- sr-users - kamailio.org

New MariaDB server "limiting" Kamailio CPS
by Stephen Lokker 01 Jun '21

01 Jun '21

Good afternoon All, Im hoping for some advise or direction with the below. We are running a Kamailio SBC used for high call volumes and recently decided to upgrade the Database servers as the existing servers were struggling. Both the old and new databases are using MariaDB(version details all below), the problem we face is that since migrating to the new database there appears to a limitation to the amount of DB connections that can be processed or accepted. The SBC functions normally until the CPS starts to exceed around 100cps at which point the SBC stops responding to invites(in bursts which resembles flood prevention). We have checked both the Kamailio logs and the MySQL logs on the DB, both log files have no indication of errors that would cause the problem experienced(the logs attached are seen when the SBC is cut back to use the old DB, not sure they are related to the fault), the connections/processes in mysql do not drastically increase or get stuck either. The fault points to a db connection limit of some kind but all users have no limitations set. If anyone has experienced a similar problem or has any advise it would be greatly appreciated. Some info on the Kamialio and DB versions New DB root@voice-jhb-db01:~# mysql --version mysql Ver 15.1 Distrib 10.3.27-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2 Old DB [root@isadb01 ~]# mysql --version mysql Ver 15.1 Distrib 10.0.36-MariaDB, for Linux (x86_64) using readline 5.1 Kamailio SBC [root@sbc01 ~]# kamailio -v version: kamailio 5.1.4 (x86_64/linux) Kind regards, Stephen Voice Engineer Tel: 2404 Fax: 3468 Email:

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How To setup Provider SIP trunk with registration
by zak ast 31 May '21

31 May '21

Hi, I know I can you uacreg to register a SIP trunk as a SIP client with username/password. But how can I set up a SIP trunk as a provider and get the other end (SIP client) to register with username/password. Regards, Zak.

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Architecture - your opinion is welcome and your experience too
by Daian Conrad 28 May '21

28 May '21

Hi all, I am studying and improving my understanding of how kamailio works, I have used asterisk for a few years(and starting learning freeswitch too), and I use an application to manage extensions, trunks and other media services. I'm building a new scenario where kamailio is facing the internet and the asterisk(s) are internally on private networks, would like the opinion of you who are more experienced with more complex environments than me. I cannot change the application that uses asterisk for now, due to the effort to develop event monitoring, cdr and other features that I use today. I imagined and working on making kamailio functional by saving the registration of extensions with location on DB, and with the UAC module making the registration of extensions on asterisk(s), replacing the registration address with the address of kamailio, its functional at this point, but data replication with extensions turns things some hard to mantain manually. I think about using the DNS domain for each asterisk and make this forwarded, each asterisk response for a fqdn and its extensions, like ( pbx1.example.com is forwarded to asterisk1, pbx2.example.com to asterisk2) and so on. Read about the dispatcher, rr, htable,carrierroute module to identify the domain and forward based on that , none made me sure to be chosen for the role, however, everyone has the resources to do it. I am very wrong to follow this path, which option in your opinion is "less" painful for a beginner apprentice like me? I think about simple proxy based on domain requests(all messages and dialogs) to asterisk where is responsible for the domain(realm) but, loss the ability to use great security features of kamailio, is what I understand now with the knowledge I have. Sorry and forgive my english, i'm not very good at writing. -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

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No Audio For Outbound Calls
by Kashish Raheja 28 May '21

28 May '21

Hi All, I have set up Kamailio in the following manner: Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP) I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides. However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here. Note: I am running the RTP proxy on Kamailio server. Any help on why this might be happening? Thanks. Regards Kashish +919413745250

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 28 May '21

28 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

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Setting the prefix in Json formatted logs
by Abdirahman Osman 27 May '21

27 May '21

Hello everyone, Tried to use the log-engine to push json formatted logs, here is a what kamailio is sending, my question is, How can I set the kamailio prefix? (232) { "kamailio.level": "DEBUG", "kamailio.module": "tm", "kamailio.file": "timer.c", "kamailio.line": 634, "kamailio.function": "wait_handler", "kamailio.callid": "", "kamailio.logprefix": "", "message": "finished transaction: 0x7f232236b1c0 (p:0x7f232216ae58/n:0x7f232216ae58)" } Thank you! Abdirahman

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 27 May '21

27 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

1 0

Body formats for filter_body()
by Duarte Rocha 27 May '21

27 May '21

Hello all, A couple years ago the function filter_body() from textops would only work with "application/sdp" if the string on the body was "Content-Type: application/sdp\r\n". It would fail if there wasn't a space after ":" and it would fail if there was present "\r\n\r\n" instead of "\r\n". Is this the actual behaviour of the function or has it been changed to also identify SDP in the formats mentioned above? Best Regards,

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Collecting stats from Kamailio.cfg using Kamailio_exporter Prometheus
by Hisham Ahmad Jan 26 May '21

26 May '21

Hi, I'm using Kamailio version 5.3.7 with DISPATCHER module and I would like to integrate my kamailio with a kamailio_exporter for monitoring using Prometheus server. Is it possible to collect data such as the number of total calls (inbound, outbound) and plot time-graphs using those stats? If so, what modules/functions would I need apart from the Dispatcher module to be able to have efficient results? Any degree of help will be appreciated. *Hisham Jan* *VoIP Engineer* *0322-6398775*

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not able to display number in SCA module
by Aatif Shaikh 26 May '21

26 May '21

Hello All, I am using sca module to handle SLA presence and I am using sca.update_appearance RPC command to update the display name. kamcmd sca.update_appearance sip:215@voice.example.com 3 held KELVIN after executing the above command I am getting the KELVIN name on my Polycom phone display but I want to also update the display number so is there any way to update the display number for SLA presence using the SCA module

2 1

segfault kamailio 4.4.6 Registrar
by Jonathan Hunter 26 May '21

26 May '21

Hi Guys, We are running kamailio 4.4.6 on debian 8 and not had any issues, however we did suffer a segfault and a core was generated; May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:740]: handle_sigs(): child process 9575 exited by a signal 11 May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: ALERT: <core> [main.c:743]: handle_sigs(): core was generated May 25 09:17:06 Kam-HA-1 /usr/sbin/kamailio[9560]: INFO: <core> [main.c:755]: handle_sigs(): terminating due to SIGCHLD Running gdb on the core file generated returns this output; Reading symbols from /usr/sbin/kamailio...Reading symbols from /usr/lib/debug/.build-id/bc/fbd7e44c057e9e7e0680c2a5e8f6a47290c5ba.debug...done. done. [New LWP 9575] warning: Error reading shared library list entry at 0x28250c3348 warning: Error reading shared library list entry at 0x79c0850018bf73e8 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /etc/kamailio/kamailio.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 353 lvalue.c: No such file or directory. (gdb) bt full #0 0x0000000000589cce in lval_pvar_assign (lv=<optimized out>, rv=<optimized out>, msg=<optimized out>, h=<optimized out>) at lvalue.c:353 r_avp = 0x6 destroy_pval = -874501672 pvar = 0x0 pval = {rs = {s = 0x0, len = 0}, ri = 0, flags = 0} avp_val = {n = -874501704, s = {s = 0x7fb6cbe029b8 "\272/}", len = 5807505}, re = 0x7fb6cbe029b8} v = 6 #1 lval_assign (h=0x7fb6cbe029d8, msg=0x344, lv=0x0, rve=0x0) at lvalue.c:400 rv = 0x7fb6cbe029b8 ret = -874501672 __FUNCTION__ = "lval_assign" #2 0x0000000000000000 in ?? () No symbol table info available. Is this of any help? Could someone advise what next steps might be? Should we be looking to upgrade or is there a way to get more detail out of the core dump? Many Thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows 10

