- sr-users - kamailio.org

CDR preserve original call direction
by Ahmed Fouad 10 Feb '22

10 Feb '22

Hi, I've enabled ACC and CDR logging and it's working fine for the logging but I noticed something wrong in the CDR table , the call source and destinations are depending on the BYE message not the initial invite dialog ACC table +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | id | method | callid | time | src_ip | src_domain | src_user | dst_ouser | dst_domain | +-----+--------+--------------------------------------+---------------------+---------------+---------------------------+--------------+--------------+--------------------------------------+ | 216 | BYE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:57:51 | x.x.x.x | domain.com | DST_CLID | SRC_CLID | domain.com | | 215 | INVITE | 81fa79a0-3deb-47b6-80ab-831a78a5d46e | 2022-02-09 06:55:17 | x.x.x.x | domain.com | SRC_CLID | DST_CLID | dst_domain.com | The INVITE is the correct one with SRCIP x.x.x.x and domain.com but the BYE reversed the order since the the DST_CLID who sent the BYE so it's now logged as the source and the final CDR log looks like +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | cdr_id | src_username | src_domain | dst_username | dst_domain | call_start_time | call_end_time +--------+--------------+---------------------------+--------------+--------------------------------------+---------------------+---------------------+----------+-----+-------+--------+-------+---------+ | 195 | DST_CLID | domain.com | SRC_CLID | domain.com | 2022-02-09 06:55:12 | 2022-02-09 06:57:51 | the CDR as shown is reversed regarding the source/dst username I tried to enable the direction ACC direction flag but didn't help and here is my current paramters modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "cdrs") Can you please advise how to configure ACC/CDR to log the CDR with the correct initial INVITE call flow regardless of the BYE direction ?

3 5

REGISTER with http_async_client ?
by Cyril Ramière 10 Feb '22

10 Feb '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 $http* values. Do you see something that I am doing wrong or missing in my logic? Is pausing/resuming to use the async http client is allowed if I'm handling a REGISTER transaction? Here's a simplified version of my routing block (not far from reality): ##### SNIP request_route{ route(AUTH); route[AUTH]{ if (is_method("REGISTER"){ if(no_auth_header){ www_challenge("$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"); } } ##### END SNIP Best regards!

4 15

Kamailio doc request
by Mustafa Rahman 10 Feb '22

10 Feb '22

2 1

Failure route with t_relay()
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I am trying to make a re-transmission mechanism from the failure route, I've tried to use $dp and $du as a relay, but variables are not accepted in t_relay(), When using ip and port straight forward t_relay("144.0.0.0", "1111"), I am getting: [t_fwd.c:1752]: t_forward_nonack(): no branches for forwarding [tm.c:1753]: _w_t_relay_to(): t_forward_noack failed Is there a way to re-transmit the call to the same carrier again? Thanks in advance, and have a great day, Max Kemel

2 1

Failure route mechanism with counter
by Max Kemel 08 Feb '22

08 Feb '22

Hey, I would like to use a failure route with a retry mechanism, which provides a MAX_NUM_OF_RETRIES (max number of times reaching the failure route), Which can be achieved by a simple global counter, from what I've seen, the Counters module doesn't provide a GET function in script, What is the easiest solution for this kind of behavior? Thanks a lot, Max Kemel

2 1

Timers firing too soon
by Alberto Izquierdo 04 Feb '22

04 Feb '22

Hi all, I'm using Kamailio timers to execute a series of calls with a certain delay. I have found out that from time to time, the timers are triggered too soon, sometimes they are triggered 2/3 seconds before they should. I have read in the documentation that it is normal to have a bit of delay in some of the timers, but it doesn't mention anything about this behavior. Is this normal? Thank you.

1 0

Debian kamailio.init vs. kamailio.service
by Juha Heinanen 04 Feb '22

04 Feb '22

By comparing pkg/kamailio/deb/bullseye dir kamailio.init and kamailio.service files, it appears to me that they are not equal alternatives. Is this correct interpretation? -- Juha

1 0

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 ⠈⠳⣄⠀⠀⠀⠀

2 3

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

tcp_set_connection_lifetime(1) does not close connection
by Sebastian Damm 19 Dec '21

19 Dec '21

Hi, for having a maintenance mode, I need to implement some sort of draining functionality. All our clients register via TLS. So I thought, I just answer the next REGISTER of a client with a 503 and kill the tcp connection afterwards, so they need to open up a new connection to the loadbalancer in front and end up on a different server. This is my code: 510 if msg.is_method("REGISTER"): 1 if KSR.pv.get("$shv(MAINTENANCE_MODE)") == 1: 2 KSR.info("In maintenance mode, rejecting registration") 3 KSR.hdr.append_to_reply("Retry-After: 2\r\n") 4 KSR.sl.sl_send_reply(503, "Draining") 5 if KSR.tcpops.tcp_set_connection_lifetime(1) > 0: 6 KSR.info("Set Connection lifetime to 1") 7 return False (Side note: msg is some internal class, this is obviously KEMI with python) So now when I set the MAINTENANCE_MODE var to 1, Kamailio rejects the REGISTER requests, logs both lines and sends out a 503. Debug log tells me this: Dec 10 15:32:10 ifens5 /usr/sbin/kamailio[3851]: DEBUG: tcpops [tcpops.c:229]: tcpops_set_connection_lifetime(): new connection lifetime for conid=1: 908866403 Dec 10 15:32:21 ifens5 /usr/sbin/kamailio[3855]: DEBUG: <core> [core/tcp_main.c:4554]: tcpconn_main_timeout(): entering timer for 0x7f759cb26218 (ticks=908866499, timeout=908866499 (0 s), wr_timeout=0 (211631299 s)), write queue: 0 bytes In tcpdump I can see that the connection does not get closed immediately. Is there some timer in the tcp stack which cleans up expired connections from time to time? Can I manipulate how often this timer runs? I couldn't find anything, and clearly even though I tell Kamailio to close the connection after 1 second, it doesn't do it. Thanks for any input. Regards, Sebastian

3 3

Kamailio SIP Application server
by Weiqi Yao 17 Dec '21

17 Dec '21

Hello community, We were testing using Kmailio work as an IMS, and then turn out the phone wants to do MMTel for mobile real-time multimedia communication. We are not familiar with this topic, but anyone know Kamailio have such module or any other MMTel service can work with Kamailio? BR, Weiqi

2 2

RPC command dlg.profile_get_size with per_value response
by David VILLAUME 17 Dec '21

17 Dec '21

Hello, I would like to suggest you an implementation of the rpc command dlg.profile_get_size (on a valued profile) that would return a list of values and the number of current dialogs for any of them . Din’t know where to suggest it so I sent it there, please accept my apologies if it’s not the good place. Regards, David

2 2

SRTP/DTLS + RTP Header encryption
by Rahul mm 15 Dec '21

15 Dec '21

Hi guys, Is it possible to achieve RTP header encryption along with payload in SRTP/DTLS ? So far what I've been able to see is RTP payload encryption. I'm using kamailo 5.5.0 and rtpengine 9.2.1.6 in webRTC scenario. Many thanks. BR, Rahul

2 3

Next online devel meeting - Wed, Dec 15, 2021, 15:00UTC
by Daniel-Constantin Mierla 15 Dec '21

15 Dec '21

Hello, 2021 is almost over, time ran fast somehow this year, with a major release in the middle, which delayed the development discussion for the next one, so it's time for it! As usual, the aim is to sketch the roadmap to next major release so the developers and community members can syncronize and plan what should be done till Kamailio v5.6 will be out as well as discuss about current state of the project, look for new ideas to improve collaboration within the community, a.s.o. The meeting will be done on a matrix chat room. More details are in the wiki page of the event, available at: * https://www.kamailio.org/wiki/devel/irc-meetings/2021a Feel free to add there topics that you want to be discussed. The proposed date is next Wednesday, December 15, 2021, at 15:00UTC (15:00 London/Dublin, 16:00 most of West Europe, 10:00 New York), but other dates can be proposed if they suit for more participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev

