- sr-users - kamailio.org

tel2sip is giving unpredictable result
by Jyoti Bansal 31 Jan '22

31 Jan '22

hello, we are using kamailio 5.3 . while establishing a call the tel uri is getting converted into sip uri sometimes incorrectly . In the sip uri some extra characters are getting appended in the invite message. Can somebody please provide some guidance on this, attaching the pcap for reference(packet no. 8). Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

kamailio log dialog correlation
by marek 31 Jan '22

31 Jan '22

hi, is it possible create log of SIP dialog like https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ? it looks like its possible like this xlog("L_INFO", "D$dlg(h_id) something to log \n"); but i must copy $dlg(h_id) in every xlog tried this modparam("xlog", "prefix", "D$dlg(h_id)") but it not works with variables any other tips/ideas? goal is to correlate logs for one dialog (call) thanks Marek

7 16

REGISTER with http_async_client?
by Cyril Ramière 31 Jan '22

31 Jan '22

Hi there, I have a weird issue with kamailio (latest docker image kamailio-ci:5.5.2-alpine) and http_async_client. Before posting a lot of logs, let me describe what I want to achieve. I have a Kamailio and a SIP Phone. The SIP phone sends a REGISTER to kamailio, then in my routing block, I check if I have an Authorization header. Since I don't have an Authorization (first message), I use "www_challenge()". This replies to the SIP phone, and then the SIP phone sends a new REGISTER with the correct Authorization header. So far so good. Now, when I get the REGISTER with Authorization header, I want to ask an HTTP endpoint if this user is allowed to connect and check the password using http_async_query(). The problem is that when the transaction resumes, the tmx module is unhappy and throws this error : 30(36) CRITICAL: tmx [t_var.c:546]: pv_get_tm_reply_code(): no picked branch (-1) for a final response in MODE_ONFAILURE And a 500 error is sent back to the sip phone. The AUTH_REPLY route is still called and I can use the $http* values. It seems that my AUTH_REPLY route is called but considered as a "failure_route" 19(27) exec: *** cfgtrace:dbg_cfg_trace(): failure_route=[AUTH_REPLY] c=[/etc/kamailio/kamailio.cfg] l=665 a=16 n=if 17(23) NOTICE: <script>: [AUTH_REPLY] FAILURE ROUTE If I remove the http_async_client and try with route(AUTH_REPLY) then it is considered as request_route (but this is not a solution since I need to do this http request) Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$td","1"); exit; } else{ t_newtran(); http_async_query("http://xxx.xxx.xxx.xxx:9000/auth?foo=bar", "AUTH_REPLY"); } } } route[AUTH_REPLY]{ xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); if(t_is_failure_route()) { xlog("L_NOTICE", "[AUTH_REPLY] FAILURE ROUTE \n\n"); } } } ##### END SNIP Best regards!

1 1

Use specific expires per User-Agent
by mayamatakeshi 31 Jan '22

31 Jan '22

Hi, I have a requirement to specify a REGISTER max_expires based on the User-Agent. I checked the docs and it seems there is no pseudo-variable or something like that would permit this. I would like to be able to do something like this: if($ua ~= "Some-UA") { $max_expires=100; } else { $max_expires=150; } save("location"); Is there some way to achieve this?

2 2

failed to find string from message body
by Juha Heinanen 29 Jan '22

29 Jan '22

I tried to check if message body contains a string using function call str_find("$rb", "string")) but it failed like this: Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1491]: pv_printf_mode(): no more space for spec value - printed:0 token:6880 buffer:4096 Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/pvapi.c:1503]: pv_printf_mode(): buffer overflow -- increase the buffer size... Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: <core> [core/sr_module.c:1698]: get_str_fparam(): Could not convert the PV-formated string to str Jan 29 17:37:03 siika /usr/bin/sip-proxy[2577003]: ERROR: textops [textops.c:4457]: str_find_f(): cannot get first parameter Any suggestions on how to get it done properly? -- Juha

1 0

Question about siptrace destination/source port capture in HEPv3 mode
by Karsten Horsmann 29 Jan '22

29 Jan '22

Hello List, I observe an interesting behavior. We use siptrace as an active capture agent for QXIP HOMER/HEPIC installations. The destination port and the source port in the HEPv3 Header are not correct if you use transport tcp for example. My Question: is that intended or a bug and i should fill up an issue? On udp, it's filled with 5060 or whatever your listening directive says. I tested that with kamailio 5.5 and kamailio master. I attached an HEPv3 capture from two kamailios (master) speaking with each other. You need the https://github.com/sipcapture/hep-wireshark lua dissector to read the capture. 10.0.2.15 (kamailio with uacreg and sipcapture/HEPv3 mode) 192.168.50.4 (kamailio as registrar) Frame 1 shows an tcp transport register Request from 10.0.2.15 to 192.168.50.4 in HEPv3. The Source Port in the HEP3 Protocol is not correct (its an highport / 45419). Frame 4 shows the 401 answer and the Destination port is 0. Frame 7 also had this behavior. ----- params (nearly at the end): modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", MY_HOMER_CAPTURE_ID) # capture agent id Limitation: 32-bit for HEPv3. modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_mode", 1) # default 0, if 1 then you dont need call siptrace flag or sip_trace() -- Mit freundlichen Grüßen / Kind Regards *Karsten Horsmann*

3 2

tel2sip is not working correctly
by Jyoti Bansal 29 Jan '22

29 Jan '22

Hello all, I am using kamailio code to establish calls with the real ue's. Our calls are getting established with an issue , whenever invite is send from A party to B party with teluri that tel uri is converted to sip uri. While converting to sip uri some extra bytes are getting added after user=phone and the call ends. Can someone help us on this issue of getting some extra bytes added in the sip uri? for example :- sip:405874112224411@ims.mnc123.mcc456.3gppnetwork.org;user=phone�� Thanks, *Jyoti Bansal* Software Engineer . Great Software Laboratory | www.gslab.com -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

1 0

Comunication between devices RTP/SRTP and viceversa
by Social Boh 28 Jan '22

28 Jan '22

I'm trying to configure Kamailio to permit communication between devices: some use TLS and SRTP some UDP and RTP with RTPEngine in the middle. A priori I don't know if the device support SRTP or RTP so in the routing I have to detect on the fly how to act. The problem is this: Call from RTP to SRTP device: 488 Not Acceptable here Call from SRTP to RTP device: 415 Unsupported Media Type Any hint? Thank you Regards -- --- I'm SoCIaL, MayBe