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Kamailio 5.3.8 and RTPengine 9.4.1.1+0~mr9.4.1.1
by Mihai Cezar 23 May '21

23 May '21

Hi All, I've been using Kamailio as a proxy for asterisk (with public ips), everything was fine until i've changed the setup to private ips for asterisk. What I did so far: - Kamailio and rtpengine are on a box with 1 public IP. - Asterisk containers on private ips each asterisk has 2 trunks ( 1trunk voip provider 1trunk the Proxy). What works: - Proxy registrations and sip signaling works. What it does not work: - On the asterisk I got only on rtp connexion, my guess is that has something to do with the SDP, asterisk put the private ip in the SDP header. RTPengine is configured using this guide https://github.com/sipwise/rtpengine Can you have a look at the kamailio config, i think i am missing something or do something wrong. Thank you! command to start was: rtpengine --table=42 --listen-ng=127.0.0.1:2223 --interface=public/110.20.20.1 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback and the Kamailio cfg is this: #!KAMAILIO #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define WITH_DEBUG #!define WITH_NAT ## This is the main configuration file for our proxy! ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 memdbg=5 memlog=5 log_stderror=yes disable_core_dump=no sip_warning=yes #!else debug=-0 log_stderror=no sip_warning=no disable_core_dump=yes #!endif log_facility=LOG_LOCAL0 fork=yes children=8 auto_aliases=no port=5060 disable_tcp=yes server_id = 1 server_signature=yes server_header="Server: proxy" dns_try_ipv6=no dns_try_naptr=no dns_retr_time=1 dns_retr_no=1 rev_dns=no ####### Modules Section ######## #loadmodule "db_mysql.so" 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 "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" ####### Modules Settings # ----------------- setting module-specific parameters --------------- modparam("xlog", "buf_size", 4096) modparam("xlog", "prefix", "xlog: ") modparam("xlog", "log_facility", "LOG_DAEMON") modparam("xlog", "log_colors", "L_ERR=cr") modparam("xlog", "log_colors", "L_ERR=cr;L_WARN=px") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 0) #!ifdef WITH_NAT modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@proxy") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "hash_table_size", 256) modparam("rtpengine", "hash_table_tout", 600) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "append_branches", 0) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 1) # ----- acc params ----- modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/etc/kamailio/plugins/dispatcher.list") modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:ping@proxy") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ########### # main request routing logic request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { rtpengine_delete(); 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); if (isflagset(FLT_FS)) { route(FROM_WORLD); exit; } ### only initial requests (no To tag) # 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 sl_send_reply("100","Trying"); } # 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 route(DISPATCH); } route[FROM_WORLD] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } if (!add_path()) { sl_send_reply("503", "Internal Path Error"); } 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 (has_body("application/sdp")) { rtpengine_manage(); } append_hf("Supported: path\r\n"); exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("L_WARN","Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { rtpengine_delete(); 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; } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be ACK after a 487 or e.g. 404 from upstream server t_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); } if (!add_path_received()) { sl_send_reply("503", "Internal Path Error"); }; route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not found"); exit; } # Dispatch requests route[DISPATCH] { # to add more servers in k8s if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("3")) { if (is_method("REGISTER")) { xlog("L_WARN", "natdetect fix_nated_register ip: $si\n"); fix_nated_register(); } else { xlog("L_WARN", "natdetect set_contact_alias ip: $si\n"); fix_nated_contact(); } force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { xlog("L_WARN", "sdp fix ip: $si\n"); fix_nated_sdp("10"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { xlog("L_WARN", "nat_uac_test 1..."); rtpengine_manage("replace-origin replace-session-connection direction=public ICE=force"); } else { xlog("L_WARN", "nat_uac_test addrtp..."); rtpengine_manage("codec-strip-all codec-transcode-PCMA codec-transcode-PCMU codec-transcode-G729 replace-origin replace-session-connection direction=public ICE=force "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_WARN", "new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_WARN", "incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { rtpengine_manage(); } } onreply_route[REPLY_ROUTE] { if(status=~"2[0-9][0-9]") { rtpengine_answer("replace-session-connection replace-origin"); } if (has_body("application/sdp")) { rtpengine_offer(); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_WARN", "failure route\n"); route(NATMANAGE); if (t_is_canceled()) exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xlog("Retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

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Can Kamailio help with this?
by Antony Stone 22 May '21

22 May '21

Hi. I've been aware of Kamailio, and on this list, for several years, but so far I have not implemented Kamailio for any purpose. I wonder if it could be a suitable tool for the following scenario. I currently use Asterisk to register as a client, with username and password, to other SIP PBXs in order to receive calls. As far as the other PBX is concerned, Asterisk looks like a SIP telephone on a particular extension. However, Asterisk's SIP client capabilities are limited to handling phone calls only, and it specifically cannot receive presence information about the state of other extensions from the remote PBX. So, could Kamailio be used in the path between Asterisk and the remote PBX so that the registration on the PBX comes from Kamailio (with any incoming calls being passed to Asterisk), but with Kamailio also receiving presence information from the remote PBX and making this available to some script or application (which I would expect to have to write)? The main point is that I don't want to have two things registering to the remote PBX, one for calls and one for presence, so I'm looking for a way to register with something which understands presence, and can also pass calls on to Asterisk. Can anyone suggest whether Kamailio could do this, and if so, point me at some resources to help me get started; or alternatively suggest ideas on some other tool which might be more appropriate than Kamailio? Thanks for any ideas, Antony. -- Schrödinger's rule of data integrity: the condition of any backup is unknown until a restore is attempted. Please reply to the list; please *don't* CC me.

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core kamailio 5.4
by David Escartín 21 May '21

21 May '21

Dear all we have seen a core in a kamailio instance today, i attach you the bt in dropbox https://www.dropbox.com/s/3dg72cbuz7nvx5b/core_kamailio_21mayo?dl=0 https://www.dropbox.com/s/iyko32fwfxj8aam/core_kamailio_21mayo_2?dl=0 seems there might be a relation async module when delaying a request after communicating with the rtpengine ?¿ could you please take a look? thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

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Kamailio Stop Processing Call
by M Arqum CH 21 May '21

