sr-users March 2022

sr-users@lists.kamailio.org

62 participants
94 discussions

bug in tm module?
by dpaape＠frontier.com 22 Mar '22

22 Mar '22

Hello, I'm running K5.5.3 and have checked the 'issues' link in github and saw nothing regarding what I am seeing. I think that there is a bug in the tm module with respect to the "tm:local-response" event route. In the "t_reply.c" file, there is a static variable called '_tm_local_response_set_lookup'. This variable is initialized at load time to zero. It is checked in the '_reply_light()' routine and will initiate a local callback from the config script if "armed". The problem as I see it is that if the callback is readied, the variable is set to one. But it is never reset.So the practical outworking is as follows. A register is received and the request_route arms the callback. The register requires and authorization. After the 401 is sent back, the callback via the module event route is called. All good except that the module event route script fragment is never executed again after the first call.I'm no expert on the tm module but I would think that somewhere in there, the variable should be reset to zero so that subsequent transactions can initiate the event route. thanks for any help,Dan

1 0

SERIAL FORKING : call to a group number
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear all, My use case is : I defined a username = 112 (for emergency call) This username doesn't exist in subscribers table and unavailable in location table. The idea is when Sopftphone1 call the group = 112 then Kamailio will forward the call to a group using SERIAL FORKING. => Softphone 1 -> Sends INVITE to 112 -> Kamailio -> FORWARD TO SERIAL FORKING (4x softphone all registered and in location table). The q priority could be hybrid that means 2x softphone could have same priority. My issues : - the q priority forced for each softphone is not taking into account. - the first branch created when send INVITE to 112 is still available. My setup Kamailio is behind a NAT. Kamailio version : sipsecure@kamailio:/$ kamailio -version 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 Kamailio.cfg # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 5sec modparam("tm", "fr_timer", 5000) # default invite retransmission timeout after 1xx: 10sec modparam("tm", "fr_inv_timer", 10000) # This is the name of an XAVP that the t_load_contacts() function uses to store contacts of the destination set and that t_next_contacts() function uses to restore those contacts. modparam("tm", "contacts_avp", "tm_contacts") modparam("tm", "contact_flows_avp", "tm_contact_flows") # modparam("tm|usrloc", "xavp_contact", "tm_contacts") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) route[INVITE]{ route(THREAD_CHECK); route(AUTH); route(RECORD_ROUTE); setflag(FLT_ACC); switch($tU) { case "112": xlog("MY_SEVERITY"," -- Calling Group Serial"); route(FORK_SERIAL); break; case "113": xlog("MY_SEVERITY"," -- Calling Group Parralelle"); route(FORK_PARALLEL); break; default: xlog("MY_SEVERITY"," -- Calling Someone"); route(LOCATION); route(RELAY); break; } exit; } route[FORK_SERIAL]{ t_on_reply("MANAGE_REPLY"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(IPV4V6); route(NATMANAGE); $var(uri1)="sip:9919995@192.168.1.41"; $var(uri2)="sip:9919994@192.168.1.41"; $var(uri3)="sip:9919991@192.168.1.41"; $var(uri4)="sip:9919997@192.168.1.41"; lookup("location","$var(uri4)"); append_branch("$ru","0.2"); lookup("location","$var(uri1)"); append_branch("$ru","0.8"); lookup("location","$var(uri2)"); append_branch("$ru","0.5"); lookup("location","$var(uri3)"); append_branch("$ru","0.5"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL"); #!ifdef WITH_SIPSECURE_DEBUG xlog("MY_SEVERITY"," ROUTE_SERIAL : Relaying \n"); #!endif #Relay (aka Forward) the request t_relay(); break; } failure_route[SERIAL]{ xlog("MY_SEVERITY","At failure route - Trying next destination"); if (!t_next_contacts()) { send_reply("408","Nobody available"); exit; } t_on_failure("SERIAL"); t_relay(); } Any highlight will be welcome. Best Regards, -- Youssef

1 0

$ru vs. $fu as r/w pseudo variables
by Juha Heinanen 22 Mar '22

22 Mar '22

If I set $ru in config file and then print $ru to log, the new value is printed. If I do the same with $fu, the old value is printed. I haven't found in Pseudo-Variables Wiki any documentation about this. Does it exist somewhere else? -- Juha

