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37517 discussions

(Off-topic) Vienna, meeting
by Henning Westerholt 09 Jul '22

09 Jul '22

(Off-topic) Hello, In about two weeks I will be a few days in Vienna (12h - 14th July). If you want to meet for a drink to chat about Kamailio and other topics, just contact me off-list. Thanks, and regards, Henning

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Client behind nat without stun and kamailio behind NAT
by Алексей Якимкин 08 Jul '22

08 Jul '22

Hello, I hope somebody could help me. This is my scheme. User-agent is behind NAT1. Kamailio and pbx are behind NAT2 [Client ip-phone 192.168.89.213 without stun] - LAN1 - NAT1(46.0.0.30) - (internet) - (51.0.0.60)NAT2 - Local2 - (10.130.0.23:5060)kamailio( 10.130.0.23:5070) - pbx *Questions*: *1*. The VIA header (with 10.130.0.23 and 51.0.0.60) wasn't included in SIP packets. Why? For example 200OK reply. It came from pbx through kamailio. Which setting could break it? Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK2480172053 Record-Route: <sip:c172.19.19.111.8348.call.cgatepro;lr> Record-Route: <sip:172.19.19.111:5060;lr> Record-Route: <sip:10.130.0.23:5070 ;r2=on;lr;ftag=4268683942;nat=yes>,<sip:51.0.0.60;r2=on;lr;ftag=4268683942;nat=yes> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=4268683942 To: <sip:2961@mail.domain.ru:5060 >;tag=A03E2397-404246-FA7543E4_jizmelr-582D Call-ID: 6_1903330087(a)192.168.89.213 [Generated Call-ID: 6_1903330087(a)192.168.89.213] CSeq: 2 INVITE Contact: <sip:signode-404246-FA7543E4_jizmelr-582D@172.19.19.111 ;alias=51.0.0.60~5060~1> Supported: 100rel,timer,replaces,histinfo,precondition Allow: INVITE,BYE,CANCEL,ACK,OPTIONS,INFO,MESSAGE,PRACK,UPDATE,REFER Session-Expires: 1800;refresher=uas Content-Type: application/sdp Content-Length: 1170 Message Body *2*. About Registrar, Path and $du. Phone set Register with headers: Via: SIP/2.0/UDP 192.168.89.213:5060;branch=z9hG4bK2691182696 From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060> Contact: <sip:a.yakimkin@192.168.89.213:5060> Kamailio respond Via: SIP/2.0/UDP 192.168.89.213:5060 ;rport=9570;received=46.0.0.30;branch=z9hG4bK617463686 Path: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on>,<sip:51.0.0.60;lr;received=46.0.0.30~9570~1;r2=on> From: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=3926879477 To: "Aleksey" <sip:a.yakimkin@mail.domain.ru:5060>;tag=194ED16D Contact: <sip:a.yakimkin@192.168.89.213:5060>;expires=360 Contact: <sip:2447@192.168.89.221:5060>;expires=247 Contact: <sip:2447@192.168.9.16:5060>;expires=2116 I try to make a call from pbx to ip phone. pbx inserts in Invite header Route: <sip:10.130.0.23:5070 ;lr;received=46.0.0.30~9570~1;r2=on> But kamailio relayed Invite direct to 192.168.89.213. (There is network connectivity between ip-phone and kamailio through vpn). The code below helps me to solve my issue. I saw mail-list with similar trouble. But no setting could get kamailio to relay Invite to "Route-received" ip. $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } *Kamailio settings:* I have such listeners listen=udp:10.130.0.23:5070 # to local network listen=udp:10.130.0.23:5060 advertise 51.0.0.60:5060 # to internet #MODULE SETTING #--- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") modparam("path", "use_received", 1) modparam("path", "enable_r2", 1) modparam("path", "received_format", 1) # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "auto_inv_100_reason", "Trying") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 2) modparam("rr", "force_send_socket", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) modparam("registrar", "use_path", 1) modparam("registrar", "path_use_received", 1) modparam("registrar", "path_mode", 0) #--- For register I use this code route[REGISTRAR] { ... add_path_received(); set_send_socket("udp:10.130.0.23:5070"); route(DISPATCH); ... } route[RELAY] { ... if ($Ru eq "sip:10.130.0.23:5070") { $fs = "udp:10.130.0.23:5060"; } else { $fs = "udp:10.130.0.23:5070"; } ... } 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; #send INVITE to IP:PROT from Route:...;received= $var(the_route) = $hdr(Route); $var(route0) = $(var(the_route){s.select,0,,}); $var(new_host) = $(var(route0){param.value,received}{s.select,0,~}); $var(new_port) = $(var(route0){param.value,received}{s.select,1,~}); if (!strempty($var(new_host)) && !strempty($var(new_port)) ) { $du = "sip:" + $var(new_host) + ":" + $var(new_port); } if (client_nat_test("3")) { if(nat_uac_test("18")) { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("co", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("co"); } if (is_method("REGISTER")) { #if ($Ru == "sip:10.130.0.23:5070") { # fix_nated_contact(); #} else { # set_contact_alias(); #} set_contact_alias(); } else { if(is_first_hop()) { set_contact_alias(); } else { add_contact_alias("51.0.0.60", "5060", "udp"); #fix_nated_contact(); } } } else { if ($Ru == "sip:10.130.0.23:5070") { rtpproxy_manage("cor", "51.0.0.60"); # fix_nated_sdp } else { rtpproxy_manage("cor"); } } } else { rtpproxy_manage("co"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); #fix_contact(); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); #} else { #fix_contact(); #} } } #!endif return; } Thank you. -- Best regards, Alex