1 2

ACK is sent to wrong address when using serial forking
by Minh Phan 15 Dec '21

15 Dec '21

Dear List, As shown in the image below, Proxy 1 tries to do serial forking to Proxy 3 and then Proxy 4. Upon reception of the 480 answer ( blue frame) from Proxy 2, the Proxy 1 generated an ACK (red frame) and sent it to Proxy 1. The problem I have is that the ACK generated by Proxy 1 has a wrong RURI. I expect the ACK to be sent to the URI of the CONTACT header of the 480 message (blue frame). However, its RURI is the same as the one in the first INVITE of the dialog. Note that there is NO ROUTE header in the ACK message. As there is serial forking here, the ACK is generated by the Proxy 1 itself. Any hint would be appreciated, Best Regards, Minh Phan [image: image.png]

3 2

RTP Proxy with Local IP and Public IP
by Vijo Jose 13 Dec '21

13 Dec '21

Hai Everyone. I have configured the RTP Proxy with Kamailio SIP Proxy Server. Where I have my Kamailio is configured with private IP 10.0.0.1 and Public UP 103.XX.XX.XX and the SIP user is connecting with RTP proxy with local IP 10.0.0.1 to register. I also have a SIP Asterisk server on the public Internet with IP 90.XX.XX.XX that makes the call to sipuser@103.XX.XX.XX:5060 which is coming on the sip proxy server and the sip proxy server forwards those packets to the SIP user who is registered with local IP 10.0.0.5. Once the call is answered the RTP is started to send from the SIP server to the SIP Proxy server over the public IP from 90.XX.XX.XX to 103.XX.XX.XX which is working as expected. But from the sip user 10.0.0.5 the RTP is flowing to Proxy server public IP 103.XX.XX.XX instead of RTP Proxy server Localip 10.0.0.1. How can we configure the same? Presently I have started my RTP proxy server with command /usr/local/bin/rtpproxy -F -A 103.XX.XXXX -l 10.0.0.1 -m 10000 -M 20000 -s udp:127.0.0.1:7722 -d DBUG:LOG_LOCAL5 Thanks in advance.

3 2

Re: [SR-Users] Manipulation of request-line
by Mihovil Kolaric 12 Dec '21

12 Dec '21

Hi, unfortunately, Alex is right: manipulations ot the request-URI do not seem to re-write the whole line, the malformed line-ending (SIP/2.0<blank><CR><LF>) remains in the forwarded message.

2 2

remove rport in INVITE Via HDR and make kamailio behave asymmetric
by Nuno Miguel Reis 11 Dec '21

11 Dec '21

Hi guys. I've a scenario here where I'm receiving inbound traffic from a Mitel PBX on a LAN environment with a bunch of router hops between kamailio and the Mitel. Sometimes, not always, I get traffic from the MItel PBX where the source port is changed in transit (some kind of NAT, still don't know where exactly) from 5060 to a dynamic port different from 5060. Every INVITE comes with rport in via header as: .. Via: SIP/2.0/UDP 10.21.252.35:5060 ;branch=z9hG4bK-524287-1---d6bac01809527627*;rport* .. when this happens kamailio sends all the replies to the $si:$sp symmetrically (force_rport behavior) and I would like to force the opposite instead, making it asymmetric and sending the traffic to $si:5060 everytime. I'm having a hard time forcing the *unforce*_rport() behavior. Does anyone have a good suggestion for this? Looking forward to hear from you. Cheers, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

1 1

Unquoted values from #!defenv
by Ben Kaufman 09 Dec '21

09 Dec '21

Hello all, Looking for some clarification regarding quoting and !#defenv . I understand that the example below fails because the environmental variable HOME is not quoted, thus when used in the xlog() call it fails. How is it possible to evaluate and/or use the value of the environmental variable if defined as a preprocessor variable? Looking at the documentation the reason for this directive is "It is a simplified alternative of using #!substdef with $env(NAME) in the replacement part." But #!substdef would allow setting the value inside of a quoted string, so does it really meet that requirement? I'm aware that I could use $env(HOME) rather than a preprocessor directive of HOME. I'm just trying to understand the usage of the #!defenv feature. #!KAMAILIO ## Tested in 5.5.2 loadmodule "xlog" loadmodule "pv" loadmodule "evrexec" modparam("evrexec", "exec", "name=evrexec:timer;wait=1000;workers=1;") #!defenv HOME request_route{ forward(); } event_route[evrexec:timer] { xlog("L_N", "HOME: " + HOME + "\n"); } Ben Kaufman

3 15

wireshark decode kamailio / rtpengine ng messages
by Karsten Horsmann 08 Dec '21

08 Dec '21

Hi List, maybe someone wrote a wireshark helper (didnt know the right term) to view kamailio to rtpengine communication in human form? Some hints would be great. -- Mit freundlichen Grüßen *Karsten Horsmann*

2 1

Re: [SR-Users] wireshark decode kamailio / rtpengine ng messages
by Alex Balashov 08 Dec '21

08 Dec '21

That sounds far too niche for a helper, though, then again, Wireshark has some phenomenally exotic helpers. — Sent from mobile, with due apologies for brevity and errors. > On Dec 8, 2021, at 8:14 AM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

1 0

Kamailio v5.3.9 Released
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, Kamailio SIP Server v5.3.9 stable release is out. This is a maintenance release of the old stable branch, 5.3, that includes fixes since the release of v5.3.8. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.9 or even better to v5.4.x or 5.5.x series. For more details about version 5.3.9 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/12/kamailio-v5-3-9-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.3 is an old stable branch, v5.3.9 being the last planned release out of it, to mark the end of official maintenance. The maintained stable branches are now 5.4 and 5.5, at this time with v5.5.3 being the latest stable release. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Re: [SR-Users] Manipulation of request-line
by Alex Balashov 08 Dec '21

08 Dec '21

That will manipulate the request URI, but not the request line. — Sent from mobile, with due apologies for brevity and errors. > On Dec 7, 2021, at 4:16 PM, Karsten Horsmann <khorsmann(a)gmail.com> wrote: >

2 1

Releasing Kamailio 5.3.9 - last in 5.3.x series
by Daniel-Constantin Mierla 08 Dec '21

08 Dec '21

Hello, it is time to mark the end of official packaging for 5.3.x releases, therefore during the next days I am going to cut the v5.3.9 release from branch 5.3, likely on Wednesday, Dec 8, 2021. In other words, if no major regression is detected shortly after this release, the v5.3.9 is going to be the last one from the 5.3 branch. It is strongly recommended to move to a series that is under active maintenance, respectively 5.5.x or 5.4.x. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Feb 21-24, 2022 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 1

HTTP Requests for API routing - Question on which module to use with KSR
by Emilio Panighetti 08 Dec '21

08 Dec '21

Looking into using Kamailio to route calls via an HTTP API. Basic scenario is this: INVITE comes in (no registration). And Kamailio sends an HTTP request to a server with data extracted from the INVITE. Based on the response, call KSR.setdsturi() and other actions like normalizing phone numbers, etc. (Not sure if changing Contact: headers would be a better option). For now I want to be able to make a simple call routing as a proof of concept. Planning on using Lua as a KEMI interpreter. Looking at different modules: http_client http_client_async xhttp Which module would be more appropriate? http_client_async provides a simple async method and callback, so the SIP threads can continue processing requests while the API server is preparing the response. But the call has to be parked (How is this done?) and this is also an overhead. http_client seems straightforward, but the SIP threads are tied up waiting for HTTP responses. Also looking at the KEMI documentation, the KSR function calls for HTTP_CLIENT doesn’t seem to match the config function documentation. For example, In KEMI manual: KSR.http_client.curl_connect_post() doesn’t have a corresponding in kamailio.cfg. So which would be the preferred approach? Also examples would be appreciated. Thanks in advance!

