sr-users February 2024

sr-users@lists.kamailio.org

49 participants
39 discussions

[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE

by Denys Pozniak

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 5.1.2 (x86_64/linux) -- BR, Denys Pozniak

1 month

Dialog and DMQ on restart

by Björn Klasen

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

1 month

loose_route false when using topos? At which point in the call flow, does topos restore the Route header?

by Benoit Panizzon

Hi https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html I have the exact same issue. When the B side is starting a new transaction (UPDATE to refresh the session in my case) without topos enabled, that transaction contains one or multiple route header and a to_tag. Therefore loose_route() is true and the call is more or less sent to route(RELAY) immediately without further checks. If topos is enabled, all record-route header are removed and only one Via sent to the CPE. So when the CPE replies, there is no Route header. Therefore loose_route returns false. This makes me wonder, at which stage is topos restoring the route headers? Shouldn't that happen before loose_route is evaluated? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 month

SSL key logger for Diffie-Hellman cipher

by Calvin E.

I've been tasked to use LD_PRELOAD to log SSL keys for TLS connections using a Diffie-Hellman cipher. The first attempt did not work, so I wanted to sanity check whether Kamailio's TLS support is built in such a way that would defeat LD_PRELOAD. The instructions from the vendor are to update /etc/init.d/kamailio like this: env SSLKEYLOG_UDP='127.0.0.1:1234' LD_PRELOAD="/usr/local/src/voipmonitor-git/tools/ssl_keylogger/sslkeylog.so /usr/lib/x86_64-linux-gnu/libssl.so.3" \ start-stop-daemon --start --quiet --pidfile $PIDFILE \ --exec $DAEMON -- $OPTIONS || log_failure_msg " already running" Is there anything special in Kamailio (5.7.3 on Debian 12) that would prevent this from working? Not necessarily something to defeat a keylogger, but maybe the way tls.so gets loaded? The only discrepancy I've noticed is the vendor docs refer to libssl.so.3 not libssl.so.1, but the vendor said that should be OK. I'd love to hear from someone already using VoIPmonitor with Diffie-Hellman ciphers and Kamailio.

1 month, 2 weeks

Roadmap to next major Kamailio release series v5.8.x

by Daniel-Constantin Mierla

Hello, discussed a bit during the online Kamailio devel meeting, it is time to set the milestones towards the next major Kamailio release series v5.8.x. If no other suggestions that suit more developers, I would propose to freeze by end of this month or early February, then test for about 4 weeks as usual and release by end of February or during March. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 month, 2 weeks