21 May '21

Hello Members, I have kamailio setup backed with mysql db, using db_cluster module for spliting read and write.. Lisiting on three different virtual ips are bind with same/single udp port *1* for processing calls. One static IP is bind with udp port*2* to for processing/response to option messages send from keepalived external script after every few seconds and for DMQ messages from other kamailio node. General configuation details: children=16 fork=yes listen=udp:VIP:SIP_PORT1 listen=udp:VIP2:SIP_PORT1 listen=udp:VIP3:SIP_PORT1 listen=udp:StaticIP3:SIP_PORT2 listen=tcp:SIP:RPC_PORT SH_MEM=5012 PRIV_MEM=264 System Specification : 24 GB RAM, 12 CPUs Sometimes kamailio radomly stop processing the requests or give no response on new requests recevied on virtual ips and as well it did't show any error in syslog. To monitor the kamailio memory I used promethues ( https://github.com/florentchauveau/kamailio_exporter) The moment kamailio stop processing the requests it also stop responding to kamailio exporter and scrap request get timeout. i.e error log on console debian prometheus-kamailio-exporter[83440]: 2021/05/20 12:15:27 [error] cannot read header: read unix @->/tmp/kamailio__ctl: i/o timeout After getting first timeout occurrence of kamailio-exporter scrap request, kamailio processed sip messages for almost 30 seconds then completly stop replying to any message. But it keep replying to option message arrive on different port from keepalived script. Stats : Active dialog: 75 Shared memory used: 150 MB Kamailio Installed version : kamailio -v version: kamailio 5.4.1 (x86_64/linux) 2ca223 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 2ca223 compiled on 12:51:24 Sep 21 2020 with gcc 6.3.0 Is anyone else face similar issue or idea why kamailio is showing such behaviour.? Please guide me on what I should do to find and fix the issue. Thanks -- Regards M Arqum

2 2

Video transcode for doorbell
by Luke Kenny 21 May '21

21 May '21

I have a SIP video doorbell device that is only capable of sending H264 and MJPEG video. The site is using 3CX and its associated mobile app, which only supports VP8 and VP9. Without properly reading the documentation, I went ahead and set up kamailio and rtpengine to sit in between the doorbell and 3CX. I has assumed rtpengine was capable of transcoding video as well as audio, however upon proper inspection it appears that isn't the case. Is there a kamailio module that is capable of video transcoding, even if it isn't real time, or at a reduced framerate? As the application is a doorbell, it doesn't need to be perfect. Many thanks, Luke Kenny

1 0

Eternal config - Kamailio and rtpengine
by Mihai Cezar 20 May '21

20 May '21

Hi! I use lxc containers with asterisk wich their are connecting to kamailio, so far so good they work when i put the containers with public ips, but when i switched to private ip I have one-way audio. I tried to with rtpengine but no luck: rtpengine --table=42 --listen-ng=127.0.0.1:2223 --interface=56.12.19.100 --tos=184 --pidfile=/run/rtpengine.pid --no-fallback --log-level=7 --log-facility=local3 the setup looks like this: VM-KAMAILIO-(public ip 56.12.19.100 with static routes for the private IP) --> LxC(n) (with private class 192.168.0.n) The problem i think is when LxC asterisk is sending back the 200 OK from Invite it puts his private ip in the sdp filed: v=0 o=- 2171376102 3 IN IP4 192.168.0.2 s=Asterisk c=IN IP4 192.168.0.2 t=0 0 m=audio 11336 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv i've loaded the modules, but after this i am stuck.... loadmodule "nathelper.so" loadmodule "rtpengine.so" modparam("nathelper", "received_avp", "$avp(RECEIVED)") modparam("nathelper", "nortpproxy_str", "a=sdpmangled:yes\r\n") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@proxy") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_retr", 2) modparam("rtpengine", "hash_table_size", 256) modparam("rtpengine", "hash_table_tout", 600)

2 3

Text transformations s.select and newline char
by Martin Nyström 20 May '21

20 May '21

Hello, I am simply trying to parse a SDP body, read it line by line and save the line into a variable for further processing. It seems when using a newline \n as the delimiter it breaks down with errors. How should I treat this issue and properly managed the newline as delimiter? May 17 12:00:00 ip-10-120-120-12 kamailio[12878]: 0(12878) ERROR: pv [pv_trans.c:2326]: tr_parse_string(): invalid select transformation: select,$var(i),$var(delimeter)}!! May 17 12:00:00 ip-10-120-120-12 kamailio[12878]: 0(12878) ERROR: <core> [core/pvapi.c:1717]: tr_lookup(): error parsing [{s.select,$var(i),$var(delimeter)}] Block extracted from .cfg sdp_get("$avp(sdp)"); $avp(lines) = $(avp(sdp){line.count}); $var(delimeter) = "\n"; $var(i) = 0; while($var(i) < $avp(lines)) { # This works xlog("L_INFO", "==> Position $var(i): $(avp(sdp){s.select,$var(i),\n}) \n"); # This fails $var(l1) = $(avp(sdp){s.select,$var(i),$var(delimeter)}); # This fails $var(l2) = $(avp(sdp){s.select,$var(i),\n}); # This fails $var(l3) = $(avp(sdp){s.select,$var(i),\\n}); # Supposed to do something funny with the $var(line) here $var(i) = $var(i) + 1; } /M

3 3

5.4 dependency package question
by Andrew Chen 19 May '21

19 May '21

Hi guys, I'm trying to build a deb file of kamailio 5.4 and when doing dpkg-buidpackage it gave a bunch of dependency warnings. I was able to take care a good number of them until I ran into this: *dpkg-checkbuilddeps: **error*: Unmet build dependencies: gcj-jdk libval-dev libxmlrpc-c3-dev None of these packages are in the "apt-cache search" output. Below is a list of modules currently loaded in 5.2.5 and want to use them again: INCLUDE_MODULES="exec rtpengine topoh carrierroute statsd dialplan kex tm tmx sl rr pv maxfwd usrloc textops sdpops siputils xlog sanity ctl acc dispatcher cfgutils permissions domain presence presence_xml nathelper dialog tls htable pike db_mysql userblacklist sqlops websocket xhttp siptrace tcpops jsonrpcs rtimer" Thanks -- Andy Chen Sr. Telephony Lead Engineer 415 516 5535 (M) achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 18

Permissions sometimes does not find group
by Romain Untereiner 19 May '21

19 May '21

Hello all, I am having some trouble with Permissions module of Kamailio, regarding the allow_source_address_group() function. In 99.9% of the cases the IP of our partner is allowed as expected, but randomly it can fail (returning -1 for the same IP address that was allowed for many other calls in the meantime). I can’t understand why it fails and why only « sometimes » - I confirm this append on the same Kamailio box, with same IP and Port of the partner. In my custom logs: « X.X.X.X:YYYY is allowed through group Z » —> as expected and sometimes (like 1% of the cases): « X.X.X.X:YYYY is not allowed - allow_source_address_group returned -1» It is maybe to note that in the address table, this range is defined as /24 and not /32. Until now it never happened on /32 defined address but I don’t know if it can be correlated. Would someone have a clue on this? Versions: - kamailio 5.5.0 (x86_64/linux) d4c1a1 - debian 9 stretch - postgresql 11.7 Many thanks! Romain Untereiner Co founder at Cantoo www.cantoo.co

2 8

Kamailio as Load Balancer
by Melek Oktay 19 May '21

19 May '21

Hi, I have been investigating distributing incoming/outgoing SIP traffic to behind Kamailio SIP servers in order to scale and other advantages too!. SRV type distribution is not a good option, it is already tested. So, I am trying to deploy Kamailio as a *LoadBalancer* in Tier1 , so it distributes traffic backbone Kamilio servers in Tier 2. There is a good presentation as follows, actually I want to implement it. https://www.kamailio.org/events/2013-KamailioWorld/23-Daniel-Constantin.Mie… However, I could not find sample configurations and sample study ! Would you show me sample configurations that Kamailio uses as a Load Balancer if you have it. Current Kamailio responsibilities in my system are: 1. Registar Server 2. Presence Server 3. Sip proxy server 4. WebSocket Maybe it will be very good to separate *Registar *Server & *Presence *Server, but could not find sample configurations too. Best Regards Melek P.S. If we could collect such configurations, I would share them for community on Github.

