- sr-users - kamailio.org

Sharing Database between three kamailios
by Waqar 40 01 Aug '22

01 Aug '22

I have three Kamailios in three different servers which are dispatching calls to 9 asterisks in three different servers using load balancing. All these Kamailios are using databases installed on that servers. As the entries of the dispatcher table are the same for all the Kamailio, so I want that these Kamailio use a single database installed on one of the servers. All these Kamailios instances should use the same database which is common between them. And also I want these Kamailios to write into the database by creating a table that stores all the active calls in three Kamailios. Is it possible? How can I do it? Please guide me with somewhat explanation as I am not this much expert in Kamailio. Waiting for a quick response. Regards VoIP Engineer Vicky

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Kamailio behind NAT
by Алексей Якимкин 29 Jul '22

29 Jul '22

Hello, I have kamailio in local network behind NAT. Kamailio have one interface eth1 with ip 10.130.0.23 UserAgent - (internet) - (178.0.0.169)router_with_NAT - (LAN) - kamailio+rtpproxy - pbx I made two listeners: listen=udp:10.130.0.23:5070 #to pbx in lan (I don`t want to put via header with 178.0.0.169) listen=udp:10.130.0.23:5060 advertise 178.0.0.169:5060 #to internet (UserAgent need to get via header with 178.0.0.169 ) mhomed = 1 # I think It not useful for my case UserAgent send Register and Invite to 178.0.0.169:5060, Kamailio get it on udp:10.130.0.23:5060 Kamailio Relay it to udp:10.130.0.23:5070 My pbx get requests from udp:10.130.0.23:5070 with via header 10.130.0.23 Responses to UserAgent go from udp:10.130.0.23:5060 with via 178.0.0.169 In RELAY block: if ($Ru == «udp:10.130.0.23:5070») { #set_send_socket(«udp:10.130.0.23:5060»); $fs = «udp:10.130.0.23:5060»; #force_send_socket(«udp:10.130.0.23:5060»); } else { #set_send_socket(«udp:10.130.0.23:5070»); $fs = «udp:10.130.0.23:5070»; } #$fs = «udp:10.130.0.23:5070»; if (!t_relay()) { sl_reply_error(); } route[REGISTRAR] { if ($rd == "mydomain.ru" || $rd == "mydomain2.ru") { set_send_socket("udp:10.130.0.23:5070"); } } It works for requests from UserAgent to PBX. But INVITE from my pbx to UserAgent goes only through udp:10.130.0.23:5070 , Kamailio realy it to internet from udp:10.130.0.23:5070 I get error in syslog: (pv [pv_branch.c:62]: pv_get_branchx_helper(): error accessing branch [0]) I tried set_send_socket(«udp:10.130.0.23:5070»); and $fs = «udp:10.130.0.23:5070»; Why I can`t relay outgoing requests to internet through another socket? -- Best regards, Alex

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[REGISTRAR] Registered. How to match $sut to usrloc received ?
by Richard Edmands 29 Jul '22

29 Jul '22

Hey Kamailio Users, So the scenario is we get the registers. Load them, slightly oddly as we also fill in the received section, not something I usually do. Then when they send an invite, the To and From headers are real phone numbers and for some reason the contact header isn't their username. So to compensate for that I'm trying to match their connection ($sut) using registered, but after a couple of hours I haven't been able to get it to match up. Not sure what I'm missing. ul.dump gets me Received: sip:sipclientpublicip:2353;transport=udp so we know that part is loading $sut correctly into the user location data. modparam("usrloc", "matching_mode", 2) modparam("registrar", "xavp_rcd", "ulrcd") modparam("registrar", "received_avp", "$avp(i:42)") $xavp(regcfg=>match_received) = $sut; if (registered("location","$sut", 2, 1)) { After that block of code $sut is sip: sipclientpublicip:2353;transport=udp which is what's in usrloc for received, but it's not match. There's probably a much smarter way of doing this that I've completely missed, if so then that'd be excellent. But if I'm on the right path somewhat then it would be great to get this working.