Answer OPTIONS from an Asterisk

by Ludovic ELUECQUE

Hello, I've been using Kamailio to replace a Telcobridge ProSBC for several days now, but I've been stuck for a whole day on a simple concept My Kamaillio is connected to an Asterisk to which it redirects requests. However, this Asterisk sends OPTIONS requests in order to ping the phone. On the Kamailio side, I don't know how to respond to these requests. I've done tons of research but I can't find anything similar, maybe there's something very obvious I'm missing.... If I t_relay the message it never receives a reply via the phone's public ip/port. Thanks in advance if anyone could help me with this. Please apologize if it's a dumb thing. SIP OPTIONS: 2024/02/28 23:28:25.181663 100.66.138.5:5060 -> 100.66.138.7:5060 OPTIONS sip:110@100.66.138.7:5060 SIP/2.0 Via: SIP/2.0/UDP 100.66.138.5:5060 ;rport;branch=z9hG4bKPje213d53f-fd1d-47f2-b0f8-fd3a8bf5188c From: <sip:110@100.66.138.5>;tag=46e51407-77b0-47f1-af73-cd8bec95b179 To: <sip:110@100.66.138.7> Contact: <sip:110@100.66.138.5:5060> Call-ID: 74401738-d874-4c5c-a4b5-b74514361043 CSeq: 50224 OPTIONS Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 ============= kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_RTPENGINE # define WITH_ANTIFLOOD #!define WITH_WEBSOCKETS #!define WITH_TLS #!define WITH_NAT ... #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### DEFINE VARIABLE ######## #!substdef "!MY_IP_ADDR!X.X.X.X!g" #!substdef "!MY_WS_PORT!8080!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" #!substdef "!MY_MSRP_PORT!4444!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 # FLB_ - per branch flags #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:X.X.X.X:5060 listen=udp:100.66.138.7:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### 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 #!ifdef WITH_JSONRPC loadmodule "xhttp.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 "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "rtpengine.so" #loadmodule "rtpproxy.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" # loadmodule "nathelper.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE #!else #!loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_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", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) #rtp modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "force_send_interface", "100.66.138.7") #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- 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) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 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 ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!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", "load_backends", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- #modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- #!modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sip.super.fr") #modparam("nathelper", "sipping_method", "OPTIONS") # 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", 40) 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_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ######## websocket ############ #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR tcp_accept_no_cl=yes # tcp_rd_buf_size=65536 #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif ######## SECURITY ####### server_signature=no sip_warning=0 user_agent_header="User-Agent: SIP Server" ####### 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 { route(REQINIT); route(NATDETECT); route(REGISTER); route(INVITE); route(ACK); route(BYE); return; } # Wrapper for relaying requests #Check initial detail route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent|pplsip") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200", "Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # 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; } route[REGISTER] { if (is_method("REGISTER")) { t_on_reply("1"); force_send_socket(100.66.138.7:5060); t_relay_to_udp("100.66.138.5", "5060"); exit(); } } route[INVITE] { if (( is_method("REFER")) || is_method("INVITE") ||is_method("NOTIFY")) { record_route(); if (src_ip == "100.66.138.5") { if(lookup("location") && !is_method("OPTIONS")) { xlog("L_NOTICE", "user enregistrax"); force_send_socket(X.X.X.X:5060); rtpengine_manage("replace-origin replace-session-connection media-address=X.X.X.X direction=priv direction=pub"); t_relay(); } else { sl_send_reply("403","Not registred"); } } else { force_send_socket(100.66.138.7:5060); rtpengine_manage("replace-origin replace-session-connection media-address=100.66.138.7 direction=pub direction=priv"); t_relay_to_udp("100.66.138.5", "5060"); } } } route[ACK] { if (is_method("ACK")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } route[BYE] { if (is_method("BYE|CANCEL")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } onreply_route { if (status=~ "180|183") { rtpengine_manage("replace-origin replace-session-connection"); } if (status=~ "200") { if(has_body("application/sdp")) { rtpengine_manage("replace-origin replace-session-connection"); } } } onreply_route[1] { if (status=~ "2[0-9][0-9]"){ save("location"); xlog("L_NOTICE", "enregistrement"); } } ====================

1 month, 4 weeks

best approach to setup SIP Session sharing between servers

by Sergio Charrua

Hi all! I am having some difficulties/doubts on what would be the best approach to let multiple Kamailio servers share their SIP sessions between each other. The goal is to have HA on Kamailio cluster such that if a call is received and initiated on Kamailio #1, and during any moment of the call (before or while established) if the server #1 goes down or Kamailio stops working for any reason, the call can be processed by Kamailio server #n available in the cluster. I do not want to mess with Virtual IPs or DNS, that is not the point, but instead need to get a proper way to share SIP sessions between Kamailio servers such that any server could continue the call without issue. I have checked the Dialog and DMQ module, but I am not sure if that is the way to go.... Could any one share a solution for this? Greatly appreciated. Cheers, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

1 month, 4 weeks

Clarification on pv_www_authenticate

by James Lipski

Hello, I would like get further information on the "pv_www_authenticate" function from auth module. For the response code "-1", which is listed as "generic error", what exactly is covered when this function returns this? Is it just everything else that isn't listed reason (invalid password, nonce expired, no credentials, etc...). On my production system, I'm seeing this as a response code on some registration requests, and just want to know what is causing this (enabling full verbose logging is a bit difficult on my production system, and want to try to replicate it on my test environment, however just need clarification on what this covers). Thank you.

1 month, 4 weeks

Topos methods_nocontact parameter

by Benoit Panizzon

Hi I'm having another go at topos trying to solve an issue of sip messages getting to big for certain clients to process. Unfortunately some methods (ACK/PRACK/BYE probably CANCEL) do not contain a contact header, therefore the topology is still revealed (and potentially a lot of via header inserted increasing message size). Is it safe to set methods_nocontact to an empty string to force creation of a Contact header for every SIP method? Is there another way to completely hide topology aka keeping header small? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 month, 4 weeks

How to keep SIP messages small, possibly below fragmentation limit?

by Benoît Panizzon

Hi In the process of getting rid of our commercial B2Bua SBC and handling all clients directly via Kamailio... We have come across some (unfortunately many we have in use) CPE which have an issue with fragmented packets or which simply do not allocate enough memory to parse or create messages of a certain size. Their SIP stack just crashes with errors like 'message to large for send buffer' and similar. This affects messages with containing as little as 5 via and 4 record-route header and a couple of codec in the SDP. Is there a recipe for trying to keep packets as small as possible? I would like to avoid using the topos module as I issues with this module completely failing in certain situations. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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sr-users February 2024