2 1

log any rpc command
by Patrick Karton 08 Jul '22

08 Jul '22

hello is there a way to log in kamailio any kamcmd entered ? Thanks.

1 0

Multiple rtpengine servers
by David Cunningham 07 Jul '22

07 Jul '22

Hello, We are looking to implement a system with one Kamailio server and multiple rtpengine servers. I see that the rtpengine module allows the server configuration to be stored in a database table with multiple entries, which is perfect. My question is - how does Kamailio distribute load between multiple rtpengine servers in the same set? If one of the rtpengine servers goes offline then will Kamailio note this and not send calls to that server until it comes back online? If not then presumably calls could be delayed by whatever the timeout for accessing a server is? Thank you very much, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

3 3

kamailio 5.6.1 minor release
by Patrick Karton 06 Jul '22

06 Jul '22

Hello Daniel, you said a couple weeks ago that you plan to release a minor release 5.6.1 for this week. its still on plan ? Thanks.

2 2

Kamailio v5.6.1 Released
by Daniel-Constantin Mierla 06 Jul '22

06 Jul '22

Hello, Kamailio SIP Server v5.6.1 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.1. For more details about version 5.6.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/07/kamailio-v5-6-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Adding a short note that the dates of next Kamailio World Conference - Online have been set for September 7-8, 2022. Event website and more details will be published soon! Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

branch_route not called on DNS failover
by Patrick Karton 06 Jul '22

06 Jul '22

im using dns failover with tm module. i set parameter reparse_on_dns_failover to 0 so that branch_route is called . but when trying next peer of dns failover initial transaction branch_route is not called. do i missing something ? for the first peer branch_route is called and the peer timeout so kamailio tries next peer but branch_route is not called !. do i need again to set branch_route in failure_route ? Thanks for help.

3 4

Changing the From User in Failure Route
by Mack Hendricks 06 Jul '22

06 Jul '22

Hey All, I'm just trying to change the From user before I do a 301 Redirect within a Failure Route. Here's the code snippet: $fu = "+15555555555"; append_to_reply("Contact: sip:+1$var(redirect_did)@$si:$sp\r\n"); t_reply("302", "Redirect"); The From user is not changed and msg_apply_changes() isn't allowed. What am I doing wrong? Thanks

2 1

Kamailio 5.5 STIR SHAKEN private key buffer size error
by Maharaja Azhagiah 05 Jul '22

05 Jul '22

Hi, I am trying STIR/SHAKEN using libstirshaken in Kamailio 5.5. I used a self signed certificate as this is just a test in the local docker environment. However, when I try to add identity with private key (stirshaken_add_identity_with_key), I get "[error_code: 447] Buffer for key from file /tmp/cert/private.pem too short (2000 <= 3247)" I have tried using 2048 and 4096 size root@5907e44bd056:/tmp/cert# openssl rsa -in private.pem -text -noout | grep "Private-Key" RSA Private-Key: (4096 bit, 2 primes) Could you tell me what is wrong with the certificate? Kamailio version: root@5907e44bd056:/usr/local/kamailio/etc/kamailio# kamailio -v version: kamailio 5.5.4 (x86_64/linux) 469465 Error: 0(404) ERROR: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:761]: ki_stirshaken_add_identity_with_key(): Failed to load private key 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:117]: stirshaken_print_error_details(): failure details: 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:118]: stirshaken_print_error_details(): failure reason is: src/stir_shaken_ssl.c:2112: [error_code: 447] Buffer for key from file /tmp/cert/private.pem too short (2000 <= 3247) 0(404) DEBUG: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} stirshaken [stirshaken_mod.c:119]: stirshaken_print_error_details(): failure error code is: 447 0(404) ERROR: {1 30587 INVITE NzIhM1-2YABveZZ1mPvs3m3tw8K7meSq} <script>: Failed Regards *Maharaja Azhagiah*

3 6

How to call RPC functions from script
by Marat Gareev 05 Jul '22

05 Jul '22

Hello! Is it possible to call RPC functions from script? In particular, I'm interested in the ability to add new groups to the dispatcher. https://kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatcher.…

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