so that we can see a snapshot of where the receiver threads are stuck. As far as I can tell, unless Kamailio itself segfaults, there's no specific PID that one can attach GDB to in order to get an overhead snapshot of all the child processes. Here's what I came up with: --------------------------------------------- #!/bin/bash kamcmd -s /tmp/kamailio_ctl ps > thread_log.txt echo >> thread_log.txt while read PID; do gdb --pid=$PID<<EOF>>thread_log.txt set print elements 0 thread apply all bt full generate-core-file detach EOF done < <(kamcmd -s /tmp/kamailio_ctl ps | grep 'udp receiver' | awk '{print $1}') --------------------------------------------- As far as I can tell, this should give me the most ample visibility into the state of the threads, with further core dumps to inspect if follow-up is needed. Hopefully this will result in some fixes back to the project. However, if there are any other suggestions for information to grab in such a scenario, I'm all ears. Thanks in advance! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

Kazoo db_url required?
by Asgaroth 29 Sep '15

29 Sep '15

Hi, I am doing some tests with the kazoo module, I only want to use it for some amqp related tasks. In our particular case, we do not require kazoo to hook into the presence tables, however, if I do *not* set the db_url parameter for kazoo, then kamailio fails to start with the following error: /usr/sbin/kamailio[11557]: CRITICAL: kazoo [kazoo.c:361]: mod_child_init(): child_init: database not bound /usr/sbin/kamailio[11557]: ERROR: <core> [sr_module.c:900]: init_mod_child(): Error while initializing module kazoo (/usr/lib64/kamailio/modules/kazoo.so) Is it possible to run the kazoo module without specifying the db_url parameter, we only need it for sending/recieving amqp messages. Additionally, I noticed that the amqp_connection parameter has a max limit of 50 characters, is it possible to increase this at all? I had a case where my url was 64 characters long and kazoo would fail with max character limit exceeded for the module parameter. Any tips much appreciated. Thanks