4 5

200 OK re transmission only on re-SUBSCRIBE
by Lewis Hutchinson 27 Jan '22

27 Jan '22

First of all I should start by saying this is my first post so go easy on me!! : ) and thank for everyone's efforts to share knowledge it has been invaluable for my kamailio learning. Build Details Version: kamailio 5.5.3 (x86_64/linux) Kamailio kemi python Scenario Relevant config def ksr_route_api_query(self, msg): KSR.log("info", "SUBSCRIBE API Query Starts") # Create new Transaction KSR.tm.t_newtran() async def main(): url = 'https://api.hidingtheurl.com/subscribe' payload = { "ruri": KSR.pv.get("$ru"), "from": KSR.pv.get("$fu"), "from_tag": KSR.pv.gete("$ft"), "to": KSR.pv.get("$tu"), "callid": KSR.pv.get("$ci"), "cseq": KSR.pv.get("$cs"), "contact": KSR.pv.gete("$ct"), "supported": KSR.pv.get("$hdr(Supported)"), "event": KSR.pv.get("$hdr(Event)"), "expires": KSR.pv.get("$hdr(Expires)"), "diversion": KSR.pv.get("$di"), "body": KSR.pv.get("$rb") } async with aiohttp.ClientSession() as session: async with session.post(url, json=payload) as response: if response.status == 0: KSR.xlog.xinfo(f"SUB Manager Timeout: {response.status}") KSR.tm.t_reply(408, "SUB Manager Timeout") elif response.status >= 500: KSR.xlog.xinfo(f"SUB Manager Down: {response.status}") KSR.tm.t_reply(500, "SUB Manager Down") elif response.status >= 400: KSR.xlog.xinfo(f"SUB Manager Error: {response.status}") KSR.tm.t_reply(400, "SUB Manager Error") else: if response.status == 200: KSR.xlog.xinfo(f"SUB Manager Success: {response.status}") KSR.tm.t_reply(200, "OK") KSR.log("info", "SUBSCRIBE API Query Ends") asyncio.run(main()) # API Query Ends Scenario Initial SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get receive 200OK response from API and send a 200OK on to Device SUBSCRIBE --------> <------------ 200 OK After Subscription Expiry re-SUBSCRIBE >>> hits Kamailio >>> send payload in JSON to API get 200OK response from API and send a 200OK on to Device but also another 200OK a fraction of a second later... SUBSCRIBE --------> <------------ 200 OK <<<--------- 200 OK This happens for all following re-SUBSCRIBE's The 200 OK are identical, and it is a re transmission, but I can't work out why.. Relevant DEBUG is below Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: INFO: <script>: SUB Manager Success: 200 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_reply Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:287]: sr_apy_kemi_exec_func_ex(): params[2] for: t_reply are int-str: [200] [OK] Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/msg_translator.c:162]: check_via_address(): (80.111.111.111, 80.111.111.111, 0) Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1763]: cleanup_uac_timers(): RETR/FR timers reset Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7eff0cd50938: SIP/2.0 200 OK#015#012Via:... shmem=0x7eff09190ca8: SIP/2.0 200 OK#015#012Via: Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:648]: _reply_light(): finished Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_precheck_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:232]: sr_apy_kemi_exec_func_ex(): execution of method: t_check_trans Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7eff0cd4e828) id=2/11234 global id=2/11234 T start=0x7eff0918d7d8 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_lookup.c:1108]: t_check_msg(): T (0x7eff0918d7d8) already found for msg (0x7eff0cd4e828)! Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: tm [t_reply.c:1703]: t_retransmit_reply(): reply retransmitted. buf=0x7eff0cb18220: SIP/2.0 2..., shmem=0x7eff09190ca8: SIP/2.0 2 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: app_python3 [apy_kemi.c:112]: sr_kemi_config_engine_python(): execution of route type 1 with no name returned 1 Jan 26 12:31:17 ip-172-20-46-39 kamailio[11234]: DEBUG: <core> [core/receive.c:514]: receive_msg(): request-route executed in: 38258 usec Everything works well as is but I want to clean up this issue or at least understand what is causing it. Lewis

2 6

Some problems with SIP TRACE
by Arnaud 27 Jan '22

27 Jan '22

Hi, I'm trying to use siptrace to getting copy of my SIP transactions, especially in TLS. Here's my configuration : *loadmodule "siptrace.so" modparam("siptrace", "trace_mode", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "duplicate_uri", "sip:MY_IP:5065") modparam("siptrace", "trace_local_ip", "MY_IP:5061") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_capture_id", 1) modparam("siptrace", "trace_flag",22); * And in my Routing Logic : *request_route {* *sip_trace();setflag(22);* *(...)* But I didn't receive anything on my listening port 5065 according my TCPDUMP. Status confirm me siptrace is working : *root@debian:/# kamcmd siptrace.status checkEnabled* I think I made a mistake in the configuration, but I don't know where. My second question is when I try to use sip_trace_mode("t"); I getting : failed to find command sip_trace_mode. Is it in an other module ? Many thanks for your help. Regards, Arnaud

2 1

Re: [SR-Users] sr-users Digest, Vol 200, Issue 25
by surendra tiwari 27 Jan '22

27 Jan '22

Hi Aditya, There are 2 options to achieve this: Option-1 use below param in auth_db module and while doing authentication change table name to your table user_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> domain_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> password_column (string) <https://www.kamailio.org/docs/modules/devel/modules/auth_db.html#auth_db.p.…> Option-2 create a new http service that will connect to kamailio via evapi and will be responsible for authentication and location service. On Wed, 26 Jan 2022 at 16:31, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Kamailio as a SIP registrar (Aditya Uppuluri) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Jan 2022 16:21:33 +0530 > From: Aditya Uppuluri <uppuluriaditya(a)gmail.com> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Kamailio as a SIP registrar > Message-ID: > <CAKWpYb= > 9EABvpmbPhvCyt-roQaOkiJnZSh17vVu6Wk26PiBbeg(a)mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hello Team, > Thank you for the amazing product. I am a newbie to kamailio and currently > trying to use kamailio as a SIP Registrar (for a multidomain). Here is the > requirement > > 1. We already have a mysql database (name: VOIP) with a table named > VoipSubscribers which contains the userid, password and domain information > of all the users. > 2. We want the ability in kamailio to use the same VoipSubscribers table > for user registrations and also store the location information in the same > database (VOIP) > > May I know how to achieve this without altering the table schema? > Thank you in advance. > > Regards, > Aditya Uppuluri >

1 0

Kamailio as a SIP registrar
by Aditya Uppuluri 26 Jan '22

26 Jan '22

Hello Team, Thank you for the amazing product. I am a newbie to kamailio and currently trying to use kamailio as a SIP Registrar (for a multidomain). Here is the requirement 1. We already have a mysql database (name: VOIP) with a table named VoipSubscribers which contains the userid, password and domain information of all the users. 2. We want the ability in kamailio to use the same VoipSubscribers table for user registrations and also store the location information in the same database (VOIP) May I know how to achieve this without altering the table schema? Thank you in advance. Regards, Aditya Uppuluri

