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Development for v5.6.x is frozen
by Daniel-Constantin Mierla 05 May '22

05 May '22

Hello, the formal notification that the development for the next major version 5.6.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.6 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.6. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.6. Once the branch 5.6 is created, new features can be pushed again to master branch as usual. From that moment, the v5.6.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.6 is going to be built at: * https://www.kamailio.org/wiki/features/new-in-5.6.x Upgrade guidelines will be collected at: * https://www.kamailio.org/wiki/install/upgrade/5.5.x-to-5.6.0 Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.5.x to 5.6.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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unexpected error log when KSR.xhttp.xhttp_reply(404...)
by hanxiaofang 04 May '22

04 May '22

Hello Kamailio community kamailio v5.5.4 debian10 Lua Scripting Routing: listen=tcp:127.0.0.1:5060 tcp_accept_no_cl=yes loadmodule "xhttp.so" loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") cfgengine "lua" function ksr_xhttp_event() KSR.set_reply_no_connect() KSR.set_reply_close() KSR.xhttp.xhttp_reply(404, "Not Found", "text/plain", "Not Found") end curl http://127.0.0.1:5060 -vvv * Expire in 0 ms for 6 (transfer 0x555d93fd9fb0) * Trying 127.0.0.1... * TCP_NODELAY set * Expire in 200 ms for 4 (transfer 0x555d93fd9fb0) * Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0) > GET / HTTP/1.1 > Host: 127.0.0.1:5060 > User-Agent: curl/7.64.0 > Accept: */* > < HTTP/1.1 404 Not Found < Sia: SIP/2.0/TCP 127.0.0.1:54944 < Content-Type: text/plain < Server: kamailio (5.4.4 (x86_64/linux)) < Content-Length: 9 < * Connection #0 to host 127.0.0.1 left intact The output of curl is OK, but have some unexpected error in kamailio.log ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 16) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 404 Not Found#015#012Sia: SIP/2.0/TCP 127.0.0.1:54946#015#012Content-Type: text/plain#015#012Server: kamailio (5.4.4 (x86_64/linux))#015#012Content-Length: 9#015#012#015#012Not Found> ERROR: <core> [core/msg_translator.c:3235]: build_sip_msg_from_buf(): parsing failed now change to Native Scripting Routing: listen=tcp:127.0.0.1:5060 tcp_accept_no_cl=yes loadmodule "xhttp.so" event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); xhttp_reply("404", "Not Found", "text/plain", "Not Found"); exit; } curl http://127.0.0.1:5060 -vvv everything is ok. here is the Navite config file: http://220.231.180.131:12296/kam/native/ here is the Lua config file: http://220.231.180.131:12296/kam/kemi/ hanxiaofang

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sanity_reply not available from sanity module
by Patrick Karton 04 May '22

04 May '22

Hello , the sanity_reply function from sanity is not available in kamailio script so i get an error when i want to use it. in the module source i see this static cmd_export_t cmds[] = { {"sanity_check", (cmd_function)w_sanity_check, 0, 0, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_check, 1, fixup_igp_null, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_check, 2, fixup_igp_igp, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"sanity_check", (cmd_function)w_sanity_reply, 0, 0, 0, REQUEST_ROUTE|ONREPLY_ROUTE}, {"bind_sanity", (cmd_function)bind_sanity, 0, 0, 0, 0 }, {0, 0, 0, 0, 0, 0} }; why is sanity_check linked again in w_sanity_reply ? is this an error ? i want to setup specific responses for some errors and let the default ones from module work as usual. Thanks.

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FAILEDOVER ROUTING WITH DNS SRV Records
by Patrick Karton 04 May '22

04 May '22

Hello, im using kamaliio 5.5.3 and I want to do failedover routing with DNS SRV Records. in my lab i have these SRV records ;; ANSWER SECTION: voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-74.voip.oov-obs-tools-dev.svc.cluster.local. voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-75.voip.oov-obs-tools-dev.svc.cluster.local. voip.oov-obs-tools-dev.svc.cluster.local. 5 IN SRV 0 33 5060 100-68-7-76.voip.oov-obs-tools-dev.svc.cluster.local. ;; ADDITIONAL SECTION: 100-68-7-76.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.76 100-68-7-75.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.75 100-68-7-74.voip.oov-obs-tools-dev.svc.cluster.local. 5 IN A 100.68.7.74 my config script use_dns_cache=on use_dns_failover=on dns_srv_lb=on $du = "voip.oov-obs-tools-dev.svc.cluster.local"; t_relay(); after t_relay() is called the INVITE is send to one the 3 ips corresponding to fqdn but if the first ip does not send response. the two others ips are not tried. what i want is that all the ips resolved are tried until one succeeds. is there something to do to achieve that ? do i need to interact with failure route to try next ips ? Thanks for help. version: kamailio 5.5.3 (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