1 2

Kamailio recvfrom out of bounds
by José Seabra 28 Sep '15

28 Sep '15

I have one kamailio server listening in 2 network interfaces, kamailio receives sip messages from interface A and then send it to destine through interface B. When we start kamailio service then run gdb for each kamailio network interface process, execute the command "bt full" the result is the following: *tmp = 0x5f801b994c4715bd <Address 0x5f801b994c4715bd out of bounds>* interface - A bt full #0 0x00007fb01bfa40c3 in recvfrom () from /lib64/libc.so.6 No symbol table info available. #1 0x000000000047f2ba in udp_rcv_loop () at udp_server.c:421 len = 13 buf = '\000' <repeats 65535 times> tmp = 0x5f801b994c4715bd <Address 0x5f801b994c4715bd out of bounds> from = 0x7faf9bf38bb8 fromlen = 16 ri = {src_ip = {af = 2309178400, len = 32765, u = {addrl = {5458819, 4284160}, addr32 = {5458819, 0, 4284160, 0}, addr16 = {19331, 83, 0, 0, 24320, 65, 0, 0}, addr = "\203KS\000\000\000\000\000\000_A\000\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {80026686, 0}, addr32 = {80026686, 0, 0, 0}, addr16 = {7230, 1221, 0, 0, 0, 0, 0, 0}, addr = ">\034\305\004", '\000' <repeats 11 times>}}, src_port = 55576, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 32687, sa_data = "\000\000\200D\243\211\375\177\000\000\213\337D"}, sin = {sin_family = 32687, sin_port = 0, sin_addr = {s_addr = 2309178496}, sin_zero = "\375\177\000\000\213\337D"}, sin6 = {sin6_family = 32687, sin6_port = 0, sin6_flowinfo = 2309178496, sin6_addr = {__in6_u = {__u6_addr8 = "\375\177\000\000\213\337D\000\000\000\000\000p", <incomplete sequence \372\233>, __u6_addr16 = {32765, 0, 57227, 68, 0, 0, 21616, 39930}, __u6_addr32 = {32765, 4513675, 0, 2616874096}}}, sin6_scope_id = 32687}}, bind_address = 0x7faf9becd7d8, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #2 0x0000000000525eae in main_loop () at main.c:1573 i = 0 pid = 0 si = 0x7faf9becd7d8 si_desc = "udp receiver child=0 sock=62.28.197.4:5060 \000\000\000\000\000\000\001\000\000\000\000\000\000\000\260\244r\023n\000\000\000\071\263q\000\000\000\000\000\020\243q\000\000\000\000\000\b\000\000\000\001\000\000\000\260\373\371\233\257\177\000\000\220E\243\211\375\177\000\000|\351O\000\000\000\000\000\000\266\253\023\257\177\000\000\260\244r\023\257\177\000" ---Type <return> to continue, or q <return> to quit--- nrprocs = 32 __FUNCTION__ = "main_loop" #3 0x000000000052c395 in main (argc=11, argv=0x7ffd89a34848) at main.c:2533 cfg_stream = 0x2296010 c = -1 r = 0 tmp = 0x7ffd89a35f61 "" tmp_len = 0 port = 0 proto = 32688 options = 0x725010 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1641321572 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd89a34710 p = 0x7fb01bf4c646 <intel_check_word+166> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \305> __FUNCTION__ = "main" interface B bt full #0 0x00007fb01bfa40c3 in recvfrom () from /lib64/libc.so.6 No symbol table info available. #1 0x000000000047f2ba in udp_rcv_loop () at udp_server.c:421 len = 45 buf = '\000' <repeats 65535 times> tmp = 0x13cb2f24296000e9 <Address 0x13cb2f24296000e9 out of bounds> from = 0x7faf9bf38bb8 fromlen = 16 ri = {src_ip = {af = 2309178400, len = 32765, u = {addrl = {5458819, 4284160}, addr32 = {5458819, 0, 4284160, 0}, addr16 = {19331, 83, 0, 0, 24320, 65, 0, 0}, addr = "\203KS\000\000\000\000\000\000_A\000\000\000\000"}}, dst_ip = {af = 2, len = 4, u = {addrl = {2764381246, 0}, addr32 = {2764381246, 0, 0, 0}, addr16 = {7230, 42181, 0, 0, 0, 0, 0, 0}, addr = ">\034Ť", '\000' <repeats 11 times>}}, src_port = 56136, dst_port = 5060, proto_reserved1 = 0, proto_reserved2 = 0, src_su = {s = {sa_family = 32687, sa_data = "\000\000\200D\243\211\375\177\000\000\213\337D"}, sin = {sin_family = 32687, sin_port = 0, sin_addr = {s_addr = 2309178496}, sin_zero = "\375\177\000\000\213\337D"}, sin6 = {sin6_family = 32687, sin6_port = 0, sin6_flowinfo = 2309178496, sin6_addr = {__in6_u = { __u6_addr8 = "\375\177\000\000\213\337D\000\000\000\000\000p", <incomplete sequence \372\233>, __u6_addr16 = {32765, 0, 57227, 68, 0, 0, 21616, 39930}, __u6_addr32 = {32765, 4513675, 0, 2616874096}}}, sin6_scope_id = 32687}}, bind_address = 0x7faf9becda08, proto = 1 '\001'} __FUNCTION__ = "udp_rcv_loop" #2 0x0000000000525eae in main_loop () at main.c:1573 ---Type <return> to continue, or q <return> to quit--- i = 0 pid = 0 si = 0x7faf9becda08 si_desc = "udp receiver child=0 sock=62.28.197.164:5060 \000\000\000\000\001\000\000\000\000\000\000\000\260\244r\023n\000\000\000\071\263q\000\000\000\000\000\020\243q\000\000\000\000\000\b\000\000\000\001\000\000\000\260\373\371\233\257\177\000\000\220E\243\211\375\177\000\000|\351O\000\000\000\000\000\000\266\253\023\257\177\000\000\260\244r\023\257\177\000" nrprocs = 32 __FUNCTION__ = "main_loop" #3 0x000000000052c395 in main (argc=11, argv=0x7ffd89a34848) at main.c:2533 cfg_stream = 0x2296010 c = -1 r = 0 tmp = 0x7ffd89a35f61 "" tmp_len = 0 port = 0 proto = 32688 options = 0x725010 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ---Type <return> to continue, or q <return> to quit--- ret = -1 seed = 1641321572 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7ffd89a34710 p = 0x7fb01bf4c646 <intel_check_word+166> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \305> __FUNCTION__ = "main" [root@c4-srv1 kamailio]# kamctl stats shmem shmem:fragments = 2552 shmem:free_size = 2082764040 shmem:max_used_size = 66449600 shmem:real_used_size = 64719608 shmem:total_size = 2147483648 shmem:used_size = 54355992 Can anyone explain the meaning of *tmp = 0x5f801b994c4715bd <Address 0x5f801b994c4715bd out of bounds>* Best Regards José Seabra

2 6

Kamailio with Radius authentication
by Ciolpan Ionut-Marian 28 Sep '15

28 Sep '15

Hi, I want to implement Kamailio with Radius authentication. To be more exactly I want to register SIP users (PhonerLite and/or Jitsi) based on the credentials stored in radcheck table from radius database. First of all, I have tried to register SIP users based on the credentials stored in users file from /etc/raddb directory. In order to do this,I am using auth_radius module (you can find the content of my kamailio.cfg file in the attachements) but without success. All of the SIP users are registeres successfully even if they are not specified in users file. Can you provide me some help to achieve this task? and also, can you explain to me how should I do this authentication based on the credentials stored in radcheck table? The content of users file in /etc/raddb: usertest@mydomain_name Auth-Type := Digest, User-Password == "*89Lob?p" Reply-Message = "Authenticated" Thank you for any help you can provide ! Best regards PS: kamailio 4.0.5 freeradius 2.2.0 radclient 2.2.0 PPS: I managed to register SIP users based on the credentials stored in subscriber table from kamailio db.

2 10

Releasing Kamailio v4.3.3
by Daniel-Constantin Mierla 28 Sep '15

28 Sep '15

Hello, I am considering to release Kamailio v4.3.3 by end of this week, most likely of Friday, Oct 2. As usual, announce any issues that you are aware and not reported to sr-dev or bug tracker. Also, if some patch was no 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

1 0

GeoIP and multiple rtp
by Infinicalls Infinicalls 28 Sep '15

28 Sep '15

Hi, I have installed Kamailio in one instance and want to have multiple RTP instances for geo loadbalancing. I can understand that rtp_cluster will do that based on number of parameters, but geo_IP is missing. Can somebody tell me how do I go about with it? Is it possible to get with Kamailio's geoip module and "instruct" rtp? regards Ganesh Kumar