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MESSAGE METHOD - Send message to a softphone - $uac_req
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, I am expecting to send a MESSAGE METHOD to a softphone supporting MESSAGE. route[SEND_MESSAGE_METHOD]{ $uac_req(method)="MESSAGE"; $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; $uac_req(callid)=$(mb{s.md5}); $uac_req(hdrs)="Subject: Emergency Alert\r\n"; $uac_req(hdrs)=$uac_req(hdrs) + "Content-Type: text/plain\r\n"; $uac_req(body)="Emergency Message"; $uac_req(evroute)=1; uac_req_send(); } event_route[uac:reply] { xlog("MY_SEVERITY"," -- MESSAGE PROCESSING - received reply code is: $uac_req(evcode)\n");} My use case is to send message from KAMAILIO -> SOFTPHONE kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:192.168.1.29:5066 (physical ip address of the softphone) $uac_req(ruri)="sip:192.168.1.29:5066"; $uac_req(furi)="sip:192.168.1.41:5066"; // $uac_req(turi)="sip:9919995@192.168.1.24:5066"; The sip capture : 2022/03/18 02:20:58.947603 192.168.1.29:5066 -> 192.168.1.41:5066 MESSAGE sip:192.168.1.41:5066@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611 Max-Forwards: 70 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41> Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE User-Agent: MicroSIP/3.20.7 Content-Type: text/plain Content-Length: 3 ;-) 2022/03/18 02:20:58.948822 192.168.1.41:5066 -> 192.168.1.29:5066 SIP/2.0 200 Stored Via: SIP/2.0/UDP 192.168.1.29:5066;branch=z9hG4bKPj40bcc72324d647f8b96fc783e181f611;rport=5066 From: "codec7-private" <sip:9919997@192.168.1.41>;tag=416038ac55514b439a5cd5894770c3d8 To: <sip:192.168.1.41:5066@192.168.1.41>;tag=cbfe6543b9079987bce3b5b9a0d1195a.b63b503b Call-ID: 9381648157be49ff845b5e413dc9a95c CSeq: 9148 MESSAGE Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 Now, if I want to use his username@domain kamailio : sip:192.168.1.41:5066 (physical ip address of the kamailio) softphone : sip:9919997@192.168.1.41:5066 The message is sent to kamailio itself not to the softphone 2022/03/18 02:30:39.285927 192.168.1.41:5060 -> 192.168.1.41:5066 MESSAGE sip:9919997@192.168.1.41:5066 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0 To: <sip:9919997@192.168.1.41:5066> From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Max-Forwards: 70 Content-Length: 17 User-Agent: kamailio (5.5.4 (x86_64/linux)) Subject: Emergency Alert Content-Type: text/plain Emergency Message 2022/03/18 02:30:39.287590 192.168.1.41:5066 -> 192.168.1.41:5060 SIP/2.0 200 Stored Via: SIP/2.0/UDP xxx.xxx.xxx.210:5060;branch=z9hG4bK0cb8.a39b5180000000000000000000000000.0;rport=5060;received=192.168.1.41 To: <sip:9919997@192.168.1.41:5066>;tag=cbfe6543b9079987bce3b5b9a0d1195a.76a88393 From: <sip:192.168.1.41:5066>;tag=b4685c5f5f5fc30f8dcbb49a48fe9e25-4047894c CSeq: 10 MESSAGE Call-ID: 4cb973823835008ec4b20d1c67d9fb84 Server: kamailio (5.5.4 (x86_64/linux)) Content-Length: 0 - finaly, how to relay the message from one softphone to another softphone through kamailio proxy? Best Regards -- Youssef