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function 'sl_send_reply' is not available
by David Villasmil 04 May '22

04 May '22

Hello guys, I'm starting to try KEMI and i already hit a snag... right now i'm simply testing with: loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/myscript.lua") cfgengine "lua" route { lua_run("send_reply"); } But i'm always getting: 0(118) ERROR: app_lua [app_lua_api.c:1633]: sr_kemi_lua_modf(): function 'sl_send_reply' is not available I'm guessing that function is not exported by the module.. if that's the case, how can i use it? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Dispatcher SRV load balancing
by Nicolas Chapleau 03 May '22

03 May '22

Just having a hard time trying to see if the Dispatcher module does LB when SRV records are used as the destination. Will the destinations be picked based on the SRV list or will it simply use the first name received? Read lots of docs, but can't seem to figure it out Thanks, Nic.

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Problem with Record-Route in Public to Private Network
by Carlos Escalona 03 May '22

03 May '22

Hello! I wish everyone who reads this message a good day. I would appreciate some guidance with the configuration of my Kamailio. Initially, I have the following architecture: Endpoints <----Public IP----> Kamailio (listen PrivateIP advertise PublicIP) <----Private IP----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/3a289af05b05c69ae563ab29c9ec4710 In general, calls initiated from outside the system work perfectly, but I have a problem with calls initiated from Asterisk. According to what I was able to investigate and understand, the problem arises from the fact that Kamailio uses the public IP for the entire routing system that involves the Record-Route and Route Headers. This in turn causes Asterisk to get lost trying to send messages outside of the INITIAL INVITE transaction, for example, the ACK response to the 200 Ok received from the Endpoint since it tries to use Kamailio's public IP to which Asterisk doesn't have access. I tried to force Asterisk to send all messages to Kamailio's private IP regardless of their nature, but it doesn't seem like an appropriate solution. On the other hand, I tried to modify the architecture a bit so that it was something similar to this: Endpoints <---- Public IP----> Kamailio (listen PrivateIP:P1 advertise PublicIP) <---- PrivateIP-----> Kamailio (listen PrivateIP:P2) <---- PrivateIP ----> Asterisk CONFIG: https://gist.github.com/Carlos-Escalona94/4d681bb189c6190941d291965e123889 It seems to me that this would solve the problem, but I have two doubts about this architecture, the first is that I would like to know if there is an easier way to solve the problem that I am not considering, and on the other hand, I have not found a way to identify from which interface is receiving the message to perform the routing properly. Thanks for the attention. -- *Esta mensagem pode conter informação confidencial ou privilegiada, sendo seu sigilo protegido por lei. Se você não for o destinatário ou a pessoa autorizada a receber esta mensagem, não pode usar, copiar ou divulgar as informações nela contidas ou tomar qualquer ação baseada nessas informações. Se você recebeu esta mensagem por engano, por favor, avise imediatamente ao remetente, respondendo o e-mail e em seguida apague-a. Agradecemos sua cooperação. ** * *This message may contain confidential or privileged information and its confidentiality is protected by law. If you are not the addressed or authorized person to receive this message, you must not use, copy, disclose or take any action based on it or any information herein. If you have received this message by mistake, please advise the sender immediately by replying the e-mail and then deleting it. Thank you for your cooperation. *

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Getting "uac.reg_add" to work (documentation unclarities)
by Rick van Rein 29 Apr '22

29 Apr '22

Hello, I am trying to make UAC register with a SIP telco. But when I run... shell$ kamctl rpc uac.reg_add mydomain.com 0123456789 myteldomain.com 0123456789 sip6.telco.nl telcoreseller.nl 999999 secretpw . udp:sip6.telco.nl 360 0 10 'udp:[fd8:9::10]:5090' ...it gives me the cryptic response... ERROR: jsonrpcs [jsonrpcs_mod.c:666]: jsonrpc_scan(): field is not a string - type 3 The module docs only give "..." as example of attributes. And, according to... http://www.kamailio.net/docs/modules/5.4.x/modules/uac.html#uac.r.uac.reg_a… https://github.com/kamailio/kamailio/blob/5.4.4/src/modules/uac/uac_reg.c#1… ...it also seems to be behind on diff_expires, timer_expires, reg_init. What I am missing is the format, and in some cases the purpose of attributes. Could you help me to an example, perhaps based on the command I am trying? Thanks, -Rick P.S. It's been years since I worked with OpenSIPS. It feels like Kamailio is friendlier, and easier to use.