1 0

Manipulation of request-line
by Mihovil Kolaric 07 Dec '21

07 Dec '21

Hi, I have an issue with a SIP-client, that occasionally puts a blank between the version SIP/2.0 and the CRLF at the and of the request-line. (i.e. sends "ACK <uri> SIP/2.0 \r\n" instead of "ACK <uri> SIP/2.0\r\n"). Kamailio (as well as most SIP UAs) does not care about that error and successfully processes/forwards the request (also when using sanity_check()). But I also deal with a client that fails to parse that request. (Furthermore, wireshark's dissector fails to recognize such malformed messages as SIP, which makes debugging even harder). I tried to fix this issue using textops, but all functions I tried operate only on headers or message body, but do not seem to affect the request line. Is there any possibility to modify the request-line, or to replace it with a new one? Best Regards, Mihovil

2 1

Announcing public beta release of TeamsPhone - powered by Kamailio
by Sergiu Pojoga 06 Dec '21

06 Dec '21

Hello Kamailio-nians, Just to bug everyone quickly: today I'm happy to announce the public beta release of *TeamsPhone <https://teamsphone.net>*, a MS Teams <=> PBX integration solution for ITSPs, IP-PBX service providers and integrators, *powered by Kamailio*. Unlike other similar services offered as SaaS pay-per-user model, our product is oriented for ITSPs to be self-hosted and fully integrated into one's telecom network, providing an add-on service to supplement any Hosted PBX offering with seamless calling using Microsoft Teams *as-as-softphone*, or as we like to call it - TeamsPhone. Some insights about the platform: - 100% automation of MS Teams Direct Routing provisioning - multi-tenant application - API-driven - Microservices oriented architecture - RabbitMQ-powered Message Broker distribution - SIP server (you guessed it - Kamailio, of course!) Seeing how it works is worth a thousand words. A cloud *demo* instance is available at https://my.teamsphone.net for anyone to try it out, use it or abuse it. If you have questions about the product - feel free to contact me in private. Respectfully, Sergiu Pojoga Founder, teamsphone.net sergiu(a)teamsphone.net

1 0

detecting compact headers?
by Brooks Bridges 06 Dec '21

06 Dec '21

We have an internal component of our callflow that does not support compact headers (vendor says it's on the roadmap but no eta, etc) so I'm trying to find a clean way to detect if a message has compact headers and reject it. I haven't had much luck in finding a way to identify specifically if any of the headers of the message are compact however. Can anyone point me in a direction please? Thanks! Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 3

ENV Values in cfg
by H Yavari 06 Dec '21

06 Dec '21

Hi to all, Is it possible to set modules parameters with env values? I know that in routes we can use $env(NAME), but what about for other parameters? for example setting DB address in K8s cluster ENV values. Regards,Hossein

2 1

DMQ Usrloc/Dialog - Experiences
by Carsten Bock 03 Dec '21

03 Dec '21

Hi, I wanted to quickly ask the group about experiences with DMQ and Usrloc/Dialog. I don't expect any issues when choosing such an approach, especially since we would have re-transmits for a message (fr_timer = 30 seconds) if we did not receive any answers or scenarios alike. So I assume, having short "outages" (less than 30 seconds) for the communication is just fine. So far I used DMQ in a non-IMS setup for synchronizing usrloc with ~15k Subscribers and it worked really well. One concern brought up (not by me) was, that synchronizing usrloc with DMQ might not be reliable at scale. I would also love to understand how to treat situations when the communication is lost for a longer period of time (e.g. if the communication between nodes is lost for 5 minutes?). Can anyone share some real-life experiences? Thanks, Carsten -- Carsten Bock I CTO & Founder ng-voice GmbH Trostbrücke 1 I 20457 Hamburg I Germany T +49 179 2021244 I www.ng-voice.com Registry Office at Local Court Hamburg, HRB 120189 Managing Directors: Dr. David Bachmann, Carsten Bock

3 5

topos xavu_field_contact_host kamailio 5.5
by Jonathan Hunter 03 Dec '21

03 Dec '21

Hi All, I was wondering if anyone can help, and advise me if I am missing something when using topos in an environment where I have more than one interface for kamailio. Essentially I would like to set the contact_host to 2 different values depending on the flow of the SIP requests, as we have a private LAN interconnect for example 10.10.10.10, and a public interconnect using 2.2.2.2 (for example). I notice there is xavu for A-side contact and B-side contact, but what about just the contact host? As I have set for example; modparam("topos", "xavu_cfg", "_tps_") modparam("topos", "xavu_field_contact_host", "contact_host") ... $xavu(_tps_=>contact_host) = "2.2.2.2"; And this works fine but obviously sets the contact_host to 8.8.8.8 across both interfaces, I ideally want it to change between 10.10.10.10 and 2.2.2.2. How can I set more than one contact_host $xavu that topos can use? As my understanding is you can only set $xavu once, so is there another way to manipulate the contact_host on requests coming in and out on different interfaces? I assume I am missing something? :) Thanks in advance Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

2 1

NIST advisory
by David Villasmil 02 Dec '21

02 Dec '21

Anyone knows about this? https://nvd.nist.gov/vuln/detail/CVE-2020-28361 -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 4

Re: [SR-Users] probles with listen advertise address trying to be resolved
by Richard Robson 01 Dec '21

01 Dec '21

no the problem appear to be with the advertise this makes all listeners listen = PRIV_SOCK advertise MY_PUB_SOCK name = "ext" but listen = PRIV_SOCK_TLS advertise MY_PUB_SOCK_TLS name = "ext_tls" is failing with a look up error on MY_PUB_SOCK_TLS if I try : listen = PRIV_SOCK_TLS advertise "MY_PUB_SOCK_TLS" name = "ext_tls" it fails on a lookup for what ever the def is for that variable e.g. 1.2.3.4:5061 the defs are of the correct syntax if it replace the defs it works e.g listen = tls:10.10.10.10:5061 advertise 1.2.3.4:5061 name = "ext_tls" the two defs math the text above the problen is its working for the "global" defs without any protocl, so its making the tls listener for the wrong port as 5060 so listen = LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS name = "ext" is working with #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" and #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" the _TLS versions are not changing the tls port so i have tls on port 5060 when iot should be on 5061 and the name ext_tls where all other sockets are called "ext" Regards, Richard On 30/11/2021 20:03, Karsten Horsmann wrote: > Hi Richard, > > Hmmm for some reason I split my vars into parts. > > And you prefix that with tls? Like this? > > "!ADVERTISE_NAT_ADDRESS_TLS!tls:1.2.3.4:5061!g" > > Best > Karsten > > Richard Robson <richard(a)rikrobson.co.uk> schrieb am Di., 30. Nov. > 2021, 17:48: > > Hi > > > the missing = was my typo in the email and not the config. > > the probelm is only with the TLS advertise and not the non tls > advertise > > these work as expected > > listen = LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS > name = "ext" > > #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" > > > these do not works > > #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" does not work > > listen = LISTEN_EXTERNAL_SOCKET_TLS advertise > ADVERTISE_NAT_ADDRESS_TL name = "ext-tls" > > or > > listen = LISTEN_EXTERNAL_SOCKET_TLS advertise > ADVERTISE_NAT_ADDRESS_TLS name = "ext-tls" > > > > if its not quoted then error = cannot resolve 1.2.3.4:5061 > <http://1.2.3.4:5061> of not quoted error = cannot resolve > ADVERTISE_NAT_ADDRESS_TLS > > > if i replace the def with the 1.2.3.4:5061 <http://1.2.3.4:5061> > it works > > > what is hte difference if also changed the name to > ADVERTISE_NAT_ADDRESS_SEC with the same results > > > > > On 30/11/2021 16:21, Karsten Horsmann wrote: >> Hi Richard, >> >> seems that you forgot the equal signs in the listen directive. >> Did kamailio -c -f /etc/kamailio/kamailio.cfg return something? >> http://www.kamailio.org/wiki/cookbooks/devel/core#listen >> >> I did the same with puppet. And I only manipulate my >> /etc/kamailio/kamailio-local.cfg with the configuration >> management tool. >> Small example from my templating. >> >> grep MY_PUB_IP /etc/kamailio/kamailio-local.cfg >> #!substdef "!MY_PUB_IP!1.2.3.4!g" >> .... >> egrep 'MY_HTTPS_ADDR|MY_PUB_IP|_PORT' /etc/kamailio/kamailio.cfg >> import_file "kamailio-local.cfg" >> .... >> #!substdef "!MY_HTTPS_ADDR!tls:MY_PUB_IP:MY_WSS_HTTPS!g" >> .... >> listen=udp:MY_PUB_IP:5060 >> listen=tcp:MY_PUB_IP:5060 >> listen=tls:MY_PUB_IP:5061 >> listen=MY_HTTPS_ADDR >> >> Kind regards >> Karsten >> >> Am Di., 30. Nov. 2021 um 16:57 Uhr schrieb Richard Robson >> <richard(a)rikrobson.co.uk>: >> >> I have a problem when trying to add an advertised address in >> Kamailio 5.4.2 >> >> >> we have a checf config that is populating various variable. >> some of these are for the Listeners >> >> >> >> we have this working and Im getting all protocols for the >> listeners based on the non _TLS Vars but the TLS config is >> trying to resolve ADVERTISE_NAT_ADDRESS_TLS. If I try and >> quote the variable "ADVERTISE_NAT_ADDRESS_TLS" id get an >> error saying 1.2.3.4:5061 <http://1.2.3.4:5061> is not resolvable >> >> >> therefroe I have udp,tcp and tls listeners all on the default >> ports where i need the tls on port 5061 ans not 5060 >> >> >> listen = LISTEN_EXTERNAL_SOCKET advertise >> ADVERTISE_NAT_ADDRESS name = "ext" >> >> listen = LISTEN_INTERNAL_SOCKET name = "int" >> >> listen = LISTEN_EXTERNAL_SOCKET_TLS advertise name = "ext-tls" >> >> listen = LISTEN_INTERNAL_SOCKET_TLS name = "int-tls" >> >> the Vars are defined as >> >> #!define TOPO_NAT >> >> #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" >> >> #!substdef "!LISTEN_INTERNAL_SOCKET!10.10.10.10:5080!g" >> >> #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" >> >> #!define ENABLE_TLS >> >> #!define TLS_CONFIG_FILE "chef-managed/tls.cfg" >> >> #!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" >> >> #!substdef "!LISTEN_INTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" >> >> #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" >> >> REgards, >> >> >> Richard >> >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> * sr-users(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not >> reply only to the sender! >> Edit mailing list options or unsubscribe: >> * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> Mit freundlichen Grüßen >> *Karsten Horsmann* >> >> __________________________________________________________ >> Kamailio - Users Mailing List - Non Commercial Discussions >> *sr-users(a)lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the sender! >> Edit mailing list options or unsubscribe: >> *https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >

2 1

Using path with DMQ to send calls to origin SBC - not sure when to modify destination URI
by Rhys Hanrahan 01 Dec '21

01 Dec '21

Hi Everyone, I am trying to add DMQ for redundancy of registrations and USRLOC, and I’m trying to send calls to the correct SBC that is the original registrar for a handset. I’ve been using the thread here where Charles gave some guidance on how to use path to store and use the original SBC: https://lists.kamailio.org/pipermail/sr-users/2018-February/100246.html and https://lists.kamailio.org/pipermail/sr-users/2013-September/079736.html I am struggling with when to decide to modify the destination URI. My testing shows this is required otherwise some handsets will ignore the invite, but right now I am doing it in a blanket form, right before SIPOUT (which is where the origin SBC handles the invite instead of LOCATION). I am sure this is being too heavy-handed though and there are some cases where I won’t want to set this, but I am not sure which? # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } … xlog("Setting du according to path. Current du is $du\n"); xlog("Current route header: $(hdr(Route))\n"); xlog("Current route: $(hdr(Route){uri.host})\n"); >>> $du = $(hdr(Route){param.value,received}); #xlog("New du destination uri is: $du\n"); # dispatch requests to foreign domains route(SIPOUT); In the linked threads, Charles mentioned that only the last-hop registrar should make this change, but what’s the best way to determine if I am on the last-hop registrar? As per the snippet above, I tried using the {uri.host} transformation to extract the origin SBC’s IP from the route header. My plan is to then compare this against “myself” but I am struggling to extract the right info from the route header. And I am not even sure if this is the right general approach? The route header looks like this: <sip:ORIGIN_SBC_IP:5060;received=sip:UAC_WAN_IP:2048;lr> Any guidance or examples would be appreciated. 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_555064778] <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.

2 5

Re-writing presentity domain for DMQ
by Rhys Hanrahan 01 Dec '21

01 Dec '21

Hi Everyone, I am trying to get DMQ working for presence/dialog notifications. Presence works fine locally on the same SBC. The DMQ part itself is also working fine and I can see that the entries in presentity are being replicated between my DMQ nodes. The problem is that when the presentity records are being replicated from another SBC, the domain for that record stays as the IP of the original SBC. Then when the local_watchers is populated, the presentity_uri doesn’t match, and no NOTIFY packets are sent as a consequence. What am I missing here? I can’t see anything in the docs about how to handle this, and I can’t find any examples in the list on how to handle this either, which suggests it isn’t as common of an issue at it seems. See example of what I see in the presentity table below. In this case handset 103 is subscribed to BLFs for 103 and 104. A call came into 103 and generated an early state dialog in the presentity table. MariaDB [kamailio]> select * from presentity; +-------+-------------+--------------+--------+-- | id | username | domain | event | +-------+-------------+--------------+--------+-- 27100 | nexusone103 | x.x.x.53 | dialog | I have modified my registrations and have re-written my SUBSCRIBE and PUBLISH packet headers so that everything uses a common FQDN for both SBCs, that does DNS RR to both SBC IPs. I am also registering from the handset using this FQDN now, and yet although my active_watchers table now expects a presentity URI of XXX(a)sbc.fqdn<mailto:XXX@sbc.fqdn>, the presentity table is still using the IP addresses of the individual node that originated the dialog. MariaDB [kamailio]> select presentity_uri,watcher_username,watcher_domain,event from active_watchers; +----------------------------------------------+------------------+----------------------+-----------------+ | presentity_uri | watcher_username | watcher_domain | event | +----------------------------------------------+------------------+----------------------+-----------------+ | sip:104@sbc.fqdn | nexusone103 | sbc.fqdn | dialog | | sip:103@sbc.fqdn | nexusone103 | sbc.fqdn | dialog | +----------------------------------------------+------------------+----------------------+-----------------+ I’m out of ideas unfortunately. Any advice would be appreciated. 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_1328564021] <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.

1 0

probles with listen advertise address trying to be resolved
by Richard Robson 30 Nov '21

30 Nov '21

I have a problem when trying to add an advertised address in Kamailio 5.4.2 we have a checf config that is populating various variable. some of these are for the Listeners we have this working and Im getting all protocols for the listeners based on the non _TLS Vars but the TLS config is trying to resolve ADVERTISE_NAT_ADDRESS_TLS. If I try and quote the variable "ADVERTISE_NAT_ADDRESS_TLS" id get an error saying 1.2.3.4:5061 is not resolvable therefroe I have udp,tcp and tls listeners all on the default ports where i need the tls on port 5061 ans not 5060 listen LISTEN_EXTERNAL_SOCKET advertise ADVERTISE_NAT_ADDRESS name = "ext" listen LISTEN_INTERNAL_SOCKET name = "int" listen LISTEN_EXTERNAL_SOCKET_TLS advertise name = "ext-tls" listen LISTEN_INTERNAL_SOCKET_TLS name = "int-tls" the Vars are defined as #!define TOPO_NAT #!substdef "!LISTEN_EXTERNAL_SOCKET!10.10.10.10:5060!g" #!substdef "!LISTEN_INTERNAL_SOCKET!10.10.10.10:5080!g" #!substdef "!ADVERTISE_NAT_ADDRESS!1.2.3.4:5060!g" #!define ENABLE_TLS #!define TLS_CONFIG_FILE "chef-managed/tls.cfg" #!substdef "!LISTEN_EXTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" #!substdef "!LISTEN_INTERNAL_SOCKET_TLS!tls:10.225.73.16:5061!g" #!substdef "!ADVERTISE_NAT_ADDRESS_TLS!1.2.3.4:5061!g" REgards, Richard

2 2

Kamailio and RTPEngine acting as SBC : wrong IP address returned
by Cyril Ramière 28 Nov '21

28 Nov '21

Hi there, I'm already using Kamailio as a simple proxy and it worked flawlessly. This time, I need to have a new setup like this: [UAC (sip phone)] <--- register & has to deal only with ----> | PUBLIC | [Kamailio + RtpEngine] | PRIVATE | <----> [Asterisk] Client = 37.100.100.100 (public) Asterisk = 172.17.200.20 (private) Kamailio/RtpEngine = 172.17.203.102 (private, advertising 3.99.99.99 public IP) Kamailio and RtpEngine will behave as a SBC/B2BUA, the client will deal only with kamailio using public access (internet), and asterisk will handle the calls but stays 100% private behind kamailio. Kamailio will handle registrations to reduce the load put on the asterisk boxes. The setup is running on AWS, my Kamailio has 1 Private IP address tied to 1 Public IP address (an Elastic public IP) so, no multi-homing, one network interface. I made a "simple" work in progress configuration, the registration is working as a PoC, and my calls are *almost* working but with some nonsense. My Main issue is that when my UAC (SIP phone) make a call, to let's say +1000, the call enters kamailio, then kamailio handle the invite, trigger rtpEngine, but the IP addresses in Record-Route and Via headers are wrong. >From Kamailio to UAC Sip Phone, it seems OK. But from Kamailio to Asterisk, the record-route/via headers are set with Kamailio Public IP when it should be the Kamailio private IP. So, when Asterisk sends a BYE, it tries to send it to the public IP of Kamailio, which is wrong. There are probably other quirks in the configuration but this one, I can't figure out what's wrong. I'm attaching a capture file, a log file and my configuration file (some informations are redacted). I am missing something and sadly I can't figure out what, a little help would be appreciated :) Regards