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[UAC] How to get the outbound registration status of uac.reg_active?
by Denys Pozniak 28 Jul '22

28 Jul '22

Hello! I have several Kamailio sharing a common uacreg table and use the RPC interface to enable and disable outgoing registration on nodes depending on the cluster status: *kamcmd uac.reg_active 0kamcmd uac.reg_active 1* https://kamailio.org/docs/modules/stable/modules/uac.html#uac.r.uac.reg_act… How can I get the current status of uac outbound registration (to get 0 or 1)? -- BR, Denys Pozniak

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Re: [SR-Users] TLS connection failure. Not sure if system behavior is correct.
by Ahmed Fouad 27 Jul '22

27 Jul '22

Hi, You are currently using TLSv1/SSLv3 ECDHE-RSA-AES256-GCM-SHA384 256 Teams stopped TLSv1/ TLSv1.1 and only allowing TLS1.2+ So change your TLS settings to enforce it only method = TLSv1.2+ after restarting kamailio your teams connections should be good.

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ACC CDR accounting
by Oleg Podguyko 27 Jul '22

27 Jul '22

HI I’m trying to get CDR function. I would like to have one CDR for whole call at the syslog. I use: …… loadmodule "db_text.so" loadmodule "dialog.so" loadmodule "acc.so" …….. # -----Dialog module ----- modparam("dialog", "db_mode", 0) # -----ACCounting module ----- modparam("acc", "early_media", 0) modparam("acc", "report_cancels", 1) modparam("acc", "log_facility", "LOG_LOCAL1") modparam("acc", "detect_direction", 0) modparam("acc", "log_level", ACC_LOG_LEVEL) modparam("acc", "log_flag", FLAG_ACC_LOG) modparam("acc", "log_missed_flag", FLAG_ACC_LOG_MISSED) modparam("acc", "failed_transaction_flag", FLAG_ACC_LOG_FAILED) modparam("acc", "log_extra", "ip_src=$si;ip_dst=$avp(ip_dst);sip_from=$fU;sip_to=$tU;ruri_user=$rU;spx_rs=$avp(spx_http_rs)") ## TODO modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_enable_log", 1) modparam("acc", "cdr_facility", "LOG_LOCAL1") request_route { if (is_method("INVITE")) { setflag(FLAG_ACC_LOG); setflag(FLAG_ACC_LOG_MISSED); setflag(FLAG_ACC_LOG_FAILED); } At the kamailio.log I see only transaction for INVITE, but I wait for some CDR for all dialog. Whats wrong in my config? -- -- Олег Подгуйко

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Cannot load usrloc table at startup using DMQ_USRLOC sync feature
by Benedito Marques 25 Jul '22

