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38 participants
37512 discussions

DMQ module failure
by Sergio Charrua 01 Mar '24

01 Mar '24

Hi all! I'm testing the DMQ module, but having difficulties setting it up. [...] My kamailio.cfg is as follows : [...] listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 [...] modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS modparam("dmq", "notification_address", "sip:10.20.0.4:5062 ")#DMQ_NOTIFICATION_ADDRESS modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) modparam("dmq_usrloc", "enable", 1) If I check syntax (using -c parameter) all is ok. Once I start Kamailio, the following error appears on system log: INFO: dmq [dmq.c:164]: make_socket_str_from_uri(): unknown transport protocol - fall back to udp ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is listening on ERROR: <core> [core/sr_module.c:1030]: init_mod(): Error while initializing module dmq (/usr/local/lib64/kamailio/modules/dmq.so) I have used UDP as well as SIP as protocols for Server Address and Notification Address, used port 5060 and 5062 (and others too), even added a new listen=udp:0.0.0.0:5062, as well as listen=tcp:0.0.0.0:5062 but the result is always the same. Using Kamailio 5.7.4 on RHEL 9 with SELINUX set to enforcing (if relevant). No firewall active. local IP address is 10.20.0.1 second kamailio node IP address is 10.20.0.4 Any ideas? At first sight, it seems to be all set as per documentation. I even found other configurations online, and apart from the IP address, all is similar. Thanks in advance, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

2 2

Answer OPTIONS from an Asterisk
by Ludovic ELUECQUE 28 Feb '24

28 Feb '24

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"); } } ====================

2 1

best approach to setup SIP Session sharing between servers
by Sergio Charrua 28 Feb '24

28 Feb '24

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/>*

8 11

Clarification on pv_www_authenticate
by James Lipski 28 Feb '24

28 Feb '24

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.

2 1

Topos methods_nocontact parameter
by Benoit Panizzon 27 Feb '24

27 Feb '24

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 ______________________________________________________

3 7

How to keep SIP messages small, possibly below fragmentation limit?
by Benoît Panizzon 26 Feb '24

26 Feb '24

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 ______________________________________________________

2 2

Development is open for future release of v5.9.x series
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 was created, therefore the master branch is open for adding new features, to be part of future release series v5.9.x (or whatever version is decided for next series). Any bug fix committed to master that applies to 5.8.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.8 or 5.7. Expect that v5.8.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.9.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2024 or the beginning of 2025. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Branch 5.8 has been created
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 has been created, to be used for releasing v5.8.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.8 cd kamailio-5.8 git checkout -b 5.8 origin/5.8 Pushing commits in this branch: git push origin 5.8:5.8 Note that 5.8 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.8 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.8 will be out, respectively Kamailio v5.8.0. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Building Debian packages for further modules
by Christian Berger 22 Feb '24

22 Feb '24

Hi, for our workflow we build our own Debian packages for kamailio. We are currently trying and failing to create a package for the kafka modules. What we have tried so far: Adding the "Build-Depends:" entries for librdkafka-dev in debian/control Adding a "Package" in debian/control Adding kafka to the PACKAGE_GROUPS in debian/rules So far it seems to build the module and a package for it, but the package only contains metadata, but no .so file. We are currently running out of ideas. What else can we do? Best wishes Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 2

Kemi questions
by Stefan-Cristian Mititelu 21 Feb '24

21 Feb '24

Hello, When writing kemi cfg (python in my case), is there any way to get rid of standard cfg shared variables and use kemi cfg variables instead? E.g. Instead of $shv(blabla) (in standard config) can I use some global python var called 'blabla'? Is it seen and shared by all kamailio processes? Or I still need to use KSR.pv.get/set("$shv(blabla)") ? Thank you, Stefan

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