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Dispatcher Usage and change of $rU
by Benoît Panizzon 29 Apr '22

29 Apr '22

Hi Gang I have a observation which I don't understand... Let's assume we have an inbound $rU = 12345 And I want to change $rU outbound to 0012345 Doing: $rU = "0012345"; ds_select_dst("2000"),"6"); route(RELAY) The call is being sent out with unaltered R-URI Username: 12345 Doing: $rU = "0012345"; ds_select_domain("2000", "6"); route(RELAY); The call is being sent out with the intended new R-URI Username: 0012345 Looking at the docs: https://kamailio.org/docs/modules/5.4.x/modules/dispatcher.html#dispatcher.… I can not figure out, why I see this difference. -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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Incorrect Socket and Tcpconn-Id with Mid-Registrar and WebSockets
by Grant Bagdasarian 29 Apr '22

29 Apr '22

Hi, We're attempting to implement a mid-registrar for websocket connections but are running into some issues where the tcpconn-id and socket attributes are not stored correctly in memory (usrloc). REGISTER Flow: UA1 -> P1 (Mid-Registrar) ~> P2 (Core Proxy/Registrar) INVITE Flow: UA2 -> P2 -> P1 -> UA1 UA1: baresip cli connecting to P1 using websockets UA2: softphone Everything is running locally on a dev machine, and we're using Docker to launch P1 and P2. The below code snippet is what we use on P1, and has been simplified. The general idea is as following: 1) UA1 sets up the websocket connection and sends a REGISTER request to P1 2) P1 checks if this request has already been sent to P2 2.a) If not, then relay the request to P2. Upon receiving a 200 OK from P2 we save the registration in usrloc 2.b) If so, then absorb the request (save) and generate a reply 3) UA2 tries to call UA1 via P1 and P2 The problem occurs at 2.a since the REGISTER request is forwarded using UDP and the 200 OK is also received using UDP. When the registration is stored in reply route, the Socket (udp:x.x.x.x:xxxx) and Tcpconn-Id (-1) are not set to what we would prefer (see Output 1) TCP was also tried as the transport between P1 and P2. Upon receiving the reply from P2 the Tcpconn-Id and Socket of the connection between P1 and P2 was saved in the binding. When 2.b occurs the Socket and Tcpconn-Id are set to the correct values (see Output 2) Making a call when the binding state is at 2.a we receive the following error: "TCP/TLS connection (id: 0) for WebSocket could not be found" Making a call when the binding state is at 2.b works perfectly fine since the correct Tcpconn-Id is set. We did find a solution, or rather a workaround to this issue: 1) Always call the save function in request_route 2) If P2 responds with an error (non-200 OK) use the unregister function to remove the binding We also tried the tcpops tcp_set_otcpid and tcp_set_otcpid_flag functions after lookup, but this didn't do anything. KEMI Script def ksr_request_route(self, msg): if kemi.is_REGISTER(): aor = "aor" if result := kemi.htable.sht_is_null("registrations", aor) == 1: kemi.tm.t_on_reply("ksr_register_reply_route") kemi.dispatcher.ds_select_dst(1, 4) kemi.tm.t_relay() return 1 kemi.registrar.save("registrations", 1) return 1 if kemi.is_INVITE(): aor = "aor" if result := kemi.registrar.lookup_uri("registrations", aor): kemi.tm.t_relay() return 1 def ksr_register_reply_route(self, msg): if kemi.tm.t_check_status("200") == 1: kemi.registrar.save("registrations", 1) aor = "aor" kemi.htable.sht_sets("registrations", aor, "") return 1 Output 1: Binding state when request is being forwarded kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 13 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34818 User-Agent: n/a Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: udp:10.2.0.15:5060 Methods: -1 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 0 Last-Keepalive: 1651146965 KA-Roundtrip: 0 Last-Modified: 1651146965 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 2 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 3s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } } Output 2: Binding state when request is being absorbed kamcmd ul.dump { Domains: { Domain: { Domain: registrations Size: 1024 AoRs: { Info: { AoR: grant(a)127.0.0.1 HashID: 409622189 Contacts: { Contact: { Address: sip:grant-0x7fe915a79138@172.16.99.12:9;transport=ws Expires: 9 Q: 1.000000 Call-ID: 7cdfbf93f23bb54f CSeq: 34819 User-Agent: baresip v2.0.2 (x86_64/darwin) Received: sip:192.168.80.1:62184;transport=ws Path: [not set] State: CS_NEW Flags: 1 CFlags: 20 Socket: tcp:192.168.80.15:80 Methods: 5087 Ruid: uloc-626a80d3-46-1 Instance: <urn:uuid:c67df38d-d5e0-e834-1d98-a8df2d3f291b> Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 1 Keepalive: 0 Last-Keepalive: 1651146979 KA-Roundtrip: 0 Last-Modified: 1651146979 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } kamcmd core.tcp_list { id: 1 type: WS state: CONN_OK timeout: 3 lifetime: 120 ref_count: 2 src_ip: 192.168.80.1 src_port: 62184 dst_ip: 192.168.80.15 dst_port: 80 } kamcmd ws.dump { connections: { 1: ws:192.168.80.1:62184 -> ws:192.168.80.15:80 (state: OPEN, last used 1s ago, sub-protocol: sip) } info: { wscounter: 1 truncated: no } }

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