25 Jul '22

I have an scenario with two kamailios, and no databases. At the first, I authenticate one subscriber successfully and save in memory usrloc: /etc/kamailio # kamailio --version version: kamailio 5.4.4 (x86_64/linux) 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: unknown compiled on 16:13:18 Feb 15 2021 with gcc 9.3.0 /etc/kamailio # /etc/kamailio # --- /etc/kamailio # cat /etc/os-release NAME="Alpine Linux" ID=alpine VERSION_ID=3.12.3 PRETTY_NAME="Alpine Linux v3.12" HOME_URL="https://alpinelinux.org/" BUG_REPORT_URL="https://bugs.alpinelinux.org/" --- # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location","0x05")) { sl_reply_error(); } exit; } -- /etc/kamailio # kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { Info: { AoR: 1014(a)my.domain HashID: -1989102610 Contacts: { Contact: { Address: sip:1014@10.0.0.2:7071;ob Expires: 299 Q: -1.000000 Call-ID: d5cf6d5f293647a59ba64ba996a793b0 CSeq: 43689 User-Agent: MicroSIP/3.20.7 Received: sip: 10.0.0.2:7071 Path: [not set] State: CS_NEW Flags: 1 CFlags: 0 Socket: [not set] Methods: 8159 Ruid: uloc-62d95ca2-3c-1 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1658412252 KA-Roundtrip: 0 Last-Modified: 1658412252 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } /etc/kamailio # *The problem is*: when I start the second kamailio server, the current usrloc users are not replicated to it, so in the second server the usrloc is empty: /etc/kamailio # kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 1024 AoRs: { } Stats: { Records: 0 Max-Slots: 0 } } } } *The modules loaded:* - dmq.so - dmq_usrloc.so *Params (sync enabled):* modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "usrloc_domain", "location") modparam("dmq_usrloc", "usrloc_delete", 1) modparam("dmq_usrloc", "replicate_socket_info", 0) *Obs*: The DMQ replication is working perfectly. So, if I register another user in server 1, server 2 receives the KDMQ message and the new user appears on usrloc. The real problem is to load pre registered users at startup. *Log messages from second server at startup:* 0(51) DEBUG: <core> [core/sr_module.c:873]: init_mod(): dmq_usrloc 0(51) INFO: dmq_usrloc [dmq_usrloc.c:81]: mod_init(): dmq usrloc replication mode = 1 0(51) DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <ul_bind_usrloc> in module usrloc [/usr/lib/kamailio/modules/usrloc.so] 0(51) DEBUG: <core> [core/sr_module.c:652]: find_mod_export_record(): found export of <bind_dmq> in module dmq [/usr/lib/kamailio/modules/dmq.so] 0(51) DEBUG: dmq_usrloc [usrloc_sync.c:286]: usrloc_dmq_initialize(): loaded dmq api 0(51) DEBUG: dmq_usrloc [usrloc_sync.c:299]: usrloc_dmq_initialize(): dmq peer registered 0(51) DEBUG: dmq_usrloc [dmq_usrloc.c:111]: mod_init(): dmq_usrloc initialized 0(51) DEBUG: <core> [core/sr_module.c:804]: init_mod_child(): idx 0 rank -127: dmq_usrloc [main] 0(51) DEBUG: dmq_usrloc [dmq_usrloc.c:127]: child_init(): child_init PROC_INIT 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:552]: usrloc_dmq_request_sync(): requesting sync from dmq peers 0(51) DEBUG: <core> [core/sr_module.c:804]: init_mod_child(): idx 0 rank 0: dialog [main] 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:568]: usrloc_dmq_request_sync(): sending serialized data {"action":3} 7(62) DEBUG: dmq_usrloc [usrloc_sync.c:317]: usrloc_dmq_send(): sending dmq broadcast... 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:10.0.2.38:5090 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:10.0.2.30:5090 7(62) DEBUG: dmq [dmq_funcs.c:166]: bcast_dmq_message1(): skipping node sip:0.0.0.0:5090 *List nodes at SERVER 1:* /etc/kamailio # kamcmd dmq.list_nodes { host: 10.0.1.36 port: 5090 resolved_ip: 10.0.1.36 status: active last_notification: 0 local: 0 } { host: 10.0.1.35 port: 5090 resolved_ip: 10.0.1.35 status: active last_notification: 0 local: 0 } { host: 0.0.0.0 port: 5090 resolved_ip: 0.0.0.0 status: active last_notification: 0 local: 1 } *List nodes at SERVER 2:* /etc/kamailio # kamcmd dmq.list_nodes { host: 10.0.1.35 port: 5090 resolved_ip: 10.0.1.35 status: active last_notification: 0 local: 0 } { host: 10.0.1.36 port: 5090 resolved_ip: 10.0.1.36 status: active last_notification: 0 local: 0 } { host: 0.0.0.0 port: 5090 resolved_ip: 0.0.0.0 status: active last_notification: 0 local: 1 } -- Att, Benedito Marques

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Job oportunitie
by Jaime Gajardo 24 Jul '22

24 Jul '22

Hello, We are seeking Engineers with some experience in open source VoIP solutions like Kamilio, Asterisk, etc. The opportunity is remote in Latin america. We are a Chilean company dedicated to giving different solutions for customer service on cloud. is you are interested please send us an email to alejandro(a)elipse.ai Thank you!!

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db.c can't find dp_postgressql
by Phillip Thern 23 Jul '22