2 3

Re: [SR-Users] Kamailio : Serial forking
by Paclinio 22 Mar '22

22 Mar '22

Following the example of TM Module. https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… Code : modparam("tm", "contacts_avp", "tm_contacts"); > > route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com", "0.5"); > append_branch("sip:c@example.com", "0.5"); > append_branch("sip:d@example.com", "1.0"); > > t_load_contacts(); > > t_next_contacts(); > t_on_failure("serial"); > t_relay(); > break; > } > > failure_route["serial"] > { > if (!t_next_contacts()) { > exit; > } > > t_on_failure("serial"); > t_relay(); > } > My code : failure_route[GroupFailureRoute]{ > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > } > > route[GROUP_RELAY_SER]{ > append_branch("sip:9919995@192.168.1.41","1"); > append_branch("sip:9919992@192.168.1.41", "0.21"); > append_branch("sip:9919997@192.168.1.41", "0.22"); > append_branch("sip:9919991@192.168.1.41", "0.22"); > t_load_contacts(); > t_next_contacts(); > while(!route(GROUP_LOCATION)) > { > if (!t_next_contacts()) { > send_reply("408","Timeout or nobody available"); > exit; > } > } > t_on_failure("GroupFailureRoute"); > t_relay(); > break; > } > The result : [image: image.png] > > public2_addr > 192.168.1.41 192.168.1.24 > 192.168.1.10 > ──────────┬───────── ──────────┬───────── > ──────────┬───────── ──────────┬───────── > > │ INV (192.168.1.41) │ > | > > │ audio 39776 (g711u) │ > | > > │ > ─────────────────────────────── ──────────────> │ > > │ INV > (192.168.1.41) │ > │ > > │─┐ audio 39776 (g711u) │ > │ > > │<┘ > │ > │ > > │─┐ 407 Proxy Authentication R│ │ > > │<┘ │ > │ > > │ > CANCEL │ > │ > > │ ─────────────────────> │ > │ > > │─┐ ACK > │ > │ > > │<┘ > │ > │ > > │ 487 Request Terminated │ > │ > > │ <────────────────────── │ > │ > > │ > ACK │ > │ > > │ ─────────────────────> │ > │ > > │ 200 OK │ │ > > │ <───────────────────── │ > │ > > │ 100 Trying │ > > │ <────────────────────────────────────────────── │ > > │ 180 Ringing │ > > │ <─────────────────────────────────────────── ── │ > │ 180 Ringing > │ │ > │ > │ <─────────────────── │ > │ > │ > On the Second "wave" of INVITE with the parallel only the first one is correctly changed with the lookup("location") and not the second. Could someone give me some tips? Best regards Thomas <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Re: [SR-Users] Kamailio : Parallel Forking
by Paclinio 22 Mar '22

22 Mar '22

Dear, Following the configuration available on the TM Modules documentation. Src : https://www.kamailio.org/docs/modules/devel/modules/tm.html#tm.serial_forki… request_route { > seturi("sip:a@example.com"); > append_branch("sip:b@example.com"); > append_branch("sip:c@example.com"); > append_branch("sip:d@example.com"); > > t_relay(); > break; > } > > My code : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > append_branch ("$var(uri1"); > append_branch ("$var(uri2"); > append_branch ("$var(uri3"); > append_branct_("$var(uri4"); > t_relay(); > break; > } > The server send INVITE to himself. Fixing the problem : request_route{ > $var(uri1)="sip:9919995@192.168.1.41"; > $var(uri2)="sip:9919992@192.168.1.41"; > $var(uri3)="sip:9919991@192.168.1.41"; > $var(uri4)="sip:9919994@192.168.1.41"; > > lookup("location","$var(uri4)"); > append_branch(); > lookup("location","$var(uri1)"); > append_branch(); > lookup("location","$var(uri2)"); > append_branch(); > lookup("location","$var(uri3)"); > t_relay(); > break; > > } > Is this the correct way ? Best regards. Thomas <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

1 0

Re: [SR-Users] MESSAGE METHOD - Send message to a softphone - $uac_req
by Boujraf, Youssef 22 Mar '22

22 Mar '22

Dear, then could I use to retrieve the user from location then set the ruri ? $var(uri1)="sip:9919995@192.168.1.41"; > lookup("location","$var(uri1)"); > > $uac_req(ruri)=$ru; > Best Regards *Youssef *

1 0

Re: [SR-Users] ?==?utf-8?q? Table : address - address_reload?==?utf-8?q? (exported MI function) - expect to call this function in kamailio.cfg to reload in cache the address
by Youssef Boujraf 22 Mar '22

22 Mar '22

Dear, When using : sudo kamctl address reload sipsecure@kamailio:/$ sudo kamctl address reload [sudo] password for sipsecure: { "jsonrpc": "2.0", "result": "Reload OK", "id": 73482 } I looked for the command I could use in kamailio.cfg route[JSON_ADDRESS_RELOAD]{ xlog("MY_SEVERITY"," RPC REQUEST : RELOAD ADDRESS TABLE"); jsonrpc_exec('{"jsonrpc": "2.0","method": "permissions.addressReload"}'); exit; } Now I am able to call a rpc command from the code ;-) Best Regards -- Youssef