2 1

Problem implementing Outbound Proxy
by Christoph Russow 26 Nov '21

26 Nov '21

Hi all, im currently trying to add a Outbound Proxying to our system which right now does inbound loadbalancing for our application servers. Current: Inbound SIP Trunk -> Kamailio -> dispatcher -> ApplicationServer 1..n i now want to add the ability for our internal Application Servers to terminate their outbound calls through the kamailio system. currently i've setup a gateway in the (freeswitch) application server that has the kamailio system set as outbound-proxy. this leads to sip calls started with originate sofia/gateway/kamailio/<number>@<outbound carrier IP> to be sent to the kamailio system which then forwards it to the given carrier IP. Sadly our main carrier answers with "503 service unavailable - no bandwith available". (maybe due to the internal ip-addresses within some of the headers?) i've attached our current kamailio configuration and also a pcap of a call i tried to start with our main carrier. i tried this way as it seemed to be the easiest way so our application servers still can select the outbound carrier within their flow. maybe i'm on the wrong way here and need to solve this problem in a completely different way i don't know. TLDR: sip proxying does not work as expected. need a way to achieve outbound calls routed through kamailio from internal application servers any hint/help into the right direction is appreciated Best Regards Christoph Russow -- ________________________________________________________________ EMTEX GmbH Christoph Russow Software Engineer Bischof-Otto-Weg 9 D-91086 Aurachtal Geschäftsführer: Markus Enzinger Tel. +49 9132 7490 0 Sitz der Gesellschaft: 91086 Aurachtal Fax. +49 9132 7490 900 Amtsgericht Fürth: HRB6804 ________________________________________________________________

3 11

Call limiting with Kamailio
by Olli Attila 26 Nov '21

26 Nov '21

Hello, What choices do I have to implement a feature to Kamailio which would limit calls per customer? The environment: Two kamailio nodes sitting in different sites and they are not aware of each others sessions/dialogs and thus serving customer calls independently. Configurations on the nodes are identical excluding ip addresses. Customer call can be routed through either one of these Kam nodes. The two nodes share a common external mysql database from which all dialplan and htable entries are loaded (used in call routing for example). Customers are identified based on IP address & tel numbering and as a final result we are populating an variable which holds the customer id $var(REALM_NAME) when the identification has been done. The call limiting should be "tagged/profiled" somehow with this information if possible. Any suggestions which way to go with this? Initial ideas if one of these are worth of setting a starting point: -> Counting calls with htable and replicating with DMQ to other node (can the data survive from a restart?) -> Using the common mysql database to hold the counted calls from where both kam nodes could load data? (database load increasing?) -> Some new exciting module for this job? Cheers, Olli

4 3

cannot adopt OIDs in KAMAILIO-SIP-COMMON-MIB
by Arun Bonela 26 Nov '21

26 Nov '21

Hi we have configured snmp module on our Kamailio server following https://www.kamailio.org/dokuwiki/doku.php/utils:kamailio-and-snmp guidance. Description we could able to poll or snmpwalk most of the OIDs in KAMAILIO-REG-MIB , KAMAILIO-SIP-SERVER-MIB but having issues below from KAMAILIO-SIP-COMMON-MIB snmpd[358730]: Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsMethod ::= { kamailioSIPMethodStatsEntry 1 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsOutbounds ::= { kamailioSIPMethodStatsEntry 2 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatsInbounds ::= { kamailioSIPMethodStatsEntry 3 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeMethod ::= { kamailioSIPStatusCodesEntry 1 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeValue ::= { kamailioSIPStatusCodesEntry 2 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeIns ::= { kamailioSIPStatusCodesEntry 3 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeOuts ::= { kamailioSIPStatusCodesEntry 4 } Cannot adopt OID in KAMAILIO-SIP-COMMON-MIB: kamailioSIPStatusCodeRowStatus ::= { kamailioSIPStatusCodesEntry 5 } Troubleshooting root@host# snmpwalk -v2c -c public localhost .1.3.6.1.4.1.34352.3.1.1.1.4.1.1.2.0 SNMPv2-SMI::enterprises.34352.3.1.1.1.4.1.1.2.0 = No Such Object available on this agent at this OID root@host# snmpwalk -v2c -c public localhost .1.3.6.1.4.1.34352.3.1.1.1.4.1.1.2 SNMPv2-SMI::enterprises.34352.3.1.1.1.4.1.1.2 = No Such Object available on this agent at this OID Reproduction using snmpwalk or snmget to any of the above mentioned kamailioSIPStats* OIDs will do Additional Information * Kamailio Version - output of kamailio -v version: kamailio 5.5.1 (x86_64/linux) 307af8-dirty * Operating System: No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 20.04.2 LTS Release: 20.04 Any advice, why we seeing this issue for these OIDs ? Are these decremented now ? Regards, Arun Bonela Arun Bonela UCaaS Ops – Technical Lead Tel: 0333 043 7319 Mobile: 07468 728 007 Email: Arun.Bonela(a)gamma.co.uk This is an email from Gamma Telecom Ltd, trading as “Gamma”. The contents of this email are confidential to the ordinary user of the email address to which it was addressed. This email is not intended to create any legal relationship. No one else may place any reliance upon it, or copy or forward all or any of it in any form (unless otherwise notified). If you receive this email in error, please accept our apologies, we would be obliged if you would telephone our postmaster on +44 (0) 808 178 9652 or email postmaster(a)gamma.co.uk Gamma Telecom Limited, a company incorporated in England and Wales, with limited liability, with registered number 04340834, and whose registered office is at 5 Fleet Place, London, EC4M 7RD and whose principal place of business is at Kings House, Kings Road West, Newbury, Berkshire, RG14 5BY.

2 2

Re: [SR-Users] How to not write logs to journald with kamailio
by Ginhoux, Patrick 26 Nov '21

26 Nov '21

Hi, I resend this mail because of the initial destination address was not correct. The topic was a logging problem : With the upgrade of my SIP Router proxies to RHEL 7 (and Kamailio 5.0.7) , we encounter problems where the logs are blocked to be written in the syslog. This problem is similar as the one reported in this article : [SR-Users] Kamailio logs in RHEL 7 environment <https://lists.kamailio.org/pipermail/sr-users/2018-April/101065.html> . The resolution was to increase some journald/rsyslog settings (RateLimitBurst=1000000 / $imjournalRatelimitInterval 1 $imjournalRatelimitBurst 500000). But it is somehow a workaround. Im looking to know if it should be possible for kamailio to write directly the logs to the syslog only. Thanks in advance to those can help me. Regards Patrick Ginhoux De : Ginhoux, Patrick Envoyé : lundi 22 novembre 2021 16:49 À : 'sr-users' <sr-users-bounces(a)lists.kamailio.org> Objet : How to not write logs to journald with kamailio Hi, With the upgrade of my SIP Router proxies to RHEL 7 (and Kamailio 5.0.7) , we encounter problems where the logs are blocked to be written in the syslog. This problem is similar as the one reported in this article : [SR-Users] Kamailio logs in RHEL 7 environment <https://lists.kamailio.org/pipermail/sr-users/2018-April/101065.html> . The resolution was to increase some journald/rsyslog settings (RateLimitBurst=1000000 / $imjournalRatelimitInterval 1 $imjournalRatelimitBurst 500000). But it is somehow a workaround. Im looking to know if it should be possible for kamailio to write directly the logs to the syslog only. Thanks in advance to those can help me. Regards Patrick Ginhoux