3 2

passing multiple values in mqueue
by Thibault Gueslin 28 Sep '15

28 Sep '15

I want to pass several arguments to a mqueue to process them asynchronously. (like in the example of sip sweet). How could it be done ? Other possibility is to generate a string and parse it in the lua script. But there should something more simple ? thibault

2 1

MSRP over Websockets
by vamsi 25 Sep '15

25 Sep '15

Hi, I am trying to configure MSRP over websocket and using Crocodilr-msrp client for connecting the same but when I try to transfer any file it gives me the below error. Any thoughts is appreciated. I am able to do audio, vedio and messaging using SIP-JSSIP client and working fine. Error - Log -------------- Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: tls [tls_server.c:208]: tls_complete_init(): Using initial TLS domain TLSc<default> (dom 0x7f35500d4858 ctx 0x7f355013d620 sn []) Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: tls [tls_domain.c:701]: sr_ssl_ctx_info_callback(): SSL handshake started Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: ERROR: <core> [tcp_main.c:2654]: tcpconn_1st_send(): connect 10.205.64.24:8084 failed (RST) Connection refused Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: ERROR: <core> [tcp_main.c:2664]: tcpconn_1st_send(): 10.205.64.24:8084: connect & send for 0x7f35501bca20 failed: Connection refused (111) Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: <core> [tcp_main.c:2858]: tcpconn_chld_put(): destroying connection 0x7f35501bca20 (24, -1) flags 0060 Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: ERROR: msrp [msrp_netio.c:178]: msrp_relay(): forwarding frame failed Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: exec: *** cfgtrace:request_route=[DEFAULT_EVENT] c=[/usr/local/etc/kamailio/kamailio.cfg] l=1202 a=2 n=exit Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: <core> [tcp_read.c:1153]: ws_process_msg(): WebSocket Message: [[>>>#012??#011?i#036'?$Mu?Ik#022?#001,I?#005>t?'Z*?=q#012?#010jO?IsT?#031m#035?F/#027?[.#022?_*#011?]$#037?Q*#010?G/#027?\&#011?G+#021?]*#020?^*#034?#012n#007?#032lW?S1#010?Y0#025?\0#021?G,#023?Q.#037?Fm#011?Y*#022?G)#011?P)#021?\(#021?RjD?IsT?#031$#010?<<<]] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [22] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0x7f3550140930] refcnt [2] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_frame.c:415]: decode_and_validate_ws_frame(): decoding WebSocket frame Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_frame.c:456]: decode_and_validate_ws_frame(): supported non-control frame: 0x2 Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_frame.c:555]: decode_and_validate_ws_frame(): Rx (decoded): MSRP u5xh2ntl SEND#015#012To-Path: msrps:// 10.205.64.24:8084/s.10458.4.561447274;tcp msrps:// 10.205.64.24:8084/s.10453.7.297645663;tcp msrp://i64fn3tc.invalid:2855/g14o60krz6;ws#015#012From-Path: msrp://grtsrfd0.invalid:2855/505z40k5b1;ws#015#012Message-ID: 3652053096.cr4l1e1q#015#012Success-Report: yes#015#012Failure-Report: yes#015#012Byte-Range: 2049-4096/68240#015#012Content-Type: application/vnd.openxmlformats-officedocument.wordprocessingml.document#015#012#015#012 Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_frame.c:671]: ws_frame_receive(): Rx MSRP frame:#012MSRP u5xh2ntl SEND#015#012To-Path: msrps://10.205.64.24:8084/s.10458.4.561447274;tcp msrps://10.205.64.24:8084/s.10453.7.297645663;tcp msrp://i64fn3tc.invalid:2855/g14o60krz6;ws#015#012From-Path: msrp://grtsrfd0.invalid:2855/505z40k5b1;ws#015#012Message-ID: 3652053096.cr4l1e1q#015#012Success-Report: yes#015#012Failure-Report: yes#015#012Byte-Range: 2049-4096/68240#015#012Content-Type: application/vnd.openxmlformats-officedocument.wordprocessingml.document#015#012#015#012 Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_conn.c:382]: wsconn_put(): wsconn_put start for [0x7f3550140930] refcnt [2] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: websocket [ws_conn.c:415]: wsconn_put(): wsconn_put end for [0x7f3550140930] refcnt [1] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:171]: msrp_parse_fline(): MSRP FLine: [1] [MSRP] [u5xh2ntl] [SEND] [1] [] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c0d1b8) [To-Path] [2] [msrps://10.205.64.24:8084/s.10458.4.561447274;tcp msrps:// 10.205.64.24:8084/s.10453.7.297645663;tcp msrp://i64fn3tc.invalid:2855/g14o60krz6;ws] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c0c918) [From-Path] [1] [msrp://grtsrfd0.invalid:2855/505z40k5b1;ws] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c0ed80) [Message-ID] [4] [3652053096.cr4l1e1q] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c0ee10) [Success-Report] [7] [yes] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c1a868) [Failure-Report] [0] [yes] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c0c9a8) [Byte-Range] [5] [2049-4096/68240] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:327]: msrp_parse_headers(): MSRP Header: (0x7f3558c107d8) [Content-Type] [8] [application/vnd.openxmlformats-officedocument.wordprocessingml.document] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_parser.c:356]: msrp_parse_headers(): MSRP Body: [2050] [[#012#012]] Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: msrp [msrp_mod.c:468]: msrp_frame_received(): executing event_route[msrp:frame-in] (3) Sep 23 23:11:36 ip-10-205-64-24 kamailio[10460]: DEBUG: <core> [parser/msg_par *********** Kamailio.cfg **************** #!ifdef WITH_MSRP event_route[msrp:frame-in] { msrp_reply_flags("1"); if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS)) && $Rp != MY_MSRP_PORT) { xlog("L_WARN", "MSRP request received on $Rp\n"); msrp_reply("403", "Action-not-allowed"); exit; } if (msrp_is_reply()) { msrp_relay(); } else if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } if (!www_authenticate("MY_DOMAIN", "subscriber", "$msrp(method)")) { if (auth_get_www_authenticate("MY_DOMAIN", "1", "$var(wauth)")) { msrp_reply("401", "Unauthorized", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } if ($hdr(Expires) != $null) { $var(expires) = (int) $hdr(Expires); if ($var(expires) < MSRP_MIN_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Min-Expires: MSRP_MIN_EXPIRES\r\n"); exit; } else if ($var(expires) > MSRP_MAX_EXPIRES) { msrp_reply("423", "Interval Out-of-Bounds", "Max-Expires: MSRP_MAX_EXPIRES\r\n"); exit; } } else { $var(expires) = MSRP_MAX_EXPIRES; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); $shtex(msrp=>$var(sessid)) = $var(expires) + 5; # - Use-Path: the MSRP address for server + session id $var(hdrs) = "Use-Path: msrps://MY_IP_ADDR:MY_MSRP_PORT/" + $var(sessid) + ";tcp\r\n" + "Expires: " + $var(expires) + "\r\n"; msrp_reply("200", "OK", "$var(hdrs)"); } else if ($msrp(method)=="SEND" || $msrp(method)=="REPORT") { if ($msrp(nexthops)>1) { if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if ($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "Session-does-not-exist"); exit; } else if ($msrp(method)!="REPORT") { msrp_reply("200", "OK"); } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); } else { msrp_reply("501", "Request-method-not-understood"); } } Thanks, Vamsi