23 Jul '22

so, I don't actually know if my previous message found you, or if you replied, since my syslog suddenly grew to 26 GB and my server couldn't receive any mail because of it... Description I loaded module "db_postgres.so". I expected kamilio to start. It didn't Troubleshooting Reproduction config file |#!KAMAILIO #!define WITH_POSTGRESSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_MSGREBUILD #!define WITH_NAT #!define WITH_NATSIPPING . . . #!ifdef WITH_POSTGRESSQL loadmodule "db_postgres.so" #!endif | Log Messages |Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [db.c:204]: db_bind_mod(): Module db_postgresql not found. Missing loadmodule? Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: usrloc [usrloc_mod.c:361]: mod_init(): failed to bind database module Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) | output from ls of modules directory: |acc.so call_obj.so dialog.so ipops.so msrp.so presence_conference.so qos.so sipcapture.so statsd.so uid_avp_db.so acc_diameter.so carrierroute.so dialplan.so jsonrpcs.so mtree.so presence_dialoginfo.so ratelimit.so sipdump.so stun.so uid_domain.so alias_db.so cfg_db.so dispatcher.so keepalive.so nat_traversal.so presence_mwi.so regex.so sipjson.so tcpops.so uid_gflags.so app_jsdt.so cfg_rpc.so diversion.so kemix.so nathelper.so presence_profile.so registrar.so sipt.so textops.so uid_uri_db.so app_sqlang.so cfgt.so dmq.so kex.so nosip.so presence_reginfo.so rls.so siptrace.so textopsx.so uri_db.so async.so cfgutils.so dmq_usrloc.so lcr.so outbound.so presence_xml.so rr.so siputils.so timer.so userblocklist.so auth.so corex.so domain.so log_custom.so p_usrloc.so pua.so rtimer.so sl.so tm.so usrloc.so auth_db.so counters.so domainpolicy.so lrkproxy.so path.so pua_bla.so rtjson.so sms.so tmrec.so xcap_client.so auth_diameter.so ctl.so drouting.so mangler.so pdb.so pua_dialoginfo.so rtpengine.so smsops.so tmx.so xcap_server.so auth_xkeys.so db2_ops.so enum.so matrix.so pdt.so pua_reginfo.so rtpproxy.so speeddial.so topoh.so xhttp.so avp.so db_cluster.so evrexec.so maxfwd.so permissions.so pua_rpc.so sanity.so sqlops.so topos.so xhttp_prom.so avpops.so db_flatstore.so exec.so mediaproxy.so pike.so pua_usrloc.so sca.so ss7ops.so tsilo.so xhttp_rpc.so benchmark.so db_postgres.so group.so mohqueue.so pipelimit.so pua_xmpp.so sdpops.so sst.so uac.so xlog.so blst.so db_text.so htable.so mqueue.so prefix_route.so pv.so seas.so statistics.so uac_redirect.so xprint.so call_control.so debugger.so imc.so msilo.so presence.so pv_headers.so secfilter.so statsc.so uid_auth_db.so | Possible Solutions tried renaming db_postgres.so to db_postgresql.so, but that didn't work, so I reversed it Additional Information * *Kamailio Version* - output of |kamailio -v| |version: kamailio 5.5.4 (x86_64/linux) 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_BLOCKLIST, 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: unknown compiled with gcc 10.2.1 | * *Operating System*: |Distributor ID: Debian Description: Debian GNU/Linux 11 (bullseye) Release: 11 Codename: bullseye Linux opensips 5.10.0-16-amd64 #1 SMP Debian 5.10.127-1 (2022-06-30) x86_64 GNU/Linux |

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uac replace PRACK trouble
by Алексей Якимкин 22 Jul '22

22 Jul '22

Hello, pbx -> kamailio -> provider kamailio registers (uacreg) on provider. When I call through my provider I need to change headers To From and PAI. I tried with uac_replace_from and uac_replace_to. $tU for example 0987656789 if($fU=~"1234567") { uac_replace_from("sip:provider_login@provider.com"); uac_replace_to("sip:$tU@provider.com"); } My provider answers on Invite with 183 progress, so pbx answers with PRACK. Kamailio change To and from like this: To: <sip:0987656789@provider.comsip:0987656789@provider.com>;tag=..... To: <sip:0987656789@provider.comsip:0987656789@provider.com> From: <sip:provider_login@provider.comsip:provider_login@provider.com >;tag=..... From: <sip:provider_login@provider.comsip:provider_login@provider.com> Is it a bug? I tried to disable this modification for PRACK u REQINIT ($csb != "2 PRACK") It haven't worked for me. I solved this issue with this code. remove_hf("From"); if ($ft == $null) { insert_hf("From: <sip:provider_login@provider.com>\r\n", "To"); } else { insert_hf("From: <sip:provider_login@provider.com>;tag=$ft\r\n", "To"); } version: kamailio 5.7.0-dev0 (x86_64/linux) Ubuntu 18 -- Regards, Alex

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Module db_postgresql not found
by Phillip Thern 22 Jul '22