5 5

JSON Error
by Daniel W. Graham 26 Nov '21

26 Nov '21

My log is filled with this and have been unsuccessful locating additional information by increasing debug level. Any thoughts on how to quickly isolate? /usr/local/sbin/kamailio[7678]: ERROR: jansson [jansson_funcs.c:48]: janssonmod_get_helper(): json error at line 1, col 0: '[' or '{' expected near end of file /usr/local/sbin/kamailio[7677]: ERROR: jansson [jansson_funcs.c:46]: janssonmod_get_helper(): failed to parse json: /usr/local/sbin/kamailio[7677]: ERROR: jansson [jansson_funcs.c:48]: janssonmod_get_helper(): json error at line 1, col 0: '[' or '{' expected near end of file /usr/local/sbin/kamailio[7675]: ERROR: jansson [jansson_funcs.c:46]: janssonmod_get_helper(): failed to parse json: /usr/local/sbin/kamailio[7675]: ERROR: jansson [jansson_funcs.c:48]: janssonmod_get_helper(): json error at line 1, col 0: '[' or '{' expected near end of file -dan

3 5

LCR Module not loading all rules from DB
by Juha Heinanen 25 Nov '21

25 Nov '21

If your db supports DB_CAP_FETCH, lcr rules are loaded from the db 1024 rows per cycle. You could check what you get to syslog from this statement after you set log level to debug level and then execute lcr.reload: LM_DBG("loading, cycle %d with <%d> rows\n", n++, RES_ROW_N(res)); -- Juha

1 0

Problem with 302
by nathan Kirk 24 Nov '21

24 Nov '21

Hello, First time poster and I am stuck. I am using K as a mid registar to several AST boxes. The problem I am having is if AST or the END USER DEVICE sends a 302 for some reason I am getting a 500 THEN the 302 to the END USER DEVICE or AST depending which direction the call flow is. Example END USER DEVICE -> CALLS 700 -> K -> AST AST -> 302 -> K -> 500 & 302 -> END USER DEVICE What I would like to happen is when K receives the 302 it deals with it directly and it is not handed off to either end. I am using a configuration I found online with some base modifications to dip into the database real time to get the AST box IPS Any direction would be helpful. Thank you, Nathan

1 0

LCR Module not loading all rules from DB
by M Arqum CH 24 Nov '21

24 Nov '21

Hi all, lcr.reload command not loading all rules from rules table. More than 73K rules row in table, lcr.dump_rules only print 4096 rules. lcr module config : modparam("lcr", "db_url", "cluster://k1") modparam("lcr", "lcr_gw_table","lcr_gateways") modparam("lcr", "lcr_rule_table", "lcr_rules") modparam("lcr", "lcr_rule_target_table", "lcr_rule_targets") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip") modparam("lcr", "lcr_id_column", "lcr_id") modparam("lcr", "gw_uri_avp", "$avp(gw_uri_avp)") modparam("lcr", "ruri_user_avp", "$avp(ruri_user_avp)") modparam("lcr", "flags_avp", "$avp(s:cSId)") modparam("lcr", "flags_column", "cSId") modparam("lcr", "defunct_capability", 1) modparam("lcr", "lcr_id_avp", "$avp(lcr_id_avp)") modparam("lcr", "defunct_gw_avp", "$avp(defunct_gw_avp)") modparam("lcr", "lcr_rule_hash_size", 1024) modparam("lcr", "lcr_gw_count", 1024) modparam("lcr", "fetch_rows", 3000) modparam("lcr", "ping_interval", 15) # Ping every 15 seconds modparam("lcr", "ping_inactivate_threshold", 1) # Mark as inactive at first call of inactivate_gw() modparam("lcr", "ping_valid_reply_codes", "403,404,405,501") # Mark as active when 2XX or these reply codes modparam("lcr", "ping_from", "sip:lcr@mydomain.com") modparam("lcr", "lcr_count", 1) modparam("lcr", "dont_strip_or_prefix_flag", 1) modparam("lcr", "rule_id_avp", "$avp(lcr_rule_id)") modparam("lcr", "priority_ordering", 1) modparam("lcr", "ping_socket", "10.16.16.16:5060") Thanks & Regards -- Regards M Arqum

3 5

t_suspend() / t_continue() create new branches
by Sebastian Damm 24 Nov '21

24 Nov '21

Hi, I have the following weird problem. A Kamailio (5.4.x) with some asynchronous permission logic suspends outgoing INVITEs and resumes them after the answer of the permission system. Everything works as expected when the call is answered. However, when the called party rejects the call (with a 486 for example), the call doesn't get torn down immediately (it is rejected with a 408 eventually) because Kamailio logs the following: Nov 23 14:51:02 host /usr/sbin/kamailio[1569]: DEBUG: app_python [apy_kemi.c:106]: sr_kemi_config_engine_python(): execution of route type 4 with name [ksr_onreply_manage] returned 1 Nov 23 14:51:02 host /usr/sbin/kamailio[1569]: DEBUG: tm [t_reply.c:1363]: t_should_relay_response(): ->>>>>>>>> T_code=183, new_code=486 Nov 23 14:51:02 host /usr/sbin/kamailio[1569]: DEBUG: tm [t_reply.c:1448]: t_should_relay_response(): store - other branches still active Nov 23 14:51:02 host /usr/sbin/kamailio[1569]: DEBUG: tm [t_reply.c:1926]: relay_reply(): reply status=3 branch=2, save=1, relay=-1 icode=0 msg status=486 I don't do any forking, the call is just forwarded. I put the following line immediately after t_suspend() and in the beginning of the route specified in t_continue(): KSR.info("We're at branch index %d" % KSR.pv.get("$T(branch_index)")) This produces the following output: Nov 23 16:30:13 host /usr/sbin/kamailio[2745]: INFO: <core> [core/kemi.c:104]: sr_kemi_core_info(): We're at branch index 0 Nov 23 16:30:13 host /usr/sbin/kamailio[2725]: INFO: <core> [core/kemi.c:104]: sr_kemi_core_info(): We're at branch index 2 So apparently, the suspend and continue introduce new branches. (Other option: I'm doing something completely different creating those branches, but I don't have any append_branch() or similar in my code.) I assume, I just missed a configuration parameter. Can I get Kamailio somehow to behave as if there was just one branch and forward the negative reply? Thanks, Sebastian

2 3

KEMI Golang
by Eugeniu Istrati 24 Nov '21

24 Nov '21

Hello, Is there any plans to add golang into the KEMI framework? Thank you -- Eugeniu Istrati

2 1

[uac] How to force outbound registration?
by Denys Pozniak 24 Nov '21

24 Nov '21

Hello! In some cases, I need to forcibly unregister a remote SIP account through uac module, and then immediately register. To unregister I can use the command: *kamcmd uac.reg_unregister l_uuid <account>* But it's not clear how to immediately register back. For example, these commands do not give the desired effect: *kamcmd uac.reg_enable l_uuid <account>* *kamcmd uac.reg_reload* [root@kamailio]# kamailio -V version: kamailio 5.5.2 (x86_64/linux) 55e232 -- BR, Denys Pozniak

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Kamailio v5.5.3 Released
by Daniel-Constantin Mierla 24 Nov '21

24 Nov '21

Hello, Kamailio SIP Server v5.5.3 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.3. For more details about version 5.5.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/11/kamailio-v5-5-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Releasing Kamailio v5.5.3
by Daniel-Constantin Mierla 24 Nov '21

24 Nov '21

Hello, I am considering to release Kamailio v5.5.3 sometime next week, likely on Wednesday or Thursday (Nov 24/25, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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UDP packets lost - tuning Linux system configuration - Kamailio using default for SO_RCVBUF ?
by Chaigneau, Nicolas 23 Nov '21