2 3

Kamailio & VoLTE (powered by ng-voice) at ITU Telecom World in Budapest 2015
by Carsten Bock 25 Sep '15

25 Sep '15

Hi Everyone, We will have our own small LTE (and VoLTE enabled) network at the ITU Telecom World 2015 in Budapest. Since we are on the exhibitor list, we are pleased to share a complimentary Exhibition Visitor pass for you to visit our booth there (no. P07 in Hall F). Register at http://telecomworld.itu.int/take-part/attend/ following the guidelines, embedding the promotional code C-00001092 when prompted. Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Schomburgstr. 80 D-22767 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-0 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

RPC ul.dump
by Loïc Boissy 25 Sep '15

25 Sep '15

Hello, As mentionned in the doc, ul.dump command dumps the content of the location table. I use MULTIDOMAIN domain configuration, and i would like to use a RPC command to recover all contact present in my database (for all domains). Is there a solution to do that? Thanks you for you help. Loic BOISSY -- Visit Digigram at IBC2015 at Stand 8.C51, Sept. 11-15 in Amsterdam Register here <http://www.ibc.org/page.cfm/link=478> for your *FREE* pass with Digigram's Exhibitor Customer Code *4462*

3 5

Kamailio Asterisk MWI
by Ansuman Manna - Asergis Telecom 25 Sep '15

25 Sep '15

Hi, I am working on MWI (message waiting indication). I am using Kamailio as SIP registrar, Asterisk is for call routing. I can send notify to Kamailio from my VM server but thing is Kamailio is not handling the notify. If you have some suggestion about this. Regards, Ansuman Manna Software Developer Asergis Telecom Services Private Limited Office Tel: +91 11 6606 8400 Office Fax: +91 112554 5156 http://www.asergis.in <http://www.asergis.in>

3 3

Ragarding reginfo handling in kamailio
by Ram Anji 25 Sep '15

25 Sep '15

Hi, I am facing issue with reginfo handling in kamailio, I am getting proper notify when user is registered and deregistered properly. But i am not getting same when user's registration is expired[with out refresh/ when entry is removed from location], Can anyone suggest me how can i achieve this. I am using 4.3.1 kamailio Regards, Raman

2 1

OpenSev -> Kamailio Migration?
by Matthew Harris 25 Sep '15

25 Sep '15

Hello everyone! I am new to the VOIP world and currently researching a solution to upgrading or migrating from an antiquated openser sip server to kamailio. Has anyone ever had to do this? If so: What challenges should I expect to face? Warmest regards, Matt

2 1

SIP account.
by Grendell 25 Sep '15

25 Sep '15

Hi A few says ago I registered an account user name 08031955. However I have been unable to connect using 2 different SIP phones. Can you please check if this account is active yet. Can you advise if my user name has to include ....(a)iptel.org on the end of it? many thanks Graham Rendell

