- sr-users - kamailio.org

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I am using kamailio 5.3. Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could work. David Villasmil , i know i can use SQLOPS but i was thinking of easier method of doing this. By successful REGISTER i mean save only REGISTER attempt that authenticated correctly at SIP server and in 200 OK a binding was returned. Same if the binding is later removed i would like remove it from database. I need this because when i run REGISTER over kamailio to destination SIP server i see all … [View More]register from same IP. I need to be able to check what was the real source IP of client REGISTER. Regards, Tomasz pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I am using kamailio 5.3. > > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could > work. > > David Villasmil , i know i can use SQLOPS but i was thinking of easier > method of doing this. > By successful REGISTER i mean save only REGISTER attempt that > authenticated correctly at SIP server and in 200 OK a binding was returned. > Same if the binding is later removed i would like remove it from database. > I need this because when i run REGISTER over kamailio to destination SIP > server i see all register from same IP. > I need to be able to check what was the real source IP of client REGISTER. > > Regards, > Tomasz > > > czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> napisał(a): > >> Try to use save() when you process the 200 ok reply for REGISTER in >> the reply route. >> >> Regards, >> Ovidiu Sas >> >> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >> wrote: >> > >> > I setup kamailio to forward REGISTER packets using dispatcher to two >> servers main and backup. >> > >> > The routing logic is quite simple i use the dispatcher probing >> mechanism to detect if my main dispatcher set id (one server) is active or >> inactive once it detect a failure all packets are sent to secondary SIP >> server to achieve High Availability. >> > >> > I am looking for module or a way to save into kamailio database client >> information such as IP address, register username (from FROM header), >> digest username (authorization username), call-id for successful >> registration. >> > >> > I tried to use registrar module but once i used function >> save("location") the client was registered on kamailio and packets was not >> forwarded to main/secondary SIP server (like dispatcher stopped working, >> the destination did not override). >> > >> > REGISTER is sent to SIP URI of kamailio. >> > >> > Please let me know if you have any idea how to make this data update in >> database (insert an entry on fresh successfull REGISTER, delete on expired >> or removed binding). >> > >> > Best regards, >> > Tomasz >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > [View Less]

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Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 15 Nov '19

15 Nov '19

I setup kamailio to forward REGISTER packets using dispatcher to two servers main and backup. The routing logic is quite simple i use the dispatcher probing mechanism to detect if my main dispatcher set id (one server) is active or inactive once it detect a failure all packets are sent to secondary SIP server to achieve High Availability. I am looking for module or a way to save into kamailio database client information such as IP address, register username (from FROM header), digest username … [View More]

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dispatcher, pvar_hash and multiple kamailio instances
by Barry Flanagan 15 Nov '19

15 Nov '19

Hi, I have multiple Kamailio instances using dispatcher using alg “7” - hash over the content of PVs string, both using the same dispatcher list. Is it reasonable to expect both Kamailio, given the same hash_pvar, to select the same host from the same set? Looking at the source I do not see any reason why not, but I am getting different results from the two Kamailio servers. The only difference between them is different minor version numbers - 4.3.4 and 4.3.7 I want to ensure that calls with … [View More]

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Re: [SR-Users] Question on Database Version Table
by JR Richardson 15 Nov '19

15 Nov '19

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> … [View More]

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Processes blocking - troubleshooting slow actions
by Julien Chavanton 15 Nov '19

15 Nov '19

I am troubleshooting intermittent latency on a server, several processes will report slow action at the same time every now and then, something odd it that some of the actions are most likely not using any mutexes, I can not imagine how they can end up blocking unless the process is actually suspended by the OS, however the CPU usage and load average is quite low on this server. Is there any reason why assigning to a PKG var could be delayed 3 seconds. 2019-11-13T21:58:04.018631614Z 23(42) … [View More]

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kamailio 5.2.4 unable to load db_mysql.so
by Rupesh Kumar 15 Nov '19

15 Nov '19

I am trying to compile and load kamailio 5.2.4 on centos 7.5. I am getting below errors whenever I start kamailio . I have the latest openssl and openssl-devel packages installed on my centos . Not sure what other dependencies . Not starting Kamailio SIP Server: invalid configuration fil[FAILED] &… [View More]

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Next online devel meeting - irc/video conference - Thursday, Nov 14, 15:00UTC
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, on Thursday and Friday this week over a dozen of Kamailio developers will be in Dusseldorf for the annual coding meeting, so we thought of doing an online meeting via some video conferencing bridge, with IRC as a backup chatting channel, to allow also other community members to join remotely and discuss the development plans for short/medium term. It is more or less what we used to do during the past IRC devel meetings, but to make the discussions go faster and shorter the meeting, so … [View More]