2 1

Append HF don't work in event_route
by Arnaud 24 Jan '22

24 Jan '22

Hello, I'm trying to connect kamailio to Microsoft Teams. Dispatch is correctly configured and sended. Following this tutorial : https://skalatan.de/en/blog/kamailio-sbc-teams, I added an Event_Route to add contact header to SIP OPTION from dispath to MS : event_route[tm:local-request] { xlog("*** tm:local-request *** "); if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { xlog("L_INFO", "Append new contact !\n"); append_hf("Contact: <sip:MY_SBC_NAME.COM:5061 ;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } When I check the log to see my sended frame, I don't see a contact header field : INFO: <script>: Append new contact ! INFO: <script>: Sent out tm request : OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY_SBC_NAME.COM:5061 ;branch=z9hG4bK4497.f3e5e754000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:MY_SBC_NAME.COM.fr>;tag=a76e006ace3802951ed860e08df964af-7971 CSeq: 10 OPTIONS Call-ID: 2e1278ea77796160-9837@MY_SBC_IP Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.2.1 (x86_64/linux)) According the documentation about textop, append_hf(txt[, hdr]) : This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE. So my question is append HF can be used in Event_route ? Is there any other way to add contact header ? Regards, Arnaud

1 0

g_hash_table_insert_internal: assertion 'hash_table != NULL' failed
by Muhammad Danish Moosa 23 Jan '22

23 Jan '22

Hi, I am running stress tests on kamalilio , at times i see this on screen (process:30959): GLib-CRITICAL **: 21:43:56.961: g_hash_table_insert_internal: assertion 'hash_table != NULL' failed version: kamailio 5.5.3 Kamailio does not crash and keeps accepting calls though. Is it anything alarming. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

1 0

dispatcher lb
by Gokan Atmaca 22 Jan '22

22 Jan '22

Hello I'm load balancing with Dispatcher. The call comes in but hangs up without picking up the phone. What could be the problem? The settings are as follows. dispatcher.list 1 sip:18.X.X.X.X:5060 0 0 maxload=20 1 sip:67.X.X.X.X:5060 0 0 maxload=20 # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 4.130000 STD: 0.398000 EST: 4.006000 MAX: 6 TIMEOUT: 0 } } DEST: { URI: sip:X.X.X.X:5060 FLAGS: AP PRIORITY: 0 ATTRS: { BODY: maxload=20 DUID: MAXLOAD: 20 WEIGHT: 0 RWEIGHT: 0 SOCKET: } LATENCY: { AVG: 22.282000 STD: 2.345000 EST: 23.075000 MAX: 42 TIMEOUT: 0 } } } } } } -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

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Effect of GRUU on the behavior of loose_route()
by Denys Pozniak 21 Jan '22

21 Jan '22

Hello! I ask for help in explaining the different behavior of loose_route() function in two similar cases (depending on GRUU parameter on the end device). The call path goes through a series of Proxies: Proxy Registrar (use Path) -> Proxy Edge -> PhonerLite (GRUU on/off). If the GRUU parameter is enabled on the PhonerLite, then the in-dialog SIP ACK (on 200 OK/SDP) message is routed on the Proxy Edge successfully (loose_route() deletes Route and handle_ruri_alias() sets $du). When GRUU is disabled - loose_route() for some reason replaces RURI (but lr=on ?) and sets $du. The configuration of the [WITHINDLG] route section is similar to the default script. IP: 1.1.1.1 is the Proxy Registrar IP: 2.2.2.2 is the Proxy Edge IP: 3.3.3.3 is the Phone (Phoner Lite) Successful case (GRUU enabled): 2022/01/21 20:08:20.745859 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:20.746952 2.2.2.2:5060 -> 3.3.3.3:5065 ACK sip:userid@2.2.2.2:5060;gr=009356C7-5A79-EC11-8468-538E88103B96 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK0a34.2b5076a5473cc42d883bf38969aa7dcf.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK0a34.0bcc52c86238bd3a73bd88712c562d10.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK0a34.1bb19d987d7566fe74d300766284d73a.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bKkr2inaF~;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=41A6E617-61EB049F000008A2-86EEE700 To: <sip:userid@10.60.27.84>;tag=00fd16295b79ec118469538e88103b96 CSeq: 10 ACK Call-ID: 537f0686-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 3(11755) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 3(11755) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 3(11755) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 3(11755) DEBUG: rr [loose.c:856]: after_loose(): Topmost route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=28c.4fa' is me 3(11755) DEBUG: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header 3(11755) DEBUG: rr [loose.c:181]: find_next_route(): No next Route HF found 3(11755) DEBUG: rr [loose.c:886]: after_loose(): No next URI found 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 3(11755) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <lr=on;nat=yes;did=28c.4fa> 3(11755) DEBUG: nathelper [nathelper.c:1202]: handle_ruri_alias(): setting dst_uri to <sip:3.3.3.3:5065> 3(11755) DEBUG: nathelper [nathelper.c:1224]: handle_ruri_alias(): rewriting r-uri to <sip:userid@2.2.2.2:5060 ;gr=009356C7-5A79-EC11-8468-538E88103B96> Unsuccessful case (GRUU disabled): 2022/01/21 20:08:47.851170 1.1.1.1:5060 -> 2.2.2.2:5060 ACK sip:userid@2.2.2.2:5060;alias=3.3.3.3~5065~1 SIP/2.0 Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Route: <sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83> Max-Forwards: 63 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 2022/01/21 20:08:47.851963 2.2.2.2:5060 -> 2.2.2.2:5060 ACK sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83 SIP/2.0 Via: SIP/2.0/UDP 2.2.2.2;branch=z9hG4bK110e.af605388ece54d48387c1de53615d894.0 Via: SIP/2.0/UDP 1.1.1.1;rport=5060;branch=z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0 Via: SIP/2.0/UDP 10.60.27.126;rport=5060;branch=z9hG4bK110e.4d82c337912ca1187474aa2474a7d563.0 Via: SIP/2.0/UDP 10.60.27.84:5080 ;received=10.60.27.84;branch=z9hG4bK4bKmGaAy;rport=5080 From: "Denys VOIP" <sip:5009@1.1.1.1>;tag=0FBBBFFE-61EB04BB000A3B4F-86EEE700 To: <sip:userid@10.60.27.84>;tag=80dc2e395b79ec11846d538e88103b96 CSeq: 10 ACK Call-ID: 64957c76-f590-123a-ffb5-001dd8b71cb2_userid Max-Forwards: 62 Contact: <sip:10.60.27.84:5080;transport=udp> Content-Length: 0 DEBUG: 1(11753) DEBUG: rr [loose.c:85]: is_preloaded(): is_preloaded: No 1(11753) DEBUG: <core> [core/socket_info.c:646]: grep_sock_info(): checking if host==us: 13==13 && [2.2.2.2] == [2.2.2.2] 1(11753) DEBUG: <core> [core/socket_info.c:649]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 1(11753) DEBUG: <core> [core/forward.c:429]: check_self(): host (0: 2.2.2.2:5060) == me 1(11753) DEBUG: rr [loose.c:676]: after_strict(): Next hop: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' is loose router 1(11753) DEBUG: rr [loose.c:726]: after_strict(): The last route URI: 'sip:2.2.2.2;lr=on;nat=yes;did=3f8.6a83' 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 1 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: path [path.c:397]: path_rr_callback(): no received parameter in route header 1(11753) DEBUG: rr [rr_cb.c:95]: run_rr_callbacks(): callback id 0 entered with <alias=3.3.3.3~5065~1> 1(11753) DEBUG: rr [record.c:977]: add_rr_param(): rr_param_buf=<;did=3f8.6a83> 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(11753) DEBUG: tm [t_lookup.c:1329]: t_newtran(): msg (0x7efe343f4998) id=11/11753 global id=10/11753 T start=0xffffffffffffffff 1(11753) DEBUG: tm [t_lookup.c:498]: t_lookup_request(): start searching: hash=57361, isACK=1 1(11753) DEBUG: tm [t_lookup.c:456]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK110e.a47bb0e8f6d6d99a10ae36eb25d2f665.0] 1(11753) DEBUG: tm [t_lookup.c:675]: t_lookup_request(): no transaction found 1(11753) DEBUG: tm [t_funcs.c:286]: t_relay_to(): forwarding ACK statelessly -- BR, Denys Pozniak