2 4

(no subject)
by Melek Oktay 18 May '21

18 May '21

Hi, I have been investigating distributing incoming/outgoing SIP traffic to behind Kamailio SIP servers in order to scale and other advantages too!. SRV type distribution is not a good option, it is already tested. So, I am trying to deploy Kamailio as a *LoadBalancer* in Tier1 , so it distributes traffic backbone Kamilio servers in Tier 2. There is a good presentation as follows, actually I want to implement it. https://www.kamailio.org/events/2013-KamailioWorld/23-Daniel-Constantin.Mie… However, I could not find sample configurations and sample study ! Would you show me sample configurations that Kamailio uses as a Load Balancer if you have it. Current Kamailio responsibilities in my system are: 1. Registar Server 2. Presence Server 3. Sip proxy server 4. WebSocket Maybe it will be very good to separate *Registar *Server & *Presence *Server, but could not find sample configurations too. Best Regards Melek P.S. If we could collect such configurations, I would share them for community on Github.

1 0

Question about Kamailio IMS in box
by Ibrahim soumana djibo 17 May '21

17 May '21

Hello, I am new to kamailio. And I Am following the work of Franz Edler. I would like to test it in a local environment without internet connection on ubuntu. What I don't understand is, to download the packages I necessarily need the internet. Do I have to use two network interfaces for the same virtual machine in order to access the internet and make the DNS configuration locally with the second interface? Cordially, *SOUMANA** DJIBO **Ibrahim* *Ingénieur Réseaux & Télécoms* *5 Avenue Emile Baudot, 91300 Massy* +33 07 61 80 20 35 [image: Résultat de recherche d'images pour "logo paris saclay"] https://www.universite-paris-saclay.fr/

2 1

Getting MOS feedback from rtpengine
by Barry Flanagan 17 May '21

17 May '21

Hi Using Kamailio 5.4.4 on Debian Buster. I am trying to get feedback of the MOS scores at call end. However the variables are always <null>. I have the modparams set as follows: # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:localhost:2223") modparam("rtpengine", "rtpengine_disable_tout", -1) modparam("rtpengine", "rtpengine_tout_ms", 2000) modparam("rtpengine", "rtpengine_allow_op", 1) modparam("rtpengine", "mos_min_pv", "$avp(s:mos_min)") modparam("rtpengine", "mos_max_pv", "$avp(s:mos_max)") modparam("rtpengine", "mos_average_pv", "$avp(s:mos_avg)") modparam("rtpengine", "mos_average_packetloss_pv", "$avp(s:mos_avg_packetloss)") modparam("rtpengine", "mos_average_jitter_pv", "$avp(s:mos_avg_jitter)") modparam("rtpengine", "mos_average_roundtrip_pv", "$avp(s:mos_avg_roundtrip)") I am trying to log the values from within the dialog:end event_route: event_route[dialog:end] { rtpengine_delete(); xlog("L_INFO","mos_avg=$avp(mos_avg), packetloss_avg=$avp(mos_avg_packetloss), jitter_avg=$avp(mos_avg_jitter), latency_avg=$avp(mos_avg_roundtrip), sip_last_reply=$rs - $ci\n"); return; } Am I missing something? I have tried rtpengine_query and rtpengine_manage in place of the _delete and also tried placing this in the BYE handling section of WITHIN_DIALOG but no joy. Any assistance much appreciated! -Barry -- -Barry

5 10

Hunting Kamailio.cfg for Voice termination System
by Edo 16 May '21

16 May '21

Hi, I am looking to get a complete kamailio.cfg file that I can use to set up a voice termination system. This system will only connect with peers making multiple simultaneous calls, registering strictly via ip, accounting calls to sql db, capable of multiple rate tables for the different peers, call control, CDR tool and billing capable and call routing to multiple media servers. I am willing to try to build this myself but at this time I do not even understand why I need to use "record_route" or "t_precheck_trans" -- :)). I am a lover of information and cannot find a simple enough text to help in my kamailio journey -- unlike asterisk. If you are interested please respond to me directly at edo.ekunwe(a)gmail.com and I am willing to pay for this information or guidance(if you have the patience and time to mentor me). Thanks -- ----- Nogie Ekunwe

3 4

Help to overwrite the $od, or another solution
by Victor Velo 15 May '21

15 May '21

Hello, I'm facing to some challenge in my learning process and using Kamalio as MS Teams SBC, the request I receive from MS Teams seems to be correct but I need to overwrite the Original Domain, which is the FQDN of Kamailio itself, to another domain name: INVITE sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone;transport=tls SIP/2.0Record-Route: <sip:155.138.140.233;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>Record-Route: <sip:155.138.140.233:5061;transport=tls;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>FROM: Victor Velo<sip:+14507001777@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3e2ca0c5d0da4335b121f205a6f20541TO: <sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone> rewrite to: INVITE sip:+15142144069@dev-sbc.openfabrik.com:5061;user=phone;transport=tls SIP/2.0Record-Route: <sip:155.138.140.233;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>Record-Route: <sip:155.138.140.233:5061;transport=tls;r2=on;lr=on;ftag=3e2ca0c5d0da4335b121f205a6f20541>FROM: Victor Velo<sip:+14507001777@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3e2ca0c5d0da4335b121f205a6f20541TO: <sip:+15142144069@dev.myinternaldomain.openfabrik.com:5061;user=phone> What would the best why to achieve this rewriting? Thanks, Victor

2 2

loose_route() and FQDN
by Igor Olhovskiy 14 May '21

14 May '21

Hello, I saw there are some topics on this already and carefully walked through all of them, but can't solve following issue. For a reason I do need that in Record-Route header sent to endpoint would present FQDN. If it matters, it's UDP/WSS conversion done on Kamailio. So, scheme is quite simple Enpoint A ->UDP-> Kamailio ->WSS-> Endpoint B (NAT) Main issue here, that if in Record-Route headers it's FQDN, but not IP addresses, on a new transactions with a dialog (ACK on 200, PRACK, BYE), Kamailio*loose_route()* function resolves address of destination not current dialog, but actual R-URI (or itself, if R-URI is something from WebRTC world) that is not correct due to NAT. If in RR headers IP addresses are present - all is working as expected. As an example (RR with FQDN) B answers 200 SIP/2.0 200 OK Record-Route: <sip:KAMAILIO_FQDN:8089;transport=ws;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Record-Route: <sip:KAMAILIO_FQDN;r2=on;lr=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Via: SIP/2.0/UDP <A_IP_ADDRESS>:5060;received=A IP ADDRESS;rport=5060;branch=z9hG4bKPj67fb6d86-97d7-4231-995b-e54b0f62881e To: <sip:88290@<KAMAILIO_FQDN>>;tag=hvra7mj3q0 From: <sip:+XXXX7688881@<KAMAILIO_FQDN>>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb CSeq: 18326 INVITE Contact: <sip:88290@toleivu2gdbh.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound Content-Type: application/sdp Content-Length: 817 ACK looks like ACK sip:88290@toleivu2gdbh.invalid;transport=wss SIP/2.0 Via: SIP/2.0/UDP A_IP_ADDRESS:5060;rport;branch=z9hG4bKPj8d05548a-91ef-4332-8617-32f8eeebf8f2 From: <sip:88881@A_IP_ADDRESS>;tag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d To: <sip:88290@KAMAILIO_FQDN>;tag=hvra7mj3q0 Call-ID: 46f44741-d155-4dd5-8fd8-78e540fc1acb CSeq: 18326 ACK Route: <sip:FQDN;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Route: <sip:FQDN:8089;transport=ws;lr;r2=on;ftag=0a3e31a6-96ad-42d0-9310-81b35cedbd3d;nat=wss> Max-Forwards: 70 User-Agent: Asterisk PBX 13.33.0 Content-Length: 0 And Kamailio on *loose_route()* loops ACK to itself. (with result of function == 1) In a case if in Record-Route/Route headers just IP address of Kamailio present, all works as expected, but it's not really behavior I want to achieve. I've tried to play with *record_route_preset("...")* specifying only WSS part (as suggested in https://skalatan.de/de/blog/kamailio-sbc-teams) with FQDN, but no luck. Also wanted to try approach using record_route_preset() function in branch route, but it was working only with first branch, not affecting others (but I assume having different RR headers across branches is not a good idea) -- Regards, Igor