2 1

regEvent feature
by Stefan Ljung 25 Sep '15

25 Sep '15

(A continuation on an old thread) In the RFC 3680 it is described how a UE can subscribe to its own registration state(s). As soon as it's own registration state changes in any way - like a refresch registration - a NOTIFY is send to the subscribing terminal. What made me hopeful was when I in the Overview's use case section for the module pua_reginfo read the line: "A client could subscribe to it's own registration-status, so he would get notified as soon as his account gets administratively unregistered." That's in line with how I would like to use it - Is it in any way possible to achieve that in the Kamailio? /Stefan From: Stefan Ljung Sent: den 18 februari 2015 09:32 To: miconda(a)gmail.com Subject: RE: [SR-Users] Event reg - questions >From the pua_regInfo module description: Use cases for this might be: Keeping different Servers in Sync regarding the location database Get notified, when a user registers: A presence-server, which handles offline message storage for an account, would get notified, when the user comes online. A client could subscribe to it's own registration-status, so he would get notified as soon as his account gets administratively unregistered. ... So it's the last line I want to deploy. Maybe Im over-interpreting "A client could subscribe to it's own registration-status" I want to be able to simulate (As close a possible) a VoLTE clients reg events handling. Only the part between to and from the UE. So no PUBLISH is needed - Kamailio needs to notify the UE about its registration status. /Stefan From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: den 16 februari 2015 17:43 To: Kamailio (SER) - Users Mailing List; Stefan Ljung Subject: Re: [SR-Users] Event reg - questions Hello, afaik, the pua_reginfo is for publishing details of location records to another sip server node, main purpose being location replication. I don't think it is something for an end UA. If you want publishing online/offline states for an user based on its registration state, look at pua_usrloc module. On 12/02/15 15:17, Stefan Ljung wrote: I made a test configuration for trying the pua_regEvent module. There were two things I want to ask about. First - when the SUBSCRIBE to event 'reg' following a UEs registration to the Kamailio is responded with a NOTIFY - there was no body in the NOTIFY. I expected a XML body with registration status. Second - When the UE re-registers to Kamailio - I expected a NOTIFY to be send on the existing 'reg' subscription, but nothing happened. Or am I missing some configuration ? Here's some of the conf script: [..] modparam("pua_reginfo", "publish_reginfo", 0) Btw, with the above parameter value, the module is no longer sending PUBLISH requests, in other words, it is not informing the other peers that there is some update to location records. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

Kamailio PSTN with User/Pass authentication
by Infinicalls Infinicalls 24 Sep '15

24 Sep '15

Hi, I am trying to add PSTN gw for terminating my calls. I have changed the IP in the kamailio.cfg file, but this provider wants digest auth and so I need to pass api/pass to them. Can somebody guide me how and where to add the username+pass to the config file. thanks regards Ganesh Kuamr

4 8

Re: [SR-Users] [Kamailio 4.2] Crash during burn-in test
by Daniel-Constantin Mierla 24 Sep '15

24 Sep '15

Hello, the first frames of the backtrace don't have the symbol table, do have other core files from where you can extract the bt full? Also, 4.2.3 is old in its series, newer versions in branch 4.2 were released. Can you update the latest version in branch 4.2 and reproduce again? That will rule out is not a side effect of an issue fixed after 4.2.3. You don't need to change anything in config or database, just grab the latest sources from branch 4.3 and install over the old binaries. Cheers, Daniel On 08/09/15 07:58, Jack Wang wrote: > Hi, > > The result of 'kamailio -v' is: > > version: kamailio 4.2.3 (arm/linux) 87ca6e-dirty > flags: STATS: Off, EXTRA_DEBUG, USE_TCP, USE_TLS, TLS_HOOKS, > USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, > USE_PTHREAD_MUTEX, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, > USE_DST_BLACKLIST, HAVE_RESOLV_RES > 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: 87ca6e -dirty > compiled on 16:15:00 Sep 4 2015 with arm-marvell-linux-gnueabi-gcc > > > It can be reproduced not only once, steps like wrote before. > > Following link is the result of `bt full` from the core file generated: > > https://gist.github.com/BroWater/dd759897f4b88672720a#file-gistfile1-txt > > FYI > > > > >Hello, > > > >can you give the exact version: kamailio -v? > > > >Also, the log message indicates that a core file was generated, sent the > >backtrace taken from gdb: > > > >gdb /path/to/kamailio /path/to/corefile > >bt full > > > >Could you reproduce the issue, or did it happen only once? > > > >Cheers, > >Daniel > > > >On 03/09/15 04:50, Jack Wang wrote: > >> > >> [Steps] > >> > >> 1. one MCU call multi-endpoints > >> 2. MCU hangup all endpoints > >> > >> repeatedly. And all through Kamailio proxy. > >> > >> [Results] > >> > >> For a while, Kamailio crashed. > >> > >> Followings are related logs: > >> > >> > >> ========================================== > >> > >> Sep 2 19:08:34 ./kamailio[3712]: : tm [t_fwd.c:1632]: > >> t_send_branch(): BUG: t_send_branch: retr. already started for > 0x41848fa0 > >> Sep 2 19:08:34 ./kamailio[3712]: ERROR: sl [sl_funcs.c:387]: > >> sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on > >> sending to next hop occurred (477/SL) > >> > >> At this point the related behavior is that Kamailio received two INFO > >> requests simultaneously. I wonder whether this is the cause. > >> And maybe it's related to the later crash? > >> > >> Sep 2 19:08:35 ./kamailio[3713]: CRITICAL: <core> [pass_fd.c:293]: > >> receive_fd(): EOF on 11 > >> Sep 2 19:08:35 ./kamailio[3703]: ALERT: <core> [main.c:784]: > >> handle_sigs(): child process 3704 exited by a signal 11 > >> Sep 2 19:08:35 ./kamailio[3703]: ALERT: <core> [main.c:787]: > >> handle_sigs(): core was generated > >> Sep 2 19:08:35 ./kamailio[3703]: : <core> [mem/f_malloc.c:586]: > >> fm_free(): BUG: fm_free: bad pointer 0x7239b6c4 (out of memory > >> block!), called from tm: h_table.c: free_cell(157) - aborting > >> > >> > >> Any suggestions? > >> > >> Thanks. > >> > >> ========================================== > >> > >> > >> _______________________________________________ > >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > >> sr-users at lists.sip-router.org <http://lists.sip-router.org> > >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- 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, Sep 28-30, 2015, in Berlin - http://asipto.com/u/kat