1 0

Experiences with Kamailio on Debian 11 Bullseye
by Daniel-Constantin Mierla 20 Jan '22

20 Jan '22

Hello, has anyone experienced issues with Kamailio running on Debian 11 Bullseye? It's now more than 2 months since Debian 11 release and I wonder if people started to migrate to it for production systems. So far I did only basic tests on it, although I have a Debian Testing that I used for compilation and simple tests during the past several months before Debian 11 release. If someone discovers issues or changes in behaviour with various libraries or applications/database servers, report them to the tracker to try to address them as soon as possible. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 08-11, 2021 (Europe Timezone) - Nov 22-25, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

8 14

Handling special characters - KEMI lua redis_ndb
by Muhammad Danish Moosa 20 Jan '22

20 Jan '22

Hi Guys, I am trying to do following code KSR.ndb_redis.redis_cmd("srvN", "SET var " .. testvar, "r"); it works fine but if testvar has special character like SPACE or % , it does not work. For example when testvar="asdsada%Dasda"; it starts considering %D as place holder and does not add key to redis (does not raise error as well) With Space I tried to use delimeters like "" or '' but none of them work. For now I have got it working through , search and replace but thats not a clean solution. Any suggestions? -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 1

Does cfgutils usleep block the process?
by mayamatakeshi 20 Jan '22

20 Jan '22

Hi, it seems execution of usleep eventually reaches this code: https://github.com/kamailio/kamailio/blob/master/src/core/ut.h#L508 so it seems it would block the process and so it might cause delay of packet processing if many processes do this at the same time. Can someone confirm this?

2 2

use $var or $avp in ds_list_exists & ds_select dispatcher function
by Patrick Karton 19 Jan '22

19 Jan '22

Hello im using kamailio 5.5.3 and im trying to use pseudo vars in ds_list_exists and ds_select function but it does not work. ds_list_exists return -2 with ds_list_exists("6") it works $var(dset)=6 or $avp(dset)=6 it does not work ds_list_exists("$var(dset)") --> return -2 ds_list_exists("$avp(dset)") --> return -2 Thanks.

1 0

Fwd: Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 19 Jan '22

19 Jan '22

Forwarding my reply to the list, using gmail's reply button set Henning as the sole recipient :-\ ---------- Forwarded message --------- From: George Diamantopoulos <georgediam(a)gmail.com> Date: Sat, 26 Jun 2021 at 02:25 Subject: Re: [SR-Users] Possible memory leak on 5.5.x (new)? To: Henning Westerholt <hw(a)skalatan.de> Hello Henning, Thanks for your reply. Here's what has come up after a few hours: shm55: https://pastebin.com/h9JCePmc shm54: https://pastebin.com/Nx5xEEnA It seems to me htable is the culprit? Are you seeing anything different? 54 has been running for 77020 seconds, 55 for 28521 (significantly less). I'm going to turn it off until we figure something out... BR, George On Fri, 25 Jun 2021 at 18:17, Henning Westerholt <hw(a)skalatan.de> wrote: > Hello, > > > > Good observation. Please run the memory statistics CLI commands to get > more hints about the module that might cause it (as per below link). Then > please report more details. If you can point to a particular module, you > can also open an issue on our tracker. > > > > https://www.kamailio.org/wiki/tutorials/troubleshooting/memory > > > > Cheers, > > > > Henning > > > > *From:* sr-users <sr-users-bounces(a)lists.kamailio.org> *On Behalf Of *George > Diamantopoulos > *Sent:* Friday, June 25, 2021 4:53 PM > *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> > *Subject:* [SR-Users] Possible memory leak on 5.5.x (new)? > > > > Hello all, > > > > I'm still investigating the (most likely non-kamailio-related) memory leak > of my previous message to the list, there have been no developments so far. > I'll update if anything changes. > > > > This concerns a new finding which seems to affect kamailio 5.5.x. I have > two kamailio instances receiving the same traffic via round-robin. I > upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared > of any issues arising. I was lucky to do so, because with identical > configuration, 5.5.x seems to run out of SHM very quickly. Here are links > to graphs produced by our monitoring system: > > > > Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png > > New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png > > > > The configuration uses mtree, htable, vars and vns extensively. Has anyone > come across anything similar? Let me know if I can provide any further > information to help disect this. Thanks! > > > > BR, > > George >

2 8

Issues running mongodb command from kamailio
by Edo 18 Jan '22

18 Jan '22

Hi All, Please help! I am getting an error when trying to start kamailio after making some changes to the configuration file. Error is : *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 47-57: '('')' expected (function call)* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad expression* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 102: bad command* *atus=255/EXCEPTION* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 103: bad command* *rse error in config file /etc/kamailio/kamailio.cfg, line 878, column 105: bad command* * used inside params of another function: xlog* *rse error in config file /etc/kamailio/kamailio.cfg, line 879, column 28: use of function execution inside params not allowed* *Server.* Changes made is : # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; #Start of change if (mongodb_find_one("mongodbsrv1", “telephone", “cXXX", " { \'did\': \'2XXXXXXXXXXXXXXXXX\' } ", "mgr1")) { xlog("response from mongodb is [[$mongodb(mgr1=>value)]]\n"); } #End of change append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } Please can someone tell me what I am doing wrong. Error says syntax but I have tried many permutations of the suggested syntax with no luck. Please help Thanks -- ----- Ekunwe EDO Network Services, Inc. Tel: 601.497.3932

2 4

REGISTER for MS Teams
by Lucian Romi 18 Jan '22

18 Jan '22

Hi, can register send from ms teams or I have to use Kamailio to send to my PBX.