4 18

Setting $conid per branch
by Marrold 14 May '21

14 May '21

Hi all, I'm currently adding a feature to our Kamailio configuration to fork calls based on user agent. To do so I'm getting the registered endpoints with reg_fetch_contacts() iterating through and matching on them, then using seturi() / append_branch() and setting the dst-uri and flags as required. However, calls are intermittently going to the wrong TCP connection, despite the logs showing the correct destination IP and port in the ONSEND route. Looking in the location table I can see each registration has a connection_id, and the debug log indicates it's finding the connection by ID: DEBUG: <core> [core/tcp_main.c:1651]: _tcpconn_find(): found connection by id: 3 Is there a way to set the conid per branch? Is it necessary? We're using kamailio 5.3.3 on Debian 10. Thanks Matthew

2 8

Possible to get the value of $DLG_lifetime before the BYE hits loose_route()
by ryan embgrets 13 May '21

13 May '21

Greetings, I would like to know if it is possible to get the value of $DLG_lifetime before the BYE hits loose_route(). opensips provides a similar function which lets you get dialog life. https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_load_dialog_c… But i could not get the same from kamailio or maybe kamailio has a better way for this? Thanks for any pointer. Ryan

1 0

Bad config - you can not call 'handle_publish' function (db_url not set)
by Marrold 13 May '21

13 May '21

Hi, We're trying to upgrade from 5.3.X to 5.4.5 however kamailio will not start and produces the following error, despite the db_url being set in the config and printed in the logs: Bad config - you can not call 'handle_publish' function (db_url not set) Looking at the code this error only seems plausible if the *pres_library_mode* is set to 1 here <https://github.com/kamailio/kamailio/blob/cc2a9518207e36b7b18aef5c8714f0c21…> but we don't see the corresponding "switch to library mode" message in the debug logs which leaves me a bit stuck - I've included the grepped logs at the bottom of the message. Does anyone have any ideas what is causing this issue? I've seen a previous issue relating to interactions between presence and presence_xml which we're also using, but it looks like this was fixed in an older version. Any suggestions appreciated. # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 60) modparam("presence", "local_log_facility", "LOG_LOCAL3") modparam("presence", "max_expires", 14430) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) root@kamailio:/etc/kamailio# grep -i "presence" /var/log/kamailio/kamailio.log May 10 12:44:14 kamailio /sbin/kamailio[23729]: ERROR: presence [presence.c:656]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/kemi.c:2927]: sr_kemi_modules_add(): adding module: presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:287]: register_module(): register PV from: presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_xml.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_xml.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/kemi.c:2927]: sr_kemi_modules_add(): adding module: presence_xml May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_xml May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_mwi.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_mwi.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_mwi May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.y:1810]: yyparse(): loading module presence_dialoginfo.so May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:525]: load_module(): trying to load </lib64/kamailio/modules/presence_dialoginfo.so> May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/cfg.lex:1796]: pp_define(): defining id: MOD_presence_dialoginfo May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_url> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_url> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_update_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_update_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <clean_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <clean_period> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <local_log_facility> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <local_log_facility> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence' matches module 'presence' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <max_expires> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <max_expires> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence_xml' matches module 'presence_xml' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <db_url> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <db_url> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:107]: set_mod_param_regex(): 'presence_xml' matches module 'presence_xml' May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/sr_module.c:744]: find_param_export(): found <force_active> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:41 kamailio kamailio: DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <force_active> in module presence_xml [/lib64/kamailio/modules/presence_xml.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <handle_publish> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <handle_subscribe> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:43 kamailio kamailio: DEBUG: <core> [core/route.c:129]: route_add(): mapping routing block (0x560e73f9a440)[PRESENCE] to 11 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [presence.c:293]: mod_init(): db_url=mysql://kamailio:PASSWORD@localhost/kamailio (len=50 addr=0x7f3a33d4e288) May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [presence.c:337]: mod_init(): server_address parameter not set in configuration file May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2529]: restore_db_subs(): loading information from database 0 records May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_xml May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [presence_xml.c:170]: mod_init(): db_url=mysql://kamailio:PASSWORD@localhost/kamailio (len=50 addr=0x7f3a33d4ef00) May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence/1] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence.winfo/2] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: presence - len= 8 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:75]: xml_add_events(): added 'presence' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence.winfo/2] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [presence/1] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:188]: add_event(): Found wipeer event [presence] for event [presence.winfo] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: presence.winfo - len= 14 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:94]: xml_add_events(): added 'presence.winfo' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [xcap-diff/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [xcap-diff.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: xcap-diff - len= 9 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [add_events.c:132]: xml_add_events(): added 'xcap-diff' event to presence module May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_mwi May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [message-summary/6] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [message-summary.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: message-summary - len= 15 May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:873]: init_mod(): presence_dialoginfo May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_presence> in module presence [/lib64/kamailio/modules/presence.so] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [dialog/5] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:324]: search_event(): start event= [dialog.winfo/0] May 10 12:44:45 kamailio /sbin/kamailio[23830]: DEBUG: presence [event_list.c:239]: add_event(): successfully added event: dialog - len= 6 May 10 12:44:47 kamailio /sbin/kamailio[23830]: ERROR: presence [presence.c:656]: fixup_presence(): Bad config - you can not call 'handle_publish' function (db_url not set) May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence_xml [presence_xml.c:295]: destroy(): start May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2433]: timer_db_update(): db_update timer May 10 12:44:47 kamailio /sbin/kamailio[23830]: DEBUG: presence [subscribe.c:2137]: update_db_subs_timer(): update_db_subs_timer: start