2 2

Re: [SR-Users] load 2 500 000 rows from dialplan
by Никитенко Виталий 24 Sep '15

24 Sep '15

>Hello > > >On 24/09/15 11:06, Никитенко Виталий wrote: >> Hi! >> I have kamailio-4.2.3. And in table dialplan there is 2 500 000 rows. When i started kamailio i see error >> >> ERROR: dialplan [dp_repl.c:74]: repl_exp_parse(): out of shm memory >> ERROR: dialplan [dp_db.c:393]: build_rule(): failed to compile replacing expression >> ERROR: dialplan [dp_db.c:111]: init_db_data(): failed to load database data >> ERROR: dialplan [dialplan.c:188]: mod_init(): could not initialize data >> ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module dialplan (/usr/local/lib64/kamailio/modules/dialplan.so) >> >> then i set modparam("dialplan", "fetch_rows", 2500000) and started kamailio i see >> >> ERROR: db_mysql [km_dbase.c:422]: db_mysql_fetch_result(): no memory left >> ERROR: dialplan [dp_db.c:231]: dp_load_db(): failed to fetch >> ERROR: dialplan [dp_db.c:111]: init_db_data(): failed to load database data >> ERROR: dialplan [dialplan.c:188]: mod_init(): could not initialize data >> ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module dialplan (/usr/local/lib64/kamailio/modules/dialplan.so) >> >> Please say me how i can load to memory 2500000 rows dialplan? >/usr/local/kamailio/sbin/kamailio -h | grep '\-[mM]' > -m nr Size of shared memory allocated in Megabytes > -M nr Size of private memory allocated, in Megabytes > >In you particular case, you need more shared memory (-M). > >Regards Thanks, it works!

1 0

load 2 500 000 rows from dialplan
by Никитенко Виталий 24 Sep '15

24 Sep '15

Hi! I have kamailio-4.2.3. And in table dialplan there is 2 500 000 rows. When i started kamailio i see error ERROR: dialplan [dp_repl.c:74]: repl_exp_parse(): out of shm memory ERROR: dialplan [dp_db.c:393]: build_rule(): failed to compile replacing expression ERROR: dialplan [dp_db.c:111]: init_db_data(): failed to load database data ERROR: dialplan [dialplan.c:188]: mod_init(): could not initialize data ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module dialplan (/usr/local/lib64/kamailio/modules/dialplan.so) then i set modparam("dialplan", "fetch_rows", 2500000) and started kamailio i see ERROR: db_mysql [km_dbase.c:422]: db_mysql_fetch_result(): no memory left ERROR: dialplan [dp_db.c:231]: dp_load_db(): failed to fetch ERROR: dialplan [dp_db.c:111]: init_db_data(): failed to load database data ERROR: dialplan [dialplan.c:188]: mod_init(): could not initialize data ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module dialplan (/usr/local/lib64/kamailio/modules/dialplan.so) Please say me how i can load to memory 2500000 rows dialplan?

2 1

JSONRPC commands
by Loïc Boissy 23 Sep '15

23 Sep '15

Hello I am trying to use JSONRPC-S Kamailio module to request for example the content of the location table. More generally, i would like to send RPC commands to my kamailio server using http request. For the moment, kamailio server always returns "400 bad request: corrupted packet". To send my request, i use curl. For example: curl --header 'Content-Type: application/json' --data-binary "{"id": 1, "jsonrpc": "2.0", "method": "ul.dump" }" http://XXX.XXX.XXX.XXX/RPC/ Is there specific syntax to respect? Kamailio server version is 4.3.2. Thank you in advance. Loic BOISSY -- Visit Digigram at IBC2015 at Stand 8.C51, Sept. 11-15 in Amsterdam Register here <http://www.ibc.org/page.cfm/link=478> for your *FREE* pass with Digigram's Exhibitor Customer Code *4462*

2 7

auth_radius module
by Marian Pandoranum 23 Sep '15

23 Sep '15

2 1

loose_route function select wrong next hop!
by Yasin CANER 22 Sep '15

22 Sep '15

3 4

Two suggestions
by 天 22 Sep '15

22 Sep '15

May this mailing list have a search function? This function will make us convient to find what we concern about. OpenIMSCore mailing list have this function, and I felt it Improved productivity. And the mailing list archive system seems to be operated manually. Can the mails be archived automatically? So it can be more instant to developers from asia. BR WangXinXin

2 2

What SIP stack does Kamailio or OpenIMSCore implements?
by 天 22 Sep '15

22 Sep '15

Hi, I am a rookie in SIP related area. I run into OpenIMSCore in recent days, and then I was introduced to Kamailio. Now I am wondering what SIP stack does Kamailio or OpenIMSCore implements? Or neither of them use a SIP stack? Can anybody help me on this? Thanks !

