- sr-users - kamailio.org

Homer7 and Kamailio 5.3
by Angelo Sipper 03 Feb '22

03 Feb '22

Hi to all, We have a Kamailio 5.3 that sends all traffic to a homer7 using siptrace module with parameters force_send_sock and hep_mode_on,1 and hep_version,3 and trace_mode,1. Everything works fine getting all traffic to Homer7 but, in case Homer7 is not reachable, Kamailio then no longer accepts traffic or registrations and it is like it crashes. When we revive Homer7 and is it again reachable accepting traffic, Kamailio then works again fine without any restart or actions. Any ideas why this may be happening? Is there anything we need to add to the config to avoid this behavior as having Homer7 now is a single point of failure for Kamailio? Thank you! Kind Regards, Angelo

6 11

uac problem with 5.3.8
by Richard Robson 03 Feb '22

03 Feb '22

we have been running 5.3.4 with the uac_auth_mode() function ans it has been working we have upgraded to 5.3.8 and with the same script we are getting an error with the uac module saying it was unable to find the cmd auc_auth_mode I have tried this with uac_auth as well with the same result. it looks like the uac module is not loaded or is not loading but running a core.modules reports the uac module can anyone shed any light on this problem? is it a version pecific problem?

3 4

Updating $du for uac REGISTERs in event_route[tm:local-request]
by Raffaella Seuß 03 Feb '22

03 Feb '22

Hello all, I tried to update $du for REGISTERs sent by the uac module in event_route[tm:local-request]. (I want to overwrite the auth_proxy under some conditions.) However, the requests are still sent to the auth_proxy defined in uacreg. Am I missing something here, why doesn't it work? It's basically just: event_route[tm:local-request] { if (is_method("REGISTER")) { # ... $du = "sip:PROXY_IP:PROXY_PORT"; } } My Kamailio version is 5.5.2. With kind regards, Raffaella Seuß foncloud GmbH & Co KG Hahlweg 2a 36093 Künzell Tel: / Fax: +49 661 968990-99 Email: raffaella.seuss(a)foncloud.net Web: www.foncloud.net P.S.: Wussten Sie schon? Unter https://www.foncloud.net/wissen finden Sie zahlreiche Informationen und hilfreiche Artikel rund um unsere Produkte und Services. Registergericht: Amtsgericht Fulda, Persönlich haftende Gesellschafterin der foncloud GmbH&Co.KG: Global Brain Network GmbHGeschäftsführer der Global Brain Network GmbH: Peter Krug Sitz der Gesellschaft: Künzell. Diese E-Mail enthält vertrauliche und/oder rechtlich geschützte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet. This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorised copying, disclosure or distribution of the material in this e-mail is strictly forbidden.

1 0

Siremis configuration
by Tim Mastarone 02 Feb '22

02 Feb '22

Hi, I am using Siremis v 5.3 with kamailio. My users can log into the Siremis site just fine but the field to enter their email address is not appearing. My sql queries seem to be working besides that. Does this feature need to be enabled? I would like to use email to confirm the user login with a confirmation code. (mysql Ver 15.1 Distrib 10.1.48-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2) Thank you, [cid:image001.png@01D81830.52C21EA0]<http://www.vikingelectronics.com/> TIM MASTARONE Associate Engineer 715-386-8861 x208 vikingelectronics.com<http://www.vikingelectronics.com/>

1 0

two SIP proxies
by Uwe Sigleur 31 Jan '22

31 Jan '22

2 1

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

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20 Jan '22

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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

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