22 Jul '22

Description I loaded module "db_postgres.so". I expected kamilio to start. It didn't Troubleshooting Reproduction config file |#!KAMAILIO #!define WITH_POSTGRESSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_MSGREBUILD #!define WITH_NAT #!define WITH_NATSIPPING . . . #!ifdef WITH_POSTGRESSQL loadmodule "db_postgres.so" #!endif | Log Messages |Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [db.c:204]: db_bind_mod(): Module db_postgresql not found. Missing loadmodule? Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: usrloc [usrloc_mod.c:361]: mod_init(): failed to bind database module Jul 22 10:31:16 opensips /usr/sbin/kamailio[8975]: ERROR: <core> [core/sr_module.c:942]: init_mod(): Error while initializing module usrloc (/usr/lib/x86_64-linux-gnu/kamailio/modules/usrloc.so) | output from ls of modules directory: |acc.so call_obj.so dialog.so ipops.so msrp.so presence_conference.so qos.so sipcapture.so statsd.so uid_avp_db.so acc_diameter.so carrierroute.so dialplan.so jsonrpcs.so mtree.so presence_dialoginfo.so ratelimit.so sipdump.so stun.so uid_domain.so alias_db.so cfg_db.so dispatcher.so keepalive.so nat_traversal.so presence_mwi.so regex.so sipjson.so tcpops.so uid_gflags.so app_jsdt.so cfg_rpc.so diversion.so kemix.so nathelper.so presence_profile.so registrar.so sipt.so textops.so uid_uri_db.so app_sqlang.so cfgt.so dmq.so kex.so nosip.so presence_reginfo.so rls.so siptrace.so textopsx.so uri_db.so async.so cfgutils.so dmq_usrloc.so lcr.so outbound.so presence_xml.so rr.so siputils.so timer.so userblocklist.so auth.so corex.so domain.so log_custom.so p_usrloc.so pua.so rtimer.so sl.so tm.so usrloc.so auth_db.so counters.so domainpolicy.so lrkproxy.so path.so pua_bla.so rtjson.so sms.so tmrec.so xcap_client.so auth_diameter.so ctl.so drouting.so mangler.so pdb.so pua_dialoginfo.so rtpengine.so smsops.so tmx.so xcap_server.so auth_xkeys.so db2_ops.so enum.so matrix.so pdt.so pua_reginfo.so rtpproxy.so speeddial.so topoh.so xhttp.so avp.so db_cluster.so evrexec.so maxfwd.so permissions.so pua_rpc.so sanity.so sqlops.so topos.so xhttp_prom.so avpops.so db_flatstore.so exec.so mediaproxy.so pike.so pua_usrloc.so sca.so ss7ops.so tsilo.so xhttp_rpc.so benchmark.so db_postgres.so group.so mohqueue.so pipelimit.so pua_xmpp.so sdpops.so sst.so uac.so xlog.so blst.so db_text.so htable.so mqueue.so prefix_route.so pv.so seas.so statistics.so uac_redirect.so xprint.so call_control.so debugger.so imc.so msilo.so presence.so pv_headers.so secfilter.so statsc.so uid_auth_db.so | Possible Solutions tried renaming db_postgres.so to db_postgresql.so, but that didn't work, so I reversed it Additional Information * *Kamailio Version* - output of |kamailio -v| |version: kamailio 5.5.4 (x86_64/linux) 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_BLOCKLIST, 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: unknown compiled with gcc 10.2.1 | * *Operating System*: |Distributor ID: Debian Description: Debian GNU/Linux 11 (bullseye) Release: 11 Codename: bullseye Linux opensips 5.10.0-16-amd64 #1 SMP Debian 5.10.127-1 (2022-06-30) x86_64 GNU/Linux |

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incomplete values and Wrong Counting for $hfl and $hflc
by Patrick Karton 21 Jul '22

21 Jul '22

the counting and values provided by $hfl and $hflc are wrong and incomplete from my understanding. looking at the source code these pseudo vars count the body values of Via, Record-Route and Route headers. in kamalio 5.6.1 with Record-Route: <sip:1.1.1.1;lr=on;nat=yes> Record-Route: <sip:2.2.2.2:5060;lr;transport=udp>,<sip:3.3.3.3:22506;lr;transport=udp> i get $hflc(Record-Route) == 2 instead of 3 and $(hfl(Record-Route)[0]) == 1.1.1.1 $(hfl(Record-Route)[1]) == 2.2.2.2 $(hfl(Record-Route)[3]) does not exists.