2 1

JS Kemi TypeError: undefined not callable (property 'ifdef' of [object Object])
by Huseyin Kaan Koyukan 18 Jan '22

18 Jan '22

Hello Everyone, First of all, this is my first ever e-mail to the group so I'd like to thank all the developers who built this great product and thank everyone for the kind and helpful community. I've been using Kamailio for the past 8 months, I've had lots of fun with it. My issue is I'm trying to set-up JS Kemi interpreter for my Kamailio-Asterisk box. I translated the config file to JS kemi, which wasn't too bad but I'm getting a type error for the "KSR.kx.ifdef("WITH_ASTERISK")" statement that I used. Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:1478]: dukopen_KSR(): module 'KSR' has been initialized Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) DEBUG: app_jsdt [app_jsdt_api.c:473]: jsdt_sr_init_child(): loading js script file: /usr/local/etc/kamailio/kamailio-basic-kemi-jsdt.js Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: app_jsdt [app_jsdt_api.c:447]: jsdt_kemi_load_script(): failed running: TypeError: undefined not callable (property 'ifdef' of [object Object]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [core/sr_module.c:865]: init_mod_child(): error while initializing module app_jsdt (/usr/local/lib64/kamailio/modules/app_jsdt.so) (idx: 0 rank: -127 desc: [main]) Jan 17 15:11:26 anayurt kamailio[8261]: 0(8261) ERROR: <core> [main.c:1671]: main_loop(): error in init_child(PROC_INT) -- exiting In syslog I can see that kex.so module is loaded successfully and 'KSR' has been initialized. I don't know what I'm doing wrong. A few sample JS Kemi code that I could find did not use the kx.ifdef function. Was this a new addition to the interpreter? Perhaps my Kamailio version (kamailio 5.5.0 x86_64/linux) does not support it? Thank you all in advance. Kaan.

1 0

Help with rewriting headers for NAT manually
by Chad 17 Jan '22

17 Jan '22

We are looking for some help (possibly a paid consultant) to help us with our Kamailio setup. To keep this as short as possible: we use Kamailio as a NAT proxy to bridge our external IP and our private IP asterisk servers (via dispatcher). However both the external IP and the internal IP that the Kamailio server uses are virtual IPs created by keepalived. Because of that neither mhomed nor fix_nated_contact work, and we use force_send_socket to direct the traffic. We run linux Debian 10 for the OS. Also we do not use a DB at all, everything is done with local config files. The problem is that when traffic goes out the Contact header has a private IP in it, like: Contact: <sip:##########@10.10.10.###]:5060> There are 2 possible solutions to this: 1. Make changes to linux, keepalived and/or Kamailio so that Kamailio recognize the virtual IPs so that mhomed and fix_nated_contact work as usual. 2. Create a manual header rewrite system. If solution #2: What we need to do is create a way to rewrite the contact header to the external IP on the way out, and on the way back rewrite it back to the internal server that the call is already connected to. Not sure if we will need to store those paths on the server or if we can do some kind of cheat with another persistant header like P-Preferred-Identity or P-Asserted-Identity (i.e. store the internal IP in the name field or something). If anyone out there know of a way to do this or wants to give it a try please reach out to me. Thank you all for your time. -- ^C Chad

5 27

Alternate to FoHSS for Kamailio IMS?
by Simon,Benjamin T 17 Jan '22

17 Jan '22

Hello, We are attempting to setup an instance of Kamailio to serve as an IMS for a test setup of 4G/5G NSA with Open5GS. The instructions on the Kamailio wiki and provided by Open5GS currently specify to use FoHSS as the HSS solution for Kamailio IMS. As this HSS solution has not seen significant development in a few years we are reticent to deploy it on our implementation. Is there an alternative open source HSS which can be deployed to serve as the HSS for Kamailio instead of FoHSS, and would there need to be any adjustments to Kamailio to reflect the deployment of this HSS instead of FoHSS? Thank you, Benjamin Simon Research Assistant Florida Institute for Cybersecurity Research, University of Florida Email: bsimon4(a)ufl.edu Disclaimer: Florida has a very broad public records law. As a result, any written communication created or received by the University of Florida will be made available to the public and media, upon request, unless otherwise exempt. Under Florida law, e-mail addresses are public records. If you do not want your e-mail address released in response to a public records request, do not send electronic mail to this address. Instead, contact by phone or in writing.

2 2

RR balance kamailio
by Gokan Atmaca 16 Jan '22

16 Jan '22

Hello I installed kamailio on Ubuntu. The version information is as follows. (0) I added two dispatcher sip servers. (1) When I enter the parameter as below in the configuration file , I get an error as follows. (2) I am getting RR module not available error. Packages installed are "kamailio-berkeley-modules, kamailio-tls-modules, kamailio-presence-modules" How can I solve the problem? (0) version: 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 9.3.0 (1) Dispatcher list: { NRSETS: 1 RECORDS: { SET: { ID: 1 TARGETS: { DEST: { URI: sip:X.X.X.X FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } DEST: { URI: sip:Y.Y.Y.Y FLAGS: AP PRIORITY: 0 LATENCY: { AVG: 0.000000 STD: 0.000000 EST: 0.000000 MAX: 0 TIMEOUT: 0 } } } } } } (2) if ( method=="INVITE" ) { ds_select_dst("1","4"); sl_send_reply("100","Trying"); forward(); exit(); } Err_log: Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded Jan 12 12:15:11 kamailio /usr/sbin/kamailio[21241]: INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available -- ⢀⣴⠾⠻⢶⣦⠀ ⣾⠁⢠⠒⠀⣿⡁ Debian - The universal operating system ⢿⡄⠘⠷⠚⠋⠀ https://www.debian.org ⠈⠳⣄⠀⠀⠀⠀

2 3

Mid-dialog failover without virtual IP
by Muhammad Danish Moosa 14 Jan '22

14 Jan '22

Hi, I have a pair of kamailio stateful proxies and am trying to achieve Active-Passive failover model with DMQ dialog replication. Primary Proxy is working in the middle of 2 SBCs and all nodes are within the same network so NATing is NOT involved ;secondary proxy is accessible through VPN. It's pretty simple configuration, proxy is just adding and removing headers in the middle. So if I stop the kamailio service on primary IP , SBCs blacklist that and start sending new invites to secondary IP. That's all good. But if I reproduce this failover during a connected session , SBCs do not send mid-dialog requests/responses to failover IP. I can see they still try to send messages to primary SBC which obviously end up with timeout. I understand the most natural solution is HA with Virual IP but it's not possible because secondary IP is accessible through VPN. I tried to use DNS SRV record with priorities but that also does not work with mid-dialog messages, I also have tried to change record-route/via header to fqdn through advertise_address. Its not changing record-route but via only but I am still not getting expected results. Any guidance of possible solution(s) here will be appreciated. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