23 Nov '21

Hello, I'm using Kamailio 5.5.2 with UDP sockets, configured with 32 children processes. During load tests, some packets seem to be dropped. I'm trying to figure out why, and how to solve (or mitigate) the issue. >From what I understand : < the SIP messages send on UDP/SCTP are received directly from the buffer in kernel one by one, each being processed once read. > (this is a message from Daniel-Constantin Mierla, posted on the mailing list back in 2014 - I assume this is still accurate) So Kamailio does not handle its own buffers for incoming messages, but simply pulls one message from the kernel queue whenever a process is available for processing. (Correct me if this is not true) Hereafter some system parameters that seem interesting to me. In particular, < net.core.rmem.default > seem very low (0,33 MB). But < net.core.rmem.max > is much higher (16 MB). Looking at the code in function probe_max_receive_buffer (src\core\udp_server.c), it seems Kamailio is trying to increase the buffer size from the default. However this does not seem to be working, seeing the following logs upon startup : 0(7602) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 349520 0(7602) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 349520 0(7602) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 349520 0(7602) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 349520 0(7602) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 349520 0(7602) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 349520 Is there something I don't understand ? could it be a bug ? One quick fix I can think of would be to increase < net.core.rmem.default >. Considering that I don't have many sockets (3 used by Kamailio), I think I could set this to 16 MB (my VMs have 8 GB of allocated RAM). What do you think ? Thanks in advance for your help. My system parameters : net.core.rmem_default = 349520 net.core.rmem_max = 16777216 # => 0,33 MB / 16 MB respectively * net.core.rmem_default, net.core.rmem_max - default and max socket receive buffer size in bytes. Each socket gets rmem_default reveive buffer size by default, and can request up to rmem_max with setsockopt<http://man7.org/linux/man-pages/man2/setsockopt.2.html> option SO_RCVBUF. net.ipv4.udp_mem = 188856 251809 377712 * net.ipv4.udp_mem = "min pressure max" - these are numbers of PAGES (4KB) available for all UDP sockets in the system. min, pressure and max controls how memory is managed, but main point is that max is maximum size in PAGES for all UDP bufers in system. These values are set on boot time (if sysctls are not explicitly set) according to available RAM size. # 377712 x 4 KB => 1475 MB net.ipv4.udp_rmem_min = 4096 * net.ipv4.udp_rmem_min, net.ipv4.udp_wmem - minimal size for receive/send buffers (in bytes), guaranteed for each socket, even if if buffer size of all UDP sockets exceeds pressure parameter in net.ipv4.udp_mem. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

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Custom info in Contact header
by Igor Olhovskiy 23 Nov '21

23 Nov '21

Hello! On REGISTER I have a Contact header like Contact: <sip:90002@192.168.0.26:48579;transport=TLS>;some_id=90002-21080 But on calling fix_nated_register() -> save(), param some_id is lost from info, so, on INVITE to this account, some_id not appears in RURI. Is this a bug or a feature and I do need to find some workaround? -- Best regards, Igor

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Help with Async transaction responses
by Michael Friesen 19 Nov '21

19 Nov '21

Hello, I am attempting to perform some asynchronous lookup upon receiving an INVITE and then reply with a 3xx response with a modified From header. My workflow now is to create a transaction when receiving the INVITE, suspend the transaction, perform the asynchronous lookup, continue the transaction upon response, and then modify the From header and reply. The ONLY problem I am having is modifying the From header! How can I modify the From header in this scenario? function ksr_request_route() if KSR.is_INVITE() then if KSR.tm.t_newtran() then cache_transaction_info() KSR.tmx.t_suspend() --perform async loookup end end return 1 end --This function called on response from async lookup. Equivalent of ONREPLY_ROUTE function ksr_reply_route() --Finds the transaction_index and transaction_label from a cache KSR.tmx.t_continue(transaction_index, transaction_label, "redirect_transaction"); KSR.x.drop() return 1 end function redirect_transaction() KSR.pv.sets("$fU", "Jenny") --Doesn't actually modify the from username for 302 response! KSR.tm.t_reply(302, "Redirecting") return 1 end Thank you much! -Michael

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+ character in send_reply causes truncation of reply message?
by Benoît Panizzon 18 Nov '21

18 Nov '21

Hi send_reply("403", "+411234567 transit disallowed"); Causes the reply to be truncated after the e164 number. send_reply("403", "Forbidden +411234567 transit disallowed"); Causes the reply to be truncated after 'Forbidden'. Is there a special meaning to the + character? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

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configuration dispatcher
by sondes boughdiri 17 Nov '21

17 Nov '21

Hi ALL I need to configure dispatcher call to the server asterisk i have a 21 server and i need to dispatcher call to all the server AGENT----> kamailio --->asterisk1 ---> asteisk2 ---> - ---> - ---> - ---> asterisk21 please i need your help THANK YOU

2 1

ki_jsonrpcs_dispatch(): invalid json doc
by Patrick Karton 16 Nov '21

16 Nov '21

hello , im using kamailio and rtpengine. i use rtpengine with rtp timeout as follows rtpengine --timeout=240 --silent-timeout=600 -b2b-url=http://kamailioip:5060/RPC --xmlrpc-format=2 and in kamailio modparam("jsonrpcs", "pretty_format", 1) modparam("jsonrpcs", "transport", 7) event_route[xhttp:request] { if ($hu =~ "^/RPC") { jsonrpc_dispatch(); xhttp_reply("200", "OK", "text/xml", "<methodResponse><params><param><value><string>OK</string></value></param></params></methodResponse>"); exit; } } rtpengine successfully reaches the rpc endpoint but i have an error on the jsonrpcs module 43(50) ERROR: jsonrpcs [jsonrpcs_mod.c:1275]: ki_jsonrpcs_dispatch(): invalid json doc [[<?xml version="1.0" encoding="UTF-8"?> <methodCall> <methodName>dlg.terminate_dlg</methodName> <params> <param><value><string>1-10(a)172.22.244.139</string></value></param> <param><value><string>10SIPpTag091</string></value></param> <param><value><string>8U173ymKpc1pp</string></value></param> </params> </methodCall> ]] and so the dialog module can not ends the call Kamailio 5.5.2 on debian bullseye rtpengine 10.0.1.8 on debian bullseye Thanks.

1 0

xavp functionnality in htable
by Patrick Karton 15 Nov '21

15 Nov '21

hello, do we have something like xavp in htable or a module that implements having multiple values in stack like xavp but not tied to transaction. i want data be seen by all processes but with the functionnality of xavp. Thanks.

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Info: Astricon 2021 - Asterisk and LRKProxy Session
by Mojtaba 13 Nov '21

13 Nov '21

Hello, The Asterisk conference (Astricon 2021) happened last week, I had a session about extending Asterisk with the LRKProxy module in Kamailio in terms of RTP sessions. cheers, -- --Mojtaba Esfandiari.S

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corex send()
by Ross McKillop 12 Nov '21

12 Nov '21

Hi All, Is there an equivalent to corex's send_tcp (https://kamailio.org/docs/modules/5.2.x/modules/corex.html#corex.f.send_tcp <https://kamailio.org/docs/modules/5.2.x/modules/corex.html#corex.f.send_tcp>) for TLS? I have a bit of an odd edge case where I can't add Via headers etc, and send_udp and send_tcp allow me to achieve what I want, but it doesn't seem to have a means of doing the same with TLS? Essentially I want to transparently proxy what is received by Kamailio on one interface to a TLS connection e.g. A = (TCP) => Kamailio = (TLS) => Remote Host (A) in this instance is already setting a Via header of the Kamailio's external IP and transport tis. I appreciate this is a non-standard way of doing it, but (A) is a black box that I can't change, what I'm essentially trying to do is have Kamailio handle the TLS but essentially (A) thinks it's at the edge. I can make it work with send_tcp for a standard TCP connection, but if I use t_relay or even forward() to send onwards by TLS then Kamailio obviously adds its own headers as you'd expect. Ross

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Kamailio dropping ACKs
by Emilio Panighetti 12 Nov '21