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Using Kamailio with Teams
by Javier González Muñoz 21 Jul '22

21 Jul '22

Hi, I'm using kamailio with Teams like SBC and asterisk. Works perfectly less one thing. When I make a call asterisk to Teams, in my Teams app I see 3 popup notifications, and 3 calls in my history. If I call Teams to Asterisk, perfect.. When I make an asterisk to Teams call, I get the trace from Asterisk and I can see three 180, I don't know if this is the problem, but are there any way to fix it? Limit to 1 notification? Thanks

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iterate trough all Record-Route values via @hf_value.Record_Route
by Patrick Karton 21 Jul '22

21 Jul '22

Hello All, i would to iterate through all ips in Record-Route through @hf_value.Record_Route. if i put static index like @hf_value.Record_Route[0] it works. but $var(i) = 0; @hf_value.Record_Route[$var(i)] does not work. is there a way to get the count of all headers values through @hf_value.Record_Route For example i would like to iterate through all Record-Route Values if i receive it like this Record-Route: <sip:1.1.1.1;lr=on;nat=yes> Record-Route: <sip:2.2.2.2:5060;lr>,<sip:3.3.3.3:22506;lr>

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siremis installation
by Fernando Piana 19 Jul '22

19 Jul '22

hello, My name is Fernando Piana from Colombia. I am writing to you because I am testing Kamailio together with the Siremis graphical environment. I was visiting the siremis.org website and also the siremis wiki, but I couldn't find the administration manual. >From this medium I want to ask you if you can send me a copy of the Siremis administration manual. I await a response. Thank you very much. -- Fernando Piana Ing. Comunicaciones y Bases de Datos Cel: +57 311 641 7558

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Using usrloc db_insert_update in a dmq replicated environment with shared db ?
by Matteo Brancaleoni 19 Jul '22

19 Jul '22

Hi, I was wondering if it is safe from a cache consistency pow to enable usrloc db_insert_update in a dmq replicated environment, where all registrars share the same db. I'm already using the filter option to avoid loading/deletion by server id, but there's no similar option for writing. I know that it is possible to use db mode 4 on all registrars (which get synced by dmq) except one, which will be responsible for writing, but this creates a little unbalanced setup in addition to performing writes only from one proxy. Any hint? Maybe is time to check how to filter writes, too :) ? Thanks, Matteo

1 0

How to externally trigger htable cache flushing?
by Benoit Panizzon 18 Jul '22

18 Jul '22

Hi List We cache stuff like call forwarding information in a htable cache to minimizie SQL lookups and have it available via DMQ on all instances. But of course, if the call forwarding information for a destination changes in the database, we would like to flush either the whole cache, or even better, that single entry. What is the best way to do this? 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 3

dialog timeout_avp no longer works?
by Alex Balashov 18 Jul '22

18 Jul '22

Hi, I discovered recently that, on Kamailio 5.5.4 at least, the dialog ’timeout_avp’ no longer appears to have effect. Here are my (seemingly relevant) dialog parameters: modparam("dialog", "enable_stats", 0) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "dlgcor") # DLG "correlator" modparam("dialog", "dlg_flag", 2) modparam("dialog", "default_timeout", 28800) # 8 hours modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "table_name", "dialog") modparam("dialog", "send_bye", 1) modparam("dialog", "wait_ack", 0) modparam(“dialog”, “timeout_avp”, “$avp(dlg_timeout)”) The flow is roughly like this: request_route { ... t_check_trans(); if(is_method(“INVITE”)) { ... dlg_manage(); $avp(dlg_timeout) = 10; ... if(!t_relay()) sl_reply_error(); return; } I was able to resolve the problem by switching to dlg_set_timeout() instead, but I wonder if there has been a regression. Last version I have on which this positively works is 4.4.7, though, so not sure when it might have broken. One nuance: $avp(dlg_timeout) is not initialised prior to calling dlg_manage(), and the value is only set if certain conditions are true. Not sure if some stricter check appeared in later releases which could cause this to be a problem. Thank you! — Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Nathelper sip ping and Path support ?
by matteo brancaleoni 18 Jul '22

18 Jul '22

Hello, I'm trying to setup the following scenario with sip ping: UAC -> Kamailio Edge -> Kamailio Registrar Edge proxy is a multihomed proxy with public/private ips and the registrar is a kamailio on a private network. The edge acts as a nat proxy and loadbalancer. Edge proxy inserts a Path header (and contact alias) which get correctly saved on registrar usrloc. Registration works as expected. Now, since I need to keep nat open towards the uacs and expire them when they're not reachable anymore, I'm trying to send sip pings from the registrar via the edge proxy. I'm using nathelper sipping in the registrar. What I see is that the registrar sends out a sip options putting in the Route header the Path as expected, but the sip packet gets sent directly to the UAC contact and not via the edge, as the Route header specifies. After looking into the nathelper code, it seems that the sip packet is sent via udp directly without any specific routing. So the question: it should work what I'm trying to achieve? Maybe not? Probably I can work around it setting the ;received in the contact on registrar (it will set to the edge proxy) and since the contact is aliased on the edge when the edge receives an option from the registrar it can process the alias and send it to the correct endpoint, but seems a workaround? Or using the keepalive module maybe, even if that means setting specific timers and manually do what the nathelper module already does? Any hint? Thanks, Matteo