3 2

Massive uploading of aliases through a dbtext
by Denys Pozniak 14 Jan '22

14 Jan '22

Hello! I need help uploading a list of aliases through a dbtext to use them via myself. It is likely that there is some configuration problem: *loadmodule "db_text.so"* *modparam("db_text", "db_mode", 1)modparam("db_text", "emptystring", 1)* *loadmodule "alias_db.so"modparam("alias_db", "db_url", "text:///etc/kamailio/dbtext")* *[root@kamailio]$ cat dbtext/dbaliasesid(int,auto) alias_username(string) alias_domain(string) username(string) domain(string)1::sip.test.local::* *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { }}* If done through a script (alias=sip.test.local), then everything works fine: *[root@kamailio]$ kamcmd core.aliases_list{ myself_callbacks: yes aliases: { alias: { proto: * address: sip.test.local port: * } }}* -- BR, Denys Pozniak

2 4

Retrieve last via IP from SIP MEssage
by Adarsh Chauhan 14 Jan '22

14 Jan '22

Hi All, I have tried couple of methods to retrieve last via IP of SIP message but failed to retrieve it, trying to retrieve the source - @hf_value.via[-1].host, it simply prints the exact text back @hf_value.via[-1].host - $sel((a)hf_value.via[-1]) it doest give the IP but the entire via value SIP/2.0/TCP 172.31.30.144;branch=z9hG4bKf8bd.753341d2000000000000000000000000.0.host @via[-1].host it also simply prints the exact text back @via[-1].host Regards, Adarsh Chauha

2 1

EVAPI workload allocation across processes
by Alex Balashov 13 Jan '22

13 Jan '22

Hi, Given several EVAPI worker processes: modparam("evapi", "workers", 4) # kamcmd -s /run/kamailio/kamailio_ctl ps | fgrep -i evapi 539257 EvAPI Dispatcher 539258 EvAPI Worker 539259 EvAPI Worker 539260 EvAPI Worker 539261 EvAPI Worker How should one understand the way that the EVAPI workload is allocated among them? My best guess is that the EvAPI dispatcher process is the sole accept()or of client connections and the place where all relaying to client connections is done, regardless of their number, and that the ‘workers’ represent an async pool of processes in which TM transactions can be resumed or other work can be done when an outside caller sends a message to the dispatcher. However, the documentation[1] is a little vague on the meaning of the ‘workers’ modparam: "Number of worker processes to be started to handle incoming messages from remote applications.” Or maybe it’s not. Just wanted to confirm that my understanding is correct. Thanks! — Alex [1] https://kamailio.org/docs/modules/5.5.x/modules/evapi.html#evapi.p.workers -- 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/

2 3

Reverse Kamailio Dispatcher order
by Adarsh Chauhan 12 Jan '22

12 Jan '22

Hi All, I was looking for a way where i could reverse the order of priority set on destinations in dispatcher file. for eg: if i have the following file: 1 sip:10.252.168.41:5060;transport=tcp 0 5 # (US) 1 sip:10.252.168.42:5060;transport=tcp 0 4 # (US) 1 sip:10.252.168.43:5060;transport=tcp 0 3 # (EU) 1 sip:10.252.168.44:5060;transport=tcp 0 2 # (EU) I wanted to use top to bottom when in US region and bottom to top when in EU region. One option is to make another set, however that will force Kamailio to send ping (OPTIONS) twice when probing. Regards, Adarsh Chauhan

4 4

Suggestion on receiving AAR
by Jyoti Bansal 10 Jan '22

10 Jan '22

Hello all, We are trying to connect kamailio ims to a pcrf, connection is successful. Whenever we register through bhoghe client , we receive AAR on pcrf from kamailio ims(pcscf), so just need a suggestion can we disable that till we receive invite from the soft phone? Can anyone help on this? Thanks, Jyoti bansal -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Call Preservation Mode in Kamailio
by varun pratapsingh 10 Jan '22

10 Jan '22

Happy New Year 2022 Everyone. Wish you a lot of innovation this year! I was just curious to know if any of us had handled the mid-call failures using the Kamailio. I am writing a SIP Client to handle the failures by using the DNS SRV based failover and Fallback approach where I am using two Kamailio instances as Outbound Proxy resolved by DNS SRV lookup. Everything is fine except for on-going calls. How to handle the mid-call failovers? Is there any way in Kamailio like Call Preservation mode or something? Any suggestions or ideas to handle midcall failover so that if a current Outbound Proxy goes down then all the ongoing calls can be seamlessly routed through another Kamailio (secondary). Thanks and Regards Varun

3 2

Kamailio call flows with Asterisk
by Nauman Sulaiman (SESSIONTALK) 05 Jan '22

05 Jan '22

Hi, We are using Asterisk as a PBX with users directly registered to Asterisk and Asterisk registering to SIP trunks. We are now looking to put Kamailio in front of Asterisk to handle SIP registrations from the SIP clients. In a ‘typical’ architecture should we keep the SIP trunk registrations on Asterisk or is Kamailio used for this? We want to keep Asterisk as the RTP proxy so we don’t want a call setup by Kamailio with RTP then going direct between user agents. Regards Nauman

3 2

Calling t_relay([host, port]) with variables
by Nick J 05 Jan '22

05 Jan '22

Hi all, I've been banging my head on the table with this one and hoping someone has some insight here. I'm trying to use a variable as the host & port input in t_relay() - t_relay(host, port) This works fine if statically set: t_relay("10.0.1.1", "5060"); But if I change the host or the port to a variable like this: $avp(destip) = "10.0.1.1"; $avp(dstport) = "5060"; t_relay($avp(destip), $avp(dstport)); Then Kamailio fails to start, calling out the t_relay line: kamailio: CRITICAL: <core> [core/cfg.y:3686]: yyerror_at(): parse error in config file /etc/kamailio/kamailio-routelogic.cfg, line 62, column 35: bad command I get the same behavior with pseudovariables and vars as well: $var(destip) = "10.0.1.1"; $var(dstport) = "5060"; t_relay($var(destip), $var(dstport)) Any tips or pointers would be much appreciated. Thanks! Nick

4 3

Features of Kamailio as SBC
by Nattha Vudhiprasittipol 04 Jan '22

04 Jan '22

Hello I would like to know all the feature lists of Kamailio as SBC. Could you please send me the document? I cannot find it on the website. Best regards Nattha Vudhiprasittipol Product Development Engineer Creaturelab Network Solution Co.,Ltd 357 Charansanitwong Rd., Bang-Bamru, Bangphlat, Bangkok 10700, THAILAND Email: nattha.v(a)creaturelab.co.th Website: www.creaturelab.co.th

7 19

Recommendation for compatible IMS soft phones
by Ashay Matekar 04 Jan '22

04 Jan '22

Hi Team, We were trying to test a call flow with Kamailio IMS that requires preconditio. Do you know any free or paid IMS softphones that support preconditio and are compatible with Kamailio IMS. -- *Thanks and Regards,* *Ashay Matekar* Associate Manager Engineering | +91 932446823 Great Software Laboratory | www.gslab.com LinkedIn <https://www.linkedin.com/company/gs-lab> | Twitter <https://twitter.com/_gslab> | Email Disclaimer <https://www.gslab.com/privacy-policy/#email-disclaimer> -- Confidentiality Notice and Disclaimer: This email (including any attachments) contains information that may be confidential, privileged and/or copyrighted. If you are not the intended recipient, please notify the sender immediately and destroy this email. Any unauthorized use of the contents of this email in any manner whatsoever, is strictly prohibited. If improper activity is suspected, all available information may be used by the sender for possible disciplinary action, prosecution, civil claim or any remedy or lawful purpose. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, lost, arrive late, or contain viruses. The sender is not liable whatsoever for damage resulting from the opening of this message and/or the use of the information contained in this message and/or attachments. Expressions in this email cannot be treated as opined by the sender company management – they are solely expressed by the sender unless authorized.