3 6

secsipid
by David Villasmil 12 May '21

12 May '21

Hello guys, I'm trying to use secsipid as per https://kamailio.org/docs/modules/5.5.x/modules/secsipid.html I just installed go on a new vm: the env is as follows: # env GOPATH=/root/go then: ~# go get github.com/asipto/secsipidx ~# Then: ~# cd $GOPATH/src/github.com/asipto/secsipidx/csecsipid/ bash: cd: /root/go/src/github.com/asipto/secsipidx/csecsipid/: No such file or directory it looks like it's not downloaded where the documentation says. It is however in: go |-- bin `-- pkg |-- mod | |-- cache | | `-- download | | |-- github.com | | | |-- asipto | | | | `-- secsipidx | | | | `-- @v | | | `-- google | | | `-- uuid | | | `-- @v | | `-- sumdb | | `-- sum.golang.org | | |-- lookup | | | `-- github.com | | | |-- asipto | | | `-- google | | `-- tile | | `-- 8 | | |-- 0 | | | |-- x010 | | | |-- x011 | | | `-- x016 | | | `-- 560.p | | |-- 1 | | | `-- 064.p | | `-- 2 | | `-- 000.p | `-- github.com | |-- asipto | | `-- secsipidx(a)v1.0.0 | | |-- csecsipid | | `-- secsipid | `-- google | `-- uuid(a)v1.2.0 `-- sumdb `-- sum.golang.org 38 directories But going into ~/go/pkg/mod/github.com/asipto/secsipidx(a)v1.0.0/csecsipid we have: ~/go/pkg/mod/github.com/asipto/secsipidx(a)v1.0.0/csecsipid# ls -l total 16 -r--r--r-- 1 root root 231 May 11 13:14 Makefile -r--r--r-- 1 root root 1778 May 11 13:14 csecsipid.go -r--r--r-- 1 root root 1988 May 11 13:14 libsecsipid.h -r--r--r-- 1 root root 120 May 11 13:14 secsipid.h The Makefile contains the proper liba: # Makefile to build libraries # all: libso libso: go build -o libsecsipid.so -buildmode=c-shared csecsipid.go liba: go build -o libsecsipid.a -buildmode=c-archive csecsipid.go clean: rm -f libsecsipid.so rm -f linsecsipid.a But "make liba" returns: # make liba go build -o libsecsipid.a -buildmode=c-archive csecsipid.go csecsipid.go:6:2: no required module provides package github.com/asipto/secsipidx/secsipid: go.mod file not found in current directory or any parent directory; see 'go help modules' make: *** [Makefile:10: liba] Error 1 Obviously not a "go" guy here. What am I doing wrong? Thanks for your help! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 9

bogus event with terminated dialogs with late 200ok responses
by David Escartín 12 May '21

12 May '21

Hello all using kamailio version 5.4.4, we are seeing some errors like CRITICAL: dialog [dlg_hash.c:1205]: next_state_dlg(): bogus event 3 in state 5 (with BYE) for dlg 0x7f6ca2304b48 [15135:39671] with clid '00463385_20210512_002_HPRA_1.1.4.40.sp3_7532_0_54533900281' and tags 'Out-55089-0000000CB27947F9' '170cB11c48vmj' which seem to be generated when we receive a BYE from the remote UA but we were expecting an ACK also, For example INVITE ----------> <---------------183 <---------------200 BYE -------------> (no ACK) <--------------200 <--------------200 .... so the b2bua (freeswitch) keeps retransmitting the 200ok, and on each 200ok retransmission we are getting this type of message. These seem to be like race conditions since 200/BYE flow almost at the same time, and as I understand, the UA should send the ACK. But my doubt is if these errors can be harmful for the kamailio or they can be ignored and address the issue to the remote UA on each case thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

2 1

Kamailio as front proxy for multiple sip servers
by Eliphas Levy Theodoro 10 May '21

10 May '21

Hello! I am trying to config one kamailio as reverse proxy for a bunch of internal (no internet address) separate asterisk sip instances (per domain). The kamailio server would be the only with the valid IP address, so would use rtpengine to force to be in the media path. Like this scenario: https://opensips.org/pipermail/users/2020-August/043610.html I have used as starting point this very basic config: https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-reques… Basically just added rtpproxy support, and voilà, inter-SIP is working, media always passing into the proxy. The problem: I would have WebRTC phones connecting too. I tried setting WSS up in kamailio, and asterisk (pjsip) wouldn't know how to send the message to the proxy: on register it has trasnport=wss in the contact: header, looks like it is confusing the asterisk. So, I resort for the wisdom of the list :) What would be the good-best-path to take here, hack the header, or put the webphones registering directly on the asterisks (with a nginx reverse proxy maybe)? Someone must have already made a blog with such setup, but I could not google-unearth it at least until now. Regards, -- Eliphas

5 7

MS Teams SBC
by Victor Velo 09 May '21

09 May '21

Hello, I'm trying to test a Henning's tutorial for Teams SBC described here: https://skalatan.de/en/blog/kamailio-sbc-teams. A minor change at the MS Teams setup is just about the login. Now it requires MFA.The license I used is : Microsoft 365 developer subscriptions (E5).The domanin dv-sbc.openabrik.com is added + 1 user PS> Connect-MicrosoftTeams (MFA)PS> New-CsOnlinePSTNGateway -Fqdn dev-sbc.openfabrik.com -SipSignalingPort 5061 -MaxConcurrentSessions 10 -ForwardCallHistory $true -Enabled $true Everything seems to be OK at MS Teams side. I'm using the Kamailio coming with Kazoo. Version 5.2.5[root@dev-sbc kamailio]# kamailio -versionversion: kamailio 5.2.5 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknown compiled on 01:14:59 Nov 19 2019 with gcc 7.3.1 When Kamailio is probing MS Teams, the log shows the following errors (for the 3 GW): hG4bKc188.6a606040000000000000000000000000.0#015#012To: <sip:sip2.pstnhub.microsoft.com;transport=tls>#015#012From: <sip:dev-sbc.openfabrik.com>;tag=d9a52ca041d9e6253dd2a7d5399056e3-2f56#015#012CSeq: 10 OPTIONS#015#012Call-ID: 4ed99b8a73a96dcd-10234(a)155.138.140.233#015#012Max-Forwards: 70#015#012Content-Length: 0May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: <core> [core/tcp_main.c:618]: _wbufq_add(): (517 bytes): write queue full or timeout (0, total 1100374016, last write 41683154 s ago)May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: <core> [core/tcp_main.c:2673]: tcpconn_1st_send(): 52.114.76.76:5061: EAGAIN and write queue full or failed for 0x7fce434db860May 9 01:26:17 dev-sbc kamailio[10234]: ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failedMay 9 01:26:17 dev-sbc kamailio[10234]: ERROR: tm [uac.c:648]: send_prepared_request_impl(): Attempt to send to precreated request failedMay 9 01:26:17 dev-sbc kamailio[10234]: INFO: <script>: Sent out tm request to Teams: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0#015#012Via: SIP/2.0/TLS 155.138.140.233:5061;branc config: tls.cfg [server:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/openfabrik.com/privkey.pemcertificate = /etc/letsencrypt/live/openfabrik.com/fullchain.pemca_list = /etc/letsencrypt/live/openfabrik.com/bc2025.pemserver_name = dev-sbc.openfabrik.com [client:default]method = TLSv1.2+verify_certificate = yesrequire_certificate = yesprivate_key = /etc/letsencrypt/live/openfabrik.com/privkey.pemcertificate = /etc/letsencrypt/live/openfabrik.com/fullchain.pemca_list = /etc/letsencrypt/live/openfabrik.com/bc2025.pem ==== event_route[tm:local-request]{ setflag(FLAG_LOCAL_REQUEST); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:dev-sbc.openfabrik.com:5061;transport=tls>\r\n"); xlog("L_INFO", "Sent out tm request to Teams: $mb\n"); } xlog("L_DEBUG", "$ci|local|start $pr request $rm $ou\n"); xlog("L_DEBUG", "$ci|local|source $si:$sp -> $dd:$dp\n"); xlog("L_DEBUG", "$ci|local|from $fu\n"); xlog("L_DEBUG", "$ci|local|to $tu\n"); #!ifdef SIP_TRACE_ROLE route(SIP_TRACE); #!endif #!ifdef PRESENCE_ROLE route(PRESENCE_LOCAL_REQUEST); #!endif } ====# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)kamctl dispatcher add 2 'sip:sip.pstnhub.microsoft.com;transport=tls' 0 3 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com'kamctl dispatcher add 3 'sip:sip2.pstnhub.microsoft.com;transport=tls' 0 2 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com'kamctl dispatcher add 4 'sip:sip3.pstnhub.microsoft.com;transport=tls' 0 1 'socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com' [root@dev-sbc kamailio]# kamctl dispatcher showdispatcher gatewayssqlite_query: /usr/bin/sqlite3 /etc/kazoo/kamailio/db/kazoo.db select * FROM dispatcher ORDER BY setid; 1|1|sip:155.138.140.233:11000|9|0|zone=local;profile=sipinterface_1;duid=243;node=freeswitch@dev-sbc.openfabrik.com|added by nodes role2|2|sip:sip.pstnhub.microsoft.com;transport=tls|0|3|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com|3|3|sip:sip2.pstnhub.microsoft.com;transport=tls|0|2|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com|4|4|sip:sip3.pstnhub.microsoft.com;transport=tls|0|1|socket=tls:155.138.140.233:5061;ping_from=sip:dev-sbc.openfabrik.com| Is it a bug on the Kamailio 5.2.5 release or Do I miss something? Thanks, Victor