3 2

Does pure outbound RFC5626 work without edge proxies?
by Richard Chan 17 Jul '22

17 Jul '22

Hi sr-users, Does pure RFC5626 outbound work without an edge proxy->internal proxy design? By "pure" I mean no nathelper to rewrite addresses. I tried setting up the following: Configuration: use standard kamailio configuration without WITH_NAT (Ignore RTP-rtpengine/rtpproxy for the moment as I am just testing the control plane) 1. UA1 UA2 each behind NAT, with bogus Contact: headers 2. kamailio loaded with rr/tm (no outbound/path because I am using single proxy/registrar — without edge proxies) 3. no nathelper: no Contact rewriting, set_contact_alias etc 4. Both UA1 and UA2 establish persistent TLS to kamailio 5. INVITE/200 OK/ACK does work. 6. Contact headers show the NAT bogus addresses (as expected). UA1/UA2 are configured to be "dumb"(i.e. don't self-rewrite Contact:/SDP based on rport/received). Problem: 7. In-dialog requests don't work as kamailio tries to route to the bogus addresses per the Contact header. I.e. kamailio doesn't seem to resolve the UAs(bogus Contact addresses) to the persistent TLS connections. Now in the Route header I don't see any flow-token so probably kamailio is doing "normal" routing instead of RFC5626 routing (i.e. because it has no flow-token to match with an existing flow). This leads me to ask: for pure RFC5626 to work (no nathelper stuff at all) is it a must that there is a "edge-proxy--[1]-->internal-proxy" layout? ...and...the flow-tokens in [1] are what make it possible to skip bogus Contact: header and route to an existing flow? Regards Shih-Ping Chan

2 1

KEMI: getting ksr_request_route not found or is not callable after t_relay()
by Nigel Daniels 16 Jul '22