2 1

Re: [SR-Users] Question about in call both side redirection
by Андрей Ярин 22 Sep '15

22 Sep '15

sr-users-request(a)lists.sip-router.org пишет: > Hello On 04/09/15 07:57, ?????? ???? wrote: >> >Hello (sorry for my bad english) - i try to create voice record >> >service by request. User A call to user B. In call by pressing >> >combination like *55 Kamailio must redirect both sides to asterisk, >> >whitch create dynamic conference room with recording. As i understand >> >i need to use dlg_refer() from dialog module, but in log file i get: >> >Konsole output >> >Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: >> >dialog [dlg_req_within.c:85]: build_dlg_t(): no contact available >> >Sep 4 10:45:14 voipc-node2 /usr/local/sbin/kamailio[18941]: ERROR: >> >dialog [dlg_transfer.c:188]: dlg_refer_callee(): failed to create dlg_t >> > >> > >> >In script i try to refer with: >> >dlg_refer("callee","sip:100@10.10.9.209"); >> >dlg_refer("caller","sip:100@10.10.9.209"); >> > > in what context do you use the above actions? In other words, do you > execute them when you process a specific request? If yes, which one? > > Another question, how do you capture when *55 is pressed? Is dtmf sent > via sip info request? > > 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, Sep 28-30, 2015, in > Berlin - http://asipto.com/u/kat For now i try to use event_route[dialog:start] - i testing - can kamailio redirect both sides to external service, and will it work with event_route[dispatcher:dst-down]. If it will work, i will add SIP INFO processing for service codes -- Ярин Андрей Инженер ООО "И.С.-Телеком" 8(351)7786878 доб 113 8(351)2555724 доб 113 8(351)2555787 доб 113

2 2

carrierroute module cannot use prefix that start with +
by José Seabra 21 Sep '15

21 Sep '15

Hello there, I'm using carrierroute module to route calls based on prefix, everything was working until i tried use prefixes that start with +, kamailio core doesn't allow me use the + on carrierroute prefix, i get the following message from kamailio logs: INFO: carrierroute [cr_carrier.c:100]: add_domain_data(): adding domain 1 'main' to carrier 1 'carriers'35(17196) DEBUG: carrierroute [cr_carrier.c:101]: add_domain_data(): domain position 0 (domain_num=1, first_empty_domain=0)35(17196) INFO: carrierroute [cr_data.c:414]: get_domain_data_or_add(): added domain 1 'main' to carrier 1 'carriers'35(17196) INFO: carrierroute [cr_data.c:470]: add_route(): found carrier and domain, now adding route 35(17196) ERROR: <core> [dtrie.c:130]: dtrie_insert(): cannot insert non-numerical character 35(17196) ERROR: carrierroute [cr_domain.c:168]: add_route_to_tree(): cannot insert route flags into d-trie anyone can help in order allow use + in carrierroute prefix. Best regards José Seabra

3 4

dispatcher.list reversed by dispatch.c reindex_dests()
by Giovanni Maruzzelli 21 Sep '15

21 Sep '15

Hello, dispatcher gives me destination sets with entries in reverse order: I have installed from Kamailio deb repository packages version 4.3.2+jessie. # kamailio -V version: kamailio 4.3.2 (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, F_MALLOC, DBG_F_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 I configured dispatcher: modparam("dispatcher", "use_default", 1) modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 3) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_ping_interval", 5) modparam("dispatcher", "ds_probing_mode", 1) I have this dispatcher.list: 1 sip:192.168.1.144:5060 2 1 sip:192.168.1.145:5060 2 1 sip:192.168.1.146:5060 2 2 sip:192.168.1.146:5060 2 2 sip:192.168.1.145:5060 2 So, I expect dispatcher to have: set 1 first two destinations in distribution via hash on callid, and third destination as failover (192.168.1.146 <http://192.168.1.146:5060>) . set 2 all traffic on first destination, and if first fails, all traffic will failover to second destination (192.168.1.145 <http://192.168.1.146:5060>). Instead reindex_dests shuffle the sets upside down: DEBUG: dispatcher [dispatch.c:375]: add_dest2list(): dest [1/1] <sip: 192.168.1.144:5060> DEBUG: dispatcher [dispatch.c:375]: add_dest2list(): dest [1/2] <sip: 192.168.1.145:5060> DEBUG: dispatcher [dispatch.c:375]: add_dest2list(): dest [1/3] <sip: 192.168.1.146:5060> DEBUG: dispatcher [dispatch.c:375]: add_dest2list(): dest [2/1] <sip: 192.168.1.146:5060> DEBUG: dispatcher [dispatch.c:375]: add_dest2list(): dest [2/2] <sip: 192.168.1.145:5060> DEBUG: dispatcher [dispatch.c:494]: reindex_dests(): found [2] dest sets DEBUG: dispatcher [dispatch.c:139]: ds_print_sets(): dst>> 2 sip: 192.168.1.145:5060 2 0 (,0,0) DEBUG: dispatcher [dispatch.c:139]: ds_print_sets(): dst>> 2 sip: 192.168.1.146:5060 2 0 (,0,0) DEBUG: dispatcher [dispatch.c:139]: ds_print_sets(): dst>> 1 sip: 192.168.1.146:5060 2 0 (,0,0) DEBUG: dispatcher [dispatch.c:139]: ds_print_sets(): dst>> 1 sip: 192.168.1.145:5060 2 0 (,0,0) DEBUG: dispatcher [dispatch.c:139]: ds_print_sets(): dst>> 1 sip: 192.168.1.144:5060 2 0 (,0,0) DEBUG: dispatcher [dispatcher.c:347]: mod_init(): loaded dispatching list So, Kamailio give me set 1 and set 2 as per post-reindex_dests, and failover is 192.168.1.144 for set 1, and 192.168.1.146 for set 2. Is this the expected behavior? Am I doing something wrong in configuration? Thank you all in advance, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