2 3

Unable to parse the Cancel request
by Ilie Soltanici 05 May '21

05 May '21

Hello, We are having an issue here where Kamailio is complaining about duplicate To header, while in the SIP Packet there is just one. See below the log message: DEBUG: tm [t_msgbuilder.c:391]: build_local_reparse(): duplicate To header ERROR: tm [t_msgbuilder.c:531]: build_local_reparse(): cannot build CANCEL request ERROR: tm [t_cancel.c:297]: cancel_branch(): attempt to build a CANCEL failed ERROR: tm [t_fwd.c:1434]: e2e_cancel(): cancel error And this is the sip message: CANCEL sip:123456789@1.2.3.4:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:19373 ;branch=z9hG4bK-524287-1---7774073fad1a3264;rport Max-Forwards: 70 To: <sip:123456789@1.2.3.4:5060> From: "101"<sip:101@1.2.3.4:5060>;tag=6f373b66 Call-ID: hC2O6zx8ZaUJ1di046 CSeq: 2 CANCEL Proxy-Authorization: Digest username="101",realm="1.2.3.4",nonce="YJKbfWCSmlFTHxdJfKmySydF9L7FzfaqcgyOo4A=",uri=" sip:123456789@1.2.3.4:5060 ",response="2bf2ba8ef2374d6aa8c39d650613077d",algorithm=MD5 User-Agent: TEST Content-Length: 0 version: kamailio 5.5.0 (x86_64/linux) d4c1a1 The route itself, is like in the standard documentation: # CANCEL processing if KSR.is_CANCEL() : KSR.info(RANDOM_LOG + 'KSR.is_CANCEL') if KSR.tm.t_check_trans()>0 : self.ksr_route_relay(msg) return 1 def ksr_route_relay(self, msg): if KSR.is_method_in("IBSU") : if KSR.tm.t_is_set("branch_route")<0 : KSR.tm.t_on_branch("ksr_branch_manage") if KSR.is_method_in("ISU") : if KSR.tm.t_is_set("onreply_route")<0 : KSR.tm.t_on_reply("ksr_onreply_manage") if KSR.is_INVITE() : if KSR.tm.t_is_set("failure_route")<0 : KSR.tm.t_on_failure("ksr_failure_manage") if KSR.tm.t_relay()<0 : KSR.sl.sl_reply_error() return -255 Thanks

2 7

start_recording and stop_recording inside event_route[xhttp:request]
by Володимир Іванець 05 May '21

05 May '21

Hello! I'm testing call recording with Rtpengine. It works fine when the "record-call=on" flag is added to the *rtpengine_offer* or *start_recording* is used in the *request_route*. But I was wondering if the call recording can be managed by a separate application. So I add the following lines to the *event_route[xhttp:request]* and triggered it with an HTTP request after the call was established. The call-id value was taken from the Rtpengine log and sent with the request. * if ($hu =~ "^/CALL_RECORD_START/") {* * $var(call_id) = *"call-id=" + *$(hu{s.select,2,/});* xlog("L_DBG", "$var(call_id)"); * start_recording($var(call_id));* exit; * }* In this case, Kamailio will log errors and will not send the Rtpengine command to start call recording. *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <script>: call-id=3de74068278ede4417086eb82bdf5a06@127.0.0.1:5080.0* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine_funcs.c:294]: get_callid(): call-id not found* *Apr 27 18:25:00 test /usr/sbin/kamailio[5347]: ERROR: rtpengine [rtpengine.c:2383]: rtpp_function_call(): can't get Call-Id field* Have anyone tried to call functions *start_recording* and *stop_recording* inside *event_route[xhttp:request]*? Thanks a lot!

3 12

Kamailio v5.5.0 Released - new major version is out
by Daniel-Constantin Mierla 05 May '21

05 May '21

Kamailio v5.5.0 is out – it comes with 6 new modules and a considerable set of improvements touching again more than 80 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-5-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.5.0 brings more flexibility and optimizations across many components, enhancements to distributed message queue (dmq module), load balancer, http client, STIR/SHAKEN support, IMS/VoLTE, rtpengine and tls as well as to make available more functions to KEMI interface ... just to list only a few here. Enjoy Kamailio v5.5.0! Thank you for flying Kamailio! Stay safe and healthy! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

dialog OPTIONS
by David Villasmil 05 May '21

05 May '21

Hello guys, I'm using the dialog module, params are: # ----- dialog params ----- modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", 20) modparam("dialog", "ka_timer", 10) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_failed_limit", 2) modparam("dialog", "send_bye", 1) modparam("dialog", "timer_procs", 1) When receiving an INVITE, i'm doing dlg_set_property("ka-src"); dlg_set_property("ka-dst"); dlg_set_property("timeout-noreset"); I have a couple questions: The flow is the following: clientA>-\ kamailio<->freeswitch clientB<-/ Kamailio receives an INVITE from ClientA, which I forward to freeswitch, which in turn forwards it back to Kamailio. In reality I would like to monitor only the client legs, but I'm having problems, so I'm enabling OPTIONS on all legs. On the ClientA legs, I see OPTIONS going from kamailio to both freeswitch and the client. But on the ClientB legs i don't see any OPTIONS at all. Can someone point me in the right direction to enable it on all legs? Something else i'm seeing is even though i set "ka_failed_limit" to 2, kamailio is sending several OPTIONS before sending BYEs to both legs (clientA, since there are no OPTIONS on clientB) Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 1