2 1

Catching Register 200 OK, then using it to validate other packets.
by Richard Edmands 04 Jan '22

04 Jan '22

Hey Kamailio Users, I've written a TLS-UDP gateway. It works well, but I want to reduce the internet noise it pushes through. I have an idea on what I'd like to try. Wondering if it would be possible to catch the Register 200 OK, store it and then when receiving packets it future to then validate them against the data? I've managed to capture a 200 OK, but the save("location") function says it will return it's own 200 OK which isn't the plan since we just want to capture the location. Anyone done something similar and got some working example code? Is this a bad idea for some reason?

2 2

Re: [SR-Users] Kamailio docker image run in Kubernates
by Henning Westerholt 27 Dec '21

27 Dec '21

Hello, (please keep the list in CC) Can you pull the v2 image on the command line with the CLI commands from your repository? Do you have Maybe its some naming or access rights issue. You need to try out different things in isolating the cause, it’s difficult to guess. Have a look e.g. here for more debug options for this error https://stackoverflow.com/questions/34848422/how-can-i-debug-imagepullbacko… Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Maryam Baghdadi <maryam.baghdadi(a)gmail.com> Sent: Monday, December 27, 2021 10:15 PM To: Henning Westerholt <hw(a)gilawa.com> Subject: Re: [SR-Users] Kamailio docker image run in Kubernates I dedicate a service and demonset.apps in yaml file. When I assign a label to a node of k8 and then select node according to node label in yaml file for first image for example image:v1 every thing is OK and pulling done successfully . But for second time when I rebuild docker image after changing kamailio configuration for example image:v2 generated , pulling error happens… On Tue, Dec 28, 2021, 12:34 AM Henning Westerholt <hw(a)gilawa.com<mailto:hw@gilawa.com>> wrote: Hello, as mentioned in the issue tracker, the error seems to be related to a failure of k8s pulling the second docker image. Have you tried to debug the second services locally, for example with docker-compose or some local k8s distribution? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Maryam Baghdadi Sent: Monday, December 27, 2021 9:44 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] Kamailio docker image run in Kubernates Hello everyone, I have a problem over running local docker images of Kamailio in Kubernate cluster.. Consider I build a local docker image from Kamailio and create a pod in one of my nodes . My process created successfully and Kamailio listen over port 5060 now I need to rebuild another docker image with new Kamailio configuration and run it as another pod with new name and listening over port 5061 but I have error "ImagePullBackOff" . I test this scenario many times but problem exists.. It is devastating for me and I need emergency help.

1 0

Kamailio docker image run in Kubernates
by Maryam Baghdadi 27 Dec '21

27 Dec '21

Hello everyone, I have a problem over running local docker images of Kamailio in Kubernate cluster.. Consider I build a local docker image from Kamailio and create a pod in one of my nodes . My process created successfully and Kamailio listen over port 5060 now I need to rebuild another docker image with new Kamailio configuration and run it as another pod with new name and listening over port 5061 but I have error "ImagePullBackOff" . I test this scenario many times but problem exists.. It is devastating for me and I need emergency help.

2 1

Kamailio Stop processing SIP traffic
by pwerspire 27 Dec '21

27 Dec '21

Hi, I have been doing some load testing with Kamailio and having some issues. I have been trying with 100 CPS and at the beginning, everything is working well but after some time (it can be 10 minutes or for example 40 minutes, is just random but normally happens more in the first 10 minutes) Kamailio stop replying to all SIP messages but still processing HTTP requests, some commands, etc. If I use "kamcmd dlg.list" no output happens and after that, all kamcmd commands just keep loading with no output. Kamailio is sharing Dialogs and htables using DMQ with other Kamailio that is the failover, if I shut down the failover one (so no DMQ replication), the test works well with the 100 CPS. Also tried with modparam("dmq", "worker_usleep", 1000) but the behavior is the same, it will stop processing traffic. This is some of configuration used: # ----------------- setting module-specific parameters --------------- modparam("dmq", "server_address", "sip:INTERNAL_INSTANCE_IP:5060") modparam("dmq", "notification_address", "DMQ_NOTIFICATION_ADDRESS") modparam("dmq", "multi_notify", 1) modparam("dmq", "ping_interval", 10) modparam("dmq", "num_workers",4) modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) # ----- dialog params ----- modparam("dialog", "dlg_flag", FLD_DLG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", DBURL_RW) modparam("dialog", "db_mode", 0) modparam("dialog", "enable_dmq", 1) modparam("dialog", "db_update_period", 10) modparam("dialog", "h_id_start", H_ID_START) modparam("dialog", "h_id_step", H_ID_STEP) Memory: Shared: 4096 Private: 512 kamailio -v version: kamailio 5.5.3 (x86_64/linux) 473cef 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: 473cef compiled on 09:34:57 Dec 20 2021 with gcc 10.2.1 In attach is the trap collected after the issue happens. Any more logs or configurations that can help identify or solve the issue? Thanks for the help, Regards, Tiago

6 11

How to change language in siremis web interface
by mohammad baghdadi 27 Dec '21

27 Dec '21

Hello I’m new in siremis as web interface of kamailio and I immediately need your help,I installed kamailio and siremis and login in siremis web interface,I can work with it but now I have a problem. I want to change menu language there is an option in(setting->translate->manage languages) add my desired language and click save and i see this error: Error message: #2, count(): Parameter must be an array or an object that implements Countable Script name and line number of error: [/var/www/siremis-5.3.x/siremis/modules/translation/lib/LangPackCreator.php I needed to change the language and thanks for your the help to me.

2 1

MongoDB and SIREMIS
by Tihomir Culjaga 27 Dec '21

27 Dec '21

hi, I'm running kamailio 5.4 using MongoDB as a backend database. Can I run SIREMIS on that database as well? Regards, Tihomir.