16 Jul '22

``` 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(): KSR.info("try and relay the traffic\n") if KSR.tm.t_is_set("ksr_failure_route") < 0: KSR.tm.t_on_failure("ksr_failure_manage") t_relay_result = KSR.tm.t_relay(["10.0.0.6", "5060"]) KSR.info(f't_relay: {t_relay_result}\n') KSR.info(f'MSG: {msg}') if KSR.tm.t_is_set("ksr_failure_route_one"): KSR.tm.t_on_reply("ksr_on_reply_route_one") ``` ERROR: app_python3 [python_exec.c:91]: apy_exec(): ksr_request_route not found or is not callable Again this only happens after t_replay is called. Any help understanding why this happens would be appreciated. -- - Nigel

2 1

uacreg get error on 2nd 401: error uac_reg_tm_callback(): authentication failed
by Алексей Якимкин 15 Jul '22

15 Jul '22

Hello, I try to register on provider`s pbx version: kamailio 5.7.0-dev0 (x86_64/linux) modparam("uac", "reg_db_url", "mysql://kamailio:123@localhost/kamailio") modparam("uac", "reg_contact_addr", "1.1.1.1:5060") modparam("uac", "reg_retry_interval", 120); # for failed registrations modparam("uac", "reg_keep_callid", 1) modparam("uac","restore_mode","auto") uacreg table: +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | contact_addr | socket | +----+---------+------------+-----------+----------------+-----------------+-------+----------------+------------------+----------+---------------------+---------+-------+-----------+--------------+------------------+ | 1 | 1234567 | pbx | 127.0.0.1 | login | ip.provider.ru | | login | password| | sip:...com | 120 | 0 | 0 | | 10.130.0.23:5060 | I see this two 401 replies [image: изображение.png] 1st REGISTER Via: SIP/2.0/UDP 1.1.1.1:5060 ;branch=z9hG4bK589d.8183f2e6000000000000000000000000.0 To: <sip:login@ip.provider.com> From: <sip:login@ip.provider.com >;tag=f9cf84029bbf24b61836c3d9fef7b7ae-78a2ffe1 CSeq: 10 REGISTER Call-ID: 6818f1b436befce1-21980(a)10.130.0.23 Max-Forwards: 70 Content-Length: 0 User-Agent: server Contact: <sip:1234567@1.1.1.1:5060> Expires: 120 1st 401 WWW-Authenticate: Digest algorithm=MD5,realm=".......org",nonce=" *56806a6f32c49a3afea043c2fa42381f*",domain="sip:......org",qop="auth", *stale=FALSE* 2nd REGISTER Authorization: Digest username="login", realm=".....", nonce=" *56806a6f32c49a3afea043c2fa42381f*", uri="sip:......com", qop=auth, nc=00000001, cnonce="357420659", response="6e300c9f7f4d81095f29b5408f8d5191", algorithm=MD5 2nd 401 WWW-Authenticate: Digest algorithm=MD5,realm=".....org",nonce=" *3b9586f6d31a76e509eb204d8a030415*",domain="sip:.....org",qop="auth", *stale=TRUE* I think kamailio need to sent 3nd REGISTER with a new nonce, because stale=true. But it interrupts register with error: Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) DEBUG: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:852]: uac_reg_tm_callback(): completed with status 401 [uuid: 1234567] Jul 14 18:41:00 opensips kamailio[21936]: 10(21949) ERROR: {2 11 REGISTER 6818f1b436befcf7-21980(a)10.130.0.23} uac [uac_reg.c:933]: uac_reg_tm_callback(): authentication failed for <1234567> Can I get kamailio to send REGISTER after the second 401-reply? Can I intercept this REGISTER and 401 in kamailio.conf? Is there any examples? I know about this instruction https://kamailio.org/docs/modules/5.7.x/modules/uac.html :) Itried in request_route xlog("L_NOTICE", "uac 114411 1 remote fU $fU user [$rU] on [$rd] via [$du] $Ri \n"); if (uac_reg_request_to("$fU", 0)) { xlog("L_NOTICE", "uac 114411 Found remote user [$rU] on [$rd] via [$du]\n"); t_on_failure("REMOTE_AUTH"); #t_relay(); } And failure_route failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_NOTICE", "uac 114411 Remote asked for authentication\n"); uac_auth(); } } but I haven't got logs on register request, 401 reply. -- Best regards, Alex

1 1

usage of {s.urldecode.param} to convert special chars (why is + removed?)
by Benoit Panizzon 14 Jul '22

14 Jul '22

Hi List Service Numbers, like to activate CFW probably often start with * and end with a # # is a special character that needs to be urlencoded in an SIP URI. So best would be to use a generic kamailio urldecode transormation. I guess {s.urldecode.param} is what I was looking for. So when I get a call to the 'enable CFW' service I get $rU => "*21+419999999%23" So I want to extract the destination number and remove the urlencoded # sign. So I do $var(cfw-dest) = $var(cfw-dest) = $rU; $var(cfw-dest) = $(var(cfw-dest){s.substr,3,0}); # "+419999999%23" $var(cfw-dest) = $(var(cfw-dest){s.urldecode.param}); But now I end up with 419999999# the + sigh disappeared. Is this expected behavior? How would this be done the correct way? 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 ______________________________________________________

2 1

Releasing v5.6.1
by Daniel-Constantin Mierla 14 Jul '22

14 Jul '22

Hello, I am considering to release Kamailio v5.6.1 (out of branch 5.6) on Wednesday, July 6, 2022 -- in about 1 week and a half from now. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online: June 20-23, 2022 * https://www.asipto.com/sw/kamailio-advanced-training-online/

5 6

Need Suggestion/Idea to set call limit on 3 kamailios.
by Waqar 40 13 Jul '22

13 Jul '22

Please Read the Scenario in full and then guide me. In my scenario, I have three Kamailio (KamA, KamB, KamC) running in production. They do load balancing for the asterisks which are about 9. Kamailios randomly receive calls from the SIP trunk. I have set the limit of 750 calls on each kamailios. I want to provide only 1000 Calls to my client but they are enjoying calls more than this since each kamailio can receive 750 calls (I have set this limit for redundancy purpose). And I cannot set the limit like 750 on KamA and the rest on KamB and KamC for redundancy purposes. If one kamailio goes down then they send calls on the other kamailios so that no call drops. So, I want to know if there is a module or a way that my redundancy is not affected and I limit total calls on the three kamailios to not exceed 1000. Waiting for a prompt response. Let me know if you need any further information. Regards, Vicky

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