The plan for releasing Kamailio v5.5.0
by Daniel-Constantin Mierla 05 May '21

05 May '21

Hello, after almost 4 weeks since development was frozen, it is time to plan releasing the next major stable version, respectively 5.5.0, therefore I propose to do it on Wednesday, May 5, 2021. The branch 5.5 was already created about one week ago, several doc resources were published (core/variables/transformations/rpcs/stats cookbooks, alphabetic indexes, ...). If anyone is testing an upgrade from branch 5.4 to 5.5 and encounters changes that should be done in Kamailio config, add notes about at: * https://www.kamailio.org/wiki/install/upgrade/5.4.x-to-5.5.0 Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online May 17-20, 2021 (Europe Timezone) - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

possible bug on {param.value,name}
by David Villasmil 04 May '21

04 May '21

Hello guys, I'm possible using this wrong, but here it is; I need to get a previously set parameter to rr, named "wsconnid": $var(wsconnid) = $(hdr(Route){param.value,wsconnid}) xlog("L_ERR", "Connection ID parameter: $var(wsconnid)"); And i'm getting: May 3 19:24:45 ip-10-231-32-37 /usr/sbin/kamailio[10015]: ERROR: <script>: WS Connection ID 13> May 3 19:24:49 ip-10-231-32-37 /usr/sbin/kamailio[10014]: ERROR: <script>: Got an OPTIONS We now need to get the connection id from rr This is an example: Route: <sip:10.231.32.37;r2=on;lr;wsconnid=13> The function is returning "13>" instead of "13". Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 8

geting $conid on outbound INVITE
by David Villasmil 04 May '21

04 May '21

Hello guys, I don't have a lot of experience with webrtc (websocket). I'm trying to get $conid when i receive an INVITE via UDP which is GOING to a ws, how do i do that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Re: [SR-Users] Kamailio 5.4.4 and Siptrace
by pwer spire 04 May '21

04 May '21

Hi, Any one have some idea about this? Or how it should be configured in order to still have the original response received in kamailio instead of the mapping one? Thanks in advance, Regards, Tiago

3 7

Can't make tcp_conid_alive work with $var
by David Villasmil 03 May '21

03 May '21

Hello guys, I'm trying to do the following: On INVITE, i add a param to RR "wsconnid" which is the connection id of the WS socket. Afterwards, i get an OPTIONS including said param. Next I want to retrieve the param and check if the connection is alive. NOTE: using simply "$(hdr(Route){param.value,wsconnid})" returns an extra ">" from the end of the string, and i haven't been able to remove that any other way but with "striptail", it works fine. *** $var(conid) = $(hdr(Route){param.value,wsconnid}); $var(conid) = $(var(conid){s.striptail,1}); xlog("L_ERR", "Connection ID parameter: [$var(conid)]"); if (tcp_conid_alive("$var(conid)")) { xlog("L_ERR", "WS Connection ID $var(conid) is alive"); } else { xlog("L_ERR", "WS Connection ID $var(conid) is NOT alive"); } *** With that, "$var(conid)" contains the connection id properly: *** ERROR: <script>: Connection ID parameter: [2] *** but tcp_conid_alive complaints: *** ERROR: <core> [core/sr_module.c:1638]: get_int_fparam(): Could not convert PV to int ERROR: tcpops [tcpops_mod.c:374]: w_tcp_conid_state(): invalid parameter 'conid' (must be a number) *** Using a hardcoded number such as "1" generates no complaints. Kamailio version: version: kamailio 5.3.7 (x86_64/linux) Not sure what's going on here, any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

named account with multiple numbers
by beer Ll 03 May '21

03 May '21

Hi I have a configuration with sip accounts which authenticate with the real name of the customers ex: customer1(a)sip.myvoip.dev customer2(a)sip.myvoip.dev ... INBOUND call of multiple numbers are connected with alias_db to the accounts ex: +331234567890 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +331234567891 | sip.myvoip.dev | customer1 | sip.myvoip.dev | +341234567893 | sip.myvoip.dev | customer2 | sip.myvoip.dev | +341234567894 | sip.myvoip.dev | customer2 | sip.myvoip.dev | ... and all works perfectly. OUTBOUND call after authentication with the username kamailio change the caller id and the ip of my ISP-SIP-TRUNK to relay the call with the correct CID for the ISP using the variable $fn (display name in the From header) [SIP-GW-ISP] { #out with ISP route(CID); $ru = "sip:" + $rU + "@" + "IP_ADDRESS_MY-ISP"; route(RELAY); exit; } ROUTE[CID] { $var(caller) = $(fn{re.subst,/"//g}); uac_replace_from("$var(caller)","sip:$var(caller)@$fd"); } Is possible do it with a different approach for example using Remote-Party-ID , PPI P-Preferred-Identity ? Is possible to check if the customer uses the correct outbound number present in the alias_db table ? ex for customer1 only +331234567890 or +331234567891 numbers many thanks Luca

1 0

connect multiple asterisk to kamailio
by Vahid sadrnia 03 May '21

03 May '21

hi everybody how can i connect multiple asterisk to kamailio and user just connect to specific asterisk(not load balancing),in fact each customer have their own media gateway(asterisk) and use kamailio for topolgy hiding. by regards and thanks.

2 1

Set $uac_req(body) to raw hex value
by Nick 03 May '21

03 May '21

Hi all, It seems in the UAC module the $uac_req(body) PV is evaluated as a string, How can I set the value of the request body of a generated request, to a raw hex value without it being interpreted as a string? Cheers, Nick

2 3

new in Kamailio 5.5.x document updated
by Henning Westerholt 03 May '21

03 May '21

Hello, the "new in Kamailio 5.5.x" document has been updated. Again, a really impressive set of changes that lead to this release: Over 1350 commits have been done in the last development period from over 70 individual contributors. You find the document at the usual place in our wiki: https://www.kamailio.org/wiki/features/new-in-5.5.x Many thanks to everybody who contributed to Kamailio. Help in updating the documentation (e.g. core cookbook, pseudo-variable and transformation section) in the wiki is really welcome, as some content is probably still missing there. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

2 1

potential PRACK related topos bug [solved]
by Juha Heinanen 03 May '21

03 May '21

Thanks to Daniel who patched topos/topos_redis code, topos now works also in situations where requests spiral through the same proxy twice, for example: A - P1 - P2 - P1 - B. -- Juha

2 1

cannot xlog messages into /var/log/syslog
by Abdulaziz Alghosh 02 May '21

02 May '21

Hi Everyone, i am a beginner with Kamaili Server and I am trying to get along with Kamailio config file by logging some messages by xlog Module. xlog module logs two already-written messages which had been delivered with the default configuration file with <script> as a prefix. Unfortunately, I am not able to log anything further neither with xlog() nor xdbg() functions. may someone please advise? Best regards. Abdul

2 1

Dialog server behind proxy and ka_timer/ka_interval
by David Villasmil 30 Apr '21

30 Apr '21

Hello guys, So i have a setup where client | V (WS) [proxy (no dialog tracking)] | V [dialog server] | V [proxy (no dialog tracking)] (WS) | V client I'd like to know when any of the clients are not available (i.e.: crashed, shutdown unexpectedly. etc), so i enabled on the dialog server ka_timer/ka_interval. I see the OPTIONS going to the proxy when the call starts, and the proxy replying with 200 OK. If i shutdown the client, i see the OPTIONS still going, but the proxy is still responding. I assume the Proxy (kamailio) is simply responding to an OPTIONS message and not actually trying to resend it to the client. Is there anyway of doing this or something better to accomplish the same thing? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

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