2 1

Merry Christmas and Happy Holidays 2021-2022!
by Daniel-Constantin Mierla 25 Dec '21

25 Dec '21

Another challenging year is getting close to its end, pressuring even more everyone! Let's hope things will go back to a more usual life (as it used to be) in the near future! Kamailio project continued to develop, bringing out a new major release, preparing for the next one in a few months, relying on an amazing community of developers and users. We also trust it continued to help people and companies world wide to connect with the dear ones and build a better way of living. The season holidays are ahead, time to relax, spend time with family and friends! I take again this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2021 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

repository not present
by Jyoti bansal 23 Dec '21

23 Dec '21

2 1

Active/active keepalived - Kamailio includes non-local virtual IP in "myself" which breaks DMQ call routing
by Rhys Hanrahan 22 Dec '21

22 Dec '21

Hi Everyone, I have Kamailio acting as a registrar, using DMQ to replicate usrloc between nodes. I’m relaying INVITEs to the home registrar, but skipping this if (uri == myself) is true. In my lab I just setup my SBCs in an active/active mode, similar to the link below. So I have 2x VIPs and both SBCs listen on both, but the first VIP is only master on SBC01 and the second VIP is only master on SBC02. https://docs.nginx.com/nginx/admin-guide/high-availability/ha-keepalived-no… My issue is that now “myself” will always return true even if the virtual IP is actually not active on this node at the moment, so the invite is never relayed properly if it’s on another node. I guess “myself” is just blindly looking at the list of Ips that are configured with either listen= or alias= and doesn’t care if the IP actually exists on the server at the time, or not? I am thinking I may have to ditch relying on “myself” entirely and come up with some other way of checking for local requests, but I am wondering if there’s a good way of approaching this? I was thinking of perhaps triggering an add/remove of listen directives (or something) when a failover of a VIP happens, though this seems kind of heavy handed. Hoping there is a better way. Has anyone else tried doing active/active Kamailio instances like this? Thanks! Rhys Hanrahan | Chief Information Officer e: rhys(a)nexusone.com.au<mailto:rhys@nexusone.com.au> [www.nexusone.com.au]<http://www.nexusone.com.au/> [signature_132622389] <http://www.fusiontech.com.au/> NEXUS ONE | FUSION TECHNOLOGY SOLUTIONS p: 1800 NEXUS1 (1800 639 871) or 1800 565 845 | a: Suite 12.03 Level 12, 227 Elizabeth Street, Sydney NSW 2000 www.nexusone.com.au<http://www.nexusone.com.au/> | www.fusiontech.com.au<http://www.fusiontech.com.au/> The information in this email and any accompanying attachments may contain; a. Confidential information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; b. Legally privileged information of Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties; and or c. Copyright material Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd or third parties. If you have received this email in error, please notify the sender immediately and delete this message. Fusion Technology Solutions Pty Ltd, Nexus One Pty Ltd does not accept any responsibility for loss or damage arising from the use or distribution of this email. Please consider the environment before printing this email.

3 6

Issue Bug in SEMS (transparent SBC profile)
by Mojtaba 21 Dec '21

21 Dec '21

Hello, Why in SEMS, some changes (in transparent SBC profile) in sip headers do not apply. For example i want to change CSeq or Contact header like this: # transparent SBC profile # defaults: transparent RURI=$r From=$f To=$t Call-ID=$ci_leg2 CSeq=$H(CSeq) Contact=$H(Contact) When I run SEMS, CSeq and Contact do not change, while other headers change. [#7fc688d7b1c0/9379] [onLoad, SBC.cpp:189] INFO: loading SBC call profiles from '/usr/local/etc/sems/etc/' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:418] INFO: SBC: loaded SBC profile 'mo' - MD5: d6a2c29084043edde1357231f6866737 [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:423] INFO: SBC: RURI = '$r' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:424] INFO: SBC: RURI-host = '' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:425] INFO: SBC: From = '$f' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:426] INFO: SBC: To = '$t' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:431] INFO: SBC: Call-ID = '$ci_leg2222' [#7fc688d7b1c0/9379] [readFromConfiguration, SBCCallProfile.cpp:434] INFO: SBC: force outbound proxy: yes This issue caused the Asterisk could not pass the Authentication process in "match_req_to_dialog" function because the CSeq is the same as the dialogs. So the Asterisk thought this message was a LOOP message. Has anybody encountered this issue in SEMS? How did you solve this? I know there are some ways for doing this with Kamailio, But i focused on issues in SEMS With Best Regards *-- Mojtaba Esfandiari.S* *-- PhD student and Research Affiliate, * *-- Technical Manager at IP-PBX Laboratory, Ferdowsi University of Mashhad, Iran.* -- Address: IP-PBX Lab., Engineering Faculty, Ferdowsi University Main Campus, Mashhad, Iran. -- Tel: +98-51-38763635 -- Mobile: +98-915-117-6713

4 14

uac module - use htable instead of database
by Markus Bönke 20 Dec '21

20 Dec '21

Hello all, we want to use the "remote registrations feature" of the uac module to let kamailio register with other sip providers. As far as I see the module works only with a database backend. As our kamailio script should be independent from a database, a solution could be to load the configuration on startup (and refresh it periodically) from a database view into a htable and then trigger the remote registrations based on the expiration of the htable columns. Could something like this be implemented by using the uac module or by an other workaround ? Or does it require some enhancements in the uac module ? Thanks and regards Markus

2 2

Using kamcmd tm.t_uac_start to send SIP REGISTER messages in Kamailio 5.5
by Steve Sether 20 Dec '21

20 Dec '21

I'm trying to use the kamcmd tml.tc_uac_start command to send raw SIP messages in Kamailio 5.5, but I always get a error: 400 - Invalid headers. I've also tried sending OPTIONS messages with the same result. I've looked for examples of sending SIP messages via kamcmd, but can't find any. kamcmd help tm.t_uac_start specifies the following: kamcmd> help tm.t_uac_start starts a tm uac using a list of string parameters: method, ruri, dst_uri, send_sock, headers (CRLF separated) and body (optional) Essentially my problem is I don't really know how to properly encode these headers. How do I send special characters? Do I need to use double quotes, single quotes, etc around the whole header? Which characters are special? How do I encode the CRLF? I tried \r\n, but I still get this error. It says only From: and To: are required. It's a bit hard trying to make progress when all I ever get is one error message. The kamailio logs are a little more helpful, and largely point me down the road that it doesn't understand the CRLF encoding, and says my From: header includes the To: line, so it's not separating them. For example, I'm trying things like this: kamcmd -v tm.t_uac_start OPTIONS sip:200@exampleserver.com:5060 . . "From: <sip:example@192.168.0.1>;tag=d27ca05e-1789-4774-a698-dcfe91dec9b4\r\nTo: <sip:example@192.168.0.2>"\r\nExpires: 1200\r\nContent-Length: 0" Nothing seems to work, so whatever the syntax I'm using must be wrong. Just providing an example of sending SIP via kamcmd tm.t_uac_start would likely help a huge amount. Thanks! --

2 1

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