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Kamailio v5.3.1 Released
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, Kamailio SIP Server v5.3.1 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.1. For more details about version 5.3.1 (including links and guidelines to download the tarball or … [View More]

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Releasing Kamailio v5.3.1
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, the plan right now is to release v5.3.1 during the next Kamailio Developers Meeting, if we succeed to coordinate on site, therefore expect it to happen next week during Nov 14-15. Besides just doing this specific release, to goal is to try to build a more automatic process for releases, document how is done, etc... updates about the results will follow after the event. Anyhow, for now the usual the pre-release notice: if anyone is aware of some issue that is not yet reported, do it … [View More]

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event_route[core:msg-received] native vs kemi/lua
by GG GG 14 Nov '19

14 Nov '19

Hi! I just made a simple test performance between native script and kemi/lua with the event_route[core:msg-received] callback. # lua script children=1 cpu.process=100% function ksr_core_received() exit; end # native script children=1 cpu.process=70% event_route[core:msg-received] { exit; } # tested with udp 5060, 1000 Mbits/sec iperf -c 127.0.0.1 -p 5060 -u -b1000m I don't know if it's normal and what we should expect from kemi, but I would like to share this with you. Maybe someone … [View More]

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kemi framework design
by GG GG 12 Nov '19

12 Nov '19

Hi, looking at the documentation and the code, I couldn't answer myself. - does kamailio use one lua interpreter per fork ? - about the safety, I just made a bad lua module in c, to force a segmentation fault in the module. I would expect kamailio to protect itself from the crash, but kamailio restarted.

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Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

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Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

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Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

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Kamailio in cloud environments
by Alejandro Recarey 11 Nov '19

11 Nov '19

Hi, I am trying to get kamailio working cloud provider, autoscaling them behind a TCP load balancer. Is there a way for kamailio to discover its Public IP at startup? Or do people generally program their own startup scripts that modify the kamailio config file before starting it? Is there any gotchas you have hit while configuring this setup? Thank you! Alex

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uac_replace_from(uri) alters display name
by Sergiu Pojoga 10 Nov '19

10 Nov '19

Hi ppl, Need to replace origination carrier's domain with my own, so I do: branch_route[MANAGE_PSTN_IN_BRANCH] { uac_replace_from("sip:$fU@mydomain.net"); } The dialog-forming INVITE is relayed properly with ONLY the domain part being replaced. However, subsequent messages such as ACK and BYE have the From DISPLAY NAME (_uac_fdp) stripped off. Not that it affects real functionality in any known way (CLID is still good for callee), but curious if this is intentional or not, as *… [View More]

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OPTIONS request not authenticating
by Gustavo A. M. 10 Nov '19

10 Nov '19

Greetings, I'm trying to configure kamailio to get OPTIONS response, but it does not working properly. After a successful REGISTER, I send OPTIONS first time, get Proxy- Authenticate header to calculate Proxy-Authorization, and resend OPTIONS with Proxy-Authenticate header, but server follows answering with a 407 response, apparently ignoring Proxy-Authenticate. Note that same password is used in REGISTER request, and first time REGISTER response with a 401, not a 407 response. I think … [View More]messages sent from client are right, and problem is on the server configuration. Follow messages and kamailio.cfg. Thanks in advance, Gustavo Adolfo Moennich OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 0 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 0 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.467d Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905;received=192.168.122.12 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1wV3HNJXdfUDchNlEUkT6kHCtqBAn" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 1 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 Proxy-Authorization: Digest uri=sip:6000@kamailio.memorea.com,nonce="Xc c1wV3HNJXdfUDchNlEUkT6kHCtqBAn",realm="kamailio.memorea.com",response=" 5bcd184eea53284f3ec548c2412ce36a",username=6000 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 1 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.32de Via: SIP/2.0/UDP 192.168.122.21:5060;received=192.168.122.12;rport=58905 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1xl3HNJqRobJFuZggPSGvNdtp+GnP" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 kamailio.cfg content's: #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.or g> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:temp@database.memorea.com/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=8 /* 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:10.0.0.10:5060 #!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 */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=$si) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (uri==$ru) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == $ru) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (is_method("OPTIONS") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("OPTIONS|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!=$fu && uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==$ru) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=$fu) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } [View Less]

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dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | … [View More]

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Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [… [View More]

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_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going … [View More]to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $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; } route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif } [View Less]

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How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738"… [View More]

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES … [View More]

2 2

Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by … [View More]

3 2

MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla … [View More]

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sr-users