1 0

Periodic keepalive for WebRTC clients
by Xuo Guoto 22 Mar '22

22 Mar '22

Hello all, I have a [SIP client] <-> Kamailio <-> Asterisk installation. Sip clients are both WebRTC and Android app. WebRTC connects via 443 while Android app connects VIA TLS at port 5061. WebRTC client uses jsSIP library. My Kamailio version is 5.5.3 (x86_64/linux) I had been using asterisk directly to terminate WebRTC clients previously and was using the qualify feature to periodically send OPTIONS packet to all connected clients as a keep alive mechanism. This allowed the client to detect any loss of connectivity and reconnect back to server if there is any loss of connectivity. After moving to Kamailio I am looking for a similar feature, where Kamailio sends OPTIONS (or any other packets) periodically to the WebRTC clients. My attempts so far hasn't been a success. I am aware of couple of options for keepliave in kamailio: 1. nat_keeplaive() I have some what naively copied the example from nat_keepalive() documentation into route[NATDETECT] but that is not working. route[NATDETECT] { force_rport(); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { if (is_method("REGISTER")) { fix_nated_register(); } else { if (!add_contact_alias()) { sl_send_reply("400", "Bad Request"); exit; } } setflag(FLT_NATS); return; } #!endif if (nat_uac_test(19)) { if (is_method("REGISTER")) { fix_nated_register(); } fix_contact(); } setflag(FLT_NATS); if ((method=="REGISTER" || method=="SUBSCRIBE" || (method=="INVITE" && !has_totag())) && client_nat_test("3")) { nat_keepalive(); } return; } 2. keepalive module This looks like what I want, but I am not sure how to add a new WebRTC connection to the list of endpoints for pinging. This is the handler for xhttp:request, should the ka_add_destination be added here? event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xhttp_reply("403", "Forbidden", "", ""); exit; } if (ws_handle_handshake()) { exit; } } xhttp_reply("404", "Not Found", "", ""); } My search so far has not returned any other means to send options packets to WebRTC clients. Been banging my head around this for couple of days, and any help will be much appreciated. X.

1 0

Dialog module errors with dmq
by Olli Attila 22 Mar '22

22 Mar '22

Hello, I have two Kamailio nodes. Couple of days of ago I enabled dmq to replicate dialogs between these two Kamailio nodes. Now that the dmq is enabled, we are seeing lot of warning and notice messages on kamailio logs related to the dialog module. Is there something I have missed when configuring dlg_manage & dmq replication that might cause this? No calls have been affected negatively by this though. Kamailio 1 node prints lot of this: Mar 4 08:20:25 /usr/sbin/kamailio[14316]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:20:28 /usr/sbin/kamailio[14313]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for BYE with route param 'a5d.ebf1' [3418:8126] and call-id 'SDiu98001-dd9c12fea650b9d1408dd672a2c51c9f-v300g00020' Mar 4 08:21:56 /usr/sbin/kamailio[14315]: WARNING: dialog [dlg_handlers.c:1328]: dlg_onroute(): unable to find dialog for ACK with route param '689.f2f1' [2438:7983] and call-id 'SD8o63c01-47972a87ee1fd8791ddf27b650badcba-v300g00010' Kamailio 2 node prints lot of this: Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b84e980 ref 1) Mar 4 08:20:29 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07bb03360 ref 1) Mar 4 08:21:59 /usr/sbin/kamailio[8005]: NOTICE: dialog [dlg_hash.c:242]: dlg_clean_run(): dialog in early state is too old (0x7fe07b528300 ref 1) (Version: kamailio 5.5.3 (x86_64/linux) 473cef on both kamailio nodes) [root@kamailio-1]# kamcmd dmq.list_nodes { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 0 } { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 1 } [root@kamailio-2]# kamcmd dmq.list_nodes { host: 172.19.194.132 port: 5060 proto: * resolved_ip: 172.19.194.132 status: active last_notification: 0 local: 0 } { host: 172.19.194.140 port: 5060 proto: * resolved_ip: 172.19.194.140 status: active last_notification: 0 local: 1 } ### On Kamailio 1: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.132:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.140:5060" ### On Kamailio 2: ### #!define DMQ_LOCAL_ADDR "sip:172.19.194.140:5060" #!define DMQ_PEER_ADDR "sip:172.19.194.132:5060" ### On Kamailio 1 & 2:### modparam("dmq", "server_address", DMQ_LOCAL_ADDR) modparam("dmq", "notification_address", DMQ_PEER_ADDR) modparam("dialog", "profiles_with_value", "total_dialogs") modparam("dialog", "enable_dmq", 1) route[INITIAL] { ...located at the bottom of the initial route... if(is_method("INVITE") && !has_totag()) { dlg_manage(); } } Cheers, Olli

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sr-users March 2022