2 3

Compiling Kamailio for Cubie/RaspberryPi/others ARM arch from main repos
by nadie 21 Sep '15

21 Sep '15

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 #!/bin/bash # Compiling Kamailio for Cubie/RaspberryPi/others ARM arch from main repos: echo 'deb-src http://deb.kamailio.org/kamailio43 jessie main' > /etc/apt/sources.list.d/kamailio.list wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - dpkg --add-architecture i386 apt-get update apt-get install bison debhelper erlang-dev gcj-jdk libconfuse-dev libdb-dev libev-dev libfreeradius-client-dev libgeoip-dev libhiredis-dev libjansson-dev libjson-c-dev liblua5.1-0-dev libmemcached-dev libmono-2.0-dev libmysqlclient-dev libperl-dev libpq-dev libpurple-dev librabbitmq-dev libreadline-dev libsasl2-dev libsctp-dev libsnmp-dev libsqlite3-dev libunistring-dev libval-dev libxml2-dev python-dev unixodbc-dev uuid-dev adduser python libc6 libncurses5 libpcre3 libreadline6 libtinfo5 snmpd mkdir kamailio && cd kamailio/ apt-get -b source kamailio dpkg --remove-architecture i386 rm /etc/apt/sources.list.d/kamailio.list apt-get update dpkg -i *.deb cd .. && rm -rf kamailio/ # end :-) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.12 (GNU/Linux) iQIcBAEBAgAGBQJV+BKUAAoJEG5DH1ilMeeenM8P/jqzMKtxmRheDDT4pqxzo9W1 lSNpg+UhYr5CC/IXCgGXkP5Sup87/qiAVYz/u902pS9hdIBfdbup2PUsvTX4QK+1 Dybxli9ym4Fup9m6MuB+VLIjkiOOwtGnvChbnPgjJnMxDjzBbDFEmositlfAeiFU XcyIfjYDndvd2+0a1UkTVVuGBEbY6BdVkANvINy2BkZUkolstTgVr7jirYUSTEsa U9M+lX98Dbau5gfvyx87a6CkcW8fyKCgZGBYqjBIyr1TrPgtGcXdcgC77iD2pziW wkU9DIRX0sOaIiAb76hX64SL8U9b+dJr2t17PXjiEEcVWuc7YtV2R0YURAfquw9O ycSh+PWFJgc1PZx5dFZlWceDXjdffnPiWt3Fys3HuVNWLY6iVfOSFv7sMcJ4NyaT N+u81Dr4T9blZ2yRmibRUzlmSlcq4at2yKE19/AsejphXI7mnaARuNlHZUq+4RY9 EF4zlr5zTHmPhvLoh7rjzqWGLmun4g7a7mCnbMDUKKbndqb24QKhO9czvkTvYLGR vbAK56kM96jh3OBLenjl58YzK4qBbJ5sM+UCw6DA5HnF+/jh+oK4zeKL28yqEaG0 SxS/xuG4xxi2wPvGJ36Rg6gIB5cHAxizGLMEPGUPTKCPhKLFIbtunabNu2gj+6nh bpe/fyLw0ZKHF8R4tifU =Ty41 -----END PGP SIGNATURE-----

2 2

Kamailio ACK retransmission
by Yasin CANER 21 Sep '15

21 Sep '15

1 0

Kamailio - Asterisk ACK retransmission problem
by Yasin CANER 21 Sep '15

21 Sep '15

1 0

Kamailio presence shared DB_text
by Ewgeny 21 Sep '15

21 Sep '15

Hi list! We use Kamailio 4.2 for presence BLF with DB_TEXT. And i want to make cluster with a few Kamailio's presences, to load balance. Can you please tell me - is that possible to use same shared DB_TEXT database with 2 or more Kamailio servers ? modparam("presence", "subs_db_mode", 3) modparam ("presence","dburl","/tmp/share/dbtext") 1-st server will handle SUBSCRIBE's and this Subscribe's should be accessible also from 2-nd server. Then PUBLISH'es will be sent to both servers in round robin algorithm. Will Kamailio's work with one shared DB or - i'll got some errors ? Maybe i'm wrong somewhere ? Thanks in advance. Kind regards Ewgeny

2 3

Proxy-Require: privacy error
by Chad 21 Sep '15

21 Sep '15

When a call comes in with a SIP header: "Proxy-Require: privacy" line kamailio throws this error: Aug 21 14:27:57 load1 /usr/sbin/kamailio[29447]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f15150 Aug 21 14:27:57 load1 /usr/sbin/kamailio[29446]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f0a7b0 Aug 21 14:27:58 load1 /usr/sbin/kamailio[29447]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f13180 Aug 21 14:27:58 load1 /usr/sbin/kamailio[29446]: ERROR: <core> [parser/hf.c:224]: dump_hdr_field(): DEBUG: dump_hdr_field: type=18, name=Proxy-Require, body=privacy, parsed=(nil), next=0x7efc42f0dfb8 Anyone know why that happens or how to fix it? -- ^C Chad

2 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users