12 Nov '21

Working on a proof of concept with Kamailio 5.5.2 (Installed via kamailio-5.5.2-0.el8.centos.x86_64 RPM from rpm.kamailio.org). Also note that this is using the LUA interface Running a simple SIPp scenario without registration. Routing is set with in Lua: KSR.dbg("EP: ksr_route_relay Check for Base Route\n"); if KSR.is_INVITE() and KSR.siputils.has_totag() < 0 then KSR.dbg("EP: ksr_route_relay Base Route\n"); --KSR.sl.send_reply("503", "OLGX Debug"); KSR.setdsturi("sip:192.168.1.14:5060"); --KSR.pv.sets("$du", "sip:192.168.1.12:5062"); end The problem is that Kamailio is dropping the ACK to the 200 OK from the UAS. Wireshark shows that the UAS is retransmitting the 200 OK and the ACK is not being forwarded from the UAC to the UAS. Within the Lua config script: -- Handle requests within SIP dialogs function ksr_route_withindlg() if KSR.siputils.has_totag() < 0 then return 1; end KSR.dbg("EP: ksr_route_withindlg has to-tag, it is within a dialog\n"); -- sequential request within a dialog should -- take the path determined by record-routing if KSR.rr.loose_route() > 0 then ksr_route_dlguri(); if KSR.is_BYE() then KSR.setflag(FLT_ACC); -- do accounting ... KSR.setflag(FLT_ACCFAILED); -- ... even if the transaction fails elseif KSR.is_ACK() then -- ACK is forwarded statelessly KSR.dbg("EP: ksr_route_withindlg Forwarding ACK\n"); ksr_route_natmanage(); elseif KSR.is_NOTIFY() then -- Add Record-Route for in-dialog NOTIFY as per RFC 6665. KSR.rr.record_route(); end KSR.dbg("EP: ksr_route_withindlg loose route calling ksr_route_relay\n"); ksr_route_relay(); KSR.dbg("EP: ksr_route_withindlg exit\n"); KSR.x.exit(); end if KSR.is_ACK() then if KSR.tm.t_check_trans() > 0 then -- no loose-route, but stateful ACK; -- must be an ACK after a 487 -- or e.g. 404 from upstream server KSR.dbg("EP: ksr_route_withindlg is ACK calling ksr_route_relay\n"); ksr_route_relay(); KSR.dbg("EP: ksr_route_withindlg exit\n"); KSR.x.exit(); else -- ACK without matching transaction ... ignore and discard KSR.dbg("EP: ksr_route_withindlg Unknown ACK exit\n"); KSR.x.exit(); end end KSR.sl.sl_send_reply(404, "Not here"); KSR.dbg("EP: ksr_route_withindlg exit after 404\n"); KSR.x.exit(); end And logs for the ACK: Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/udp_server.c:493]: udp_rcv_loop(): received on udp socket: (106/100/385) [[ACK sip:service@192.168.1.15:5060 SIP/2.0 0D 0A Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-1091869]] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:677]: parse_msg(): SIP Request: Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): method: <ACK> Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:681]: parse_msg(): uri: <sip:service@192.168.1.15:5060> Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:683]: parse_msg(): version: <SIP/2.0> Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Via] type 1 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-1091869-1-5>; state=16 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [From] type 4 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [To] type 3 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=4088926SIPpTag014 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:173]: get_hdr_field(): <To> [63]; uri=[sip:service@192.168.1.15:5060] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:176]: get_hdr_field(): to body (39)[service <sip:service@192.168.1.15:5060>], to tag (17)[4088926SIPpTag014] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Call-ID] type 6 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [CSeq] type 5 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/parser/msg_parser.c:154]: get_hdr_field(): cseq <CSeq>: <1> <ACK> Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: <core> [core/receive.c:387]: receive_msg(): --- received sip message - request - call-id: [1-1091869(a)192.168.1.12] - cseq: [1 ACK] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/receive.c:259]: ksr_evrt_pre_routing(): event route core:pre-routing not defined Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/receive.c:457]: receive_msg(): preparing to run routing scripts... Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sl [sl_funcs.c:443]: sl_filter_ACK(): too late to be a local ACK! Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:489]: sr_lua_reload_script(): No need to reload [/etc/kamailio/kamailio.lua] is version 0 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:673]: app_lua_run_ex(): executing Lua function: [[ksr_request_route]] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:675]: app_lua_run_ex(): lua top index is: 21 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Contact] type 7 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Max-Forwards] type 8 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Subject] type 27 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Content-Length] type 12 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/msg_parser.c:187]: get_hdr_field(): content_length=0 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/msg_parser.c:91]: get_hdr_field(): found end of header Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1007]: sr_kemi_lua_exec_func_ex(): param[0] for: process_maxfwd is int: 69 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 70 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1007]: sr_kemi_lua_exec_func_ex(): param[0] for: is_maxfwd_lt is int: 69 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: maxfwd [maxfwd.c:187]: ki_is_maxfwd_lt(): value = 69 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1007]: sr_kemi_lua_exec_func_ex(): param[0] for: sanity_check is int: 1511 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1007]: sr_kemi_lua_exec_func_ex(): param[1] for: sanity_check is int: 7 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:247]: check_ruri_sip_version(): check_ruri_sip_version entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:270]: check_ruri_sip_version(): check_ruri_sip_version passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:277]: check_ruri_scheme(): check_ruri_scheme entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:297]: check_ruri_scheme(): check_ruri_scheme passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:318]: check_required_headers(): check_required_headers entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:340]: check_required_headers(): check_required_headers passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:506]: check_cseq_method(): check_cseq_method entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:539]: check_cseq_method(): check_cseq_method passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:548]: check_cseq_value(): check_cseq_value entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:576]: check_cseq_value(): check_cseq_value passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:585]: check_cl(): check_cl entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:610]: check_cl(): check_cl passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:622]: check_expires_value(): check_expires_value entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:659]: check_expires_value(): no expires header found Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:759]: check_parse_uris(): check_parse_uris entered Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:763]: check_parse_uris(): parsing ruri Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:777]: check_parse_uris(): looking up From header Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:791]: check_parse_uris(): parsing From header Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1091869SIPpTag001 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:817]: check_parse_uris(): parsing From URI Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:835]: check_parse_uris(): looking up To header Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:857]: check_parse_uris(): check_parse_uris(): parsing To URI Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: sanity [sanity.c:902]: check_parse_uris(): check_parse_uris passed Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: siputils [checks.c:123]: has_totag(): totag found Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_route_withindlg has to-tag, it is within a dialog Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_withindlg has to-tag, it is within a dialog Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: rr [loose.c:108]: find_first_route(): No Route headers found Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: rr [loose.c:1006]: loose_route_mode(): There is no Route HF Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:1034]: t_check_msg(): msg (0x7fef983a2608) id=2/2414 global id=1/2414 T start=0xffffffffffffffff Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=12314, isACK=1 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK-1091869-1-5] Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:675]: t_lookup_request(): no transaction found Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:1103]: t_check_msg(): msg (0x7fef983a2608) id=2/2414 global id=2/2414 T end=(ni l) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_route_withindlg Unknown ACK exit Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_withindlg Unknown ACK exit Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1837]: sr_kemi_lua_exit(): script exit call Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:729]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_mod.c:167]: sr_kemi_config_engine_lua(): execution of route type 1 with no name returned 1 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: INFO: LUA {ACK}: <core> [core/receive.c:515]: receive_msg(): request-route executed in: 364 usec Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/receive.c:606]: receive_msg(): cleaning up Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/udp_server.c:493]: udp_rcv_loop(): received on udp socket: (106/100/577) [[SIP/2.0 200 OK 0D 0A Via: SIP/2.0/UDP 192.168.1.15;branch=z9hG4bKa103.32dd17e8a1a3458280dbbe34442737e7.0,]] Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 2 (reply(status)) flags 1 Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/msg_parser.c:687]: parse_msg(): SIP Reply (status): Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/msg_parser.c:689]: parse_msg(): version: <SIP/2.0> Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/msg_parser.c:691]: parse_msg(): status: <200> Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/msg_parser.c:693]: parse_msg(): reason: <OK> Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Via] type 1 Nov 11 12:24:37 ol-vm1 /usr/sbin/kamailio[2413]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKa103.3 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: tm [t_lookup.c:1103]: t_check_msg(): msg (0x7fef983a2608) id=2/2414 global id=2/2414 T end=(nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_route_withindlg Unknown ACK exit Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_withindlg Unknown ACK exit Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:1837]: sr_kemi_lua_exit(): script exit call Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_api.c:729]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: app_lua [app_lua_mod.c:167]: sr_kemi_config_engine_lua(): execution of route type 1 with no name returned 1 Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: INFO: LUA {ACK}: <core> [core/receive.c:515]: receive_msg(): request-route executed in: 364 usec Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Nov 11 12:24:36 ol-vm1 /usr/sbin/kamailio[2414]: DEBUG: LUA {ACK}: <core> [core/receive.c:606]: receive_msg(): cleaning up The ACK is this: ACK sip:service@192.168.1.15:5060 SIP/2.0\r\n Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK-1091869-1-5\r\n From: sipp <sip:sipp@192.168.1.12:5060>;tag=1091869SIPpTag001\r\n To: service <sip:service@192.168.1.15:5060>;tag=4088926SIPpTag014\r\n Call-ID: 1-1091869(a)192.168.1.12\r\n CSeq: 1 ACK\r\n Contact: sip:sipp@192.168.1.12:5060\r\n Max-Forwards: 70\r\n Subject: Performance Test\r\n Content-Length: 0\r\n \r\n I’ve been trying to figure this out for days and don’t know what’s wrong here. Thanks

3 4

[ndb_redis] init_without_redis issue
by Denys Pozniak 11 Nov '21

11 Nov '21

Hello! In our architecture, Kamailio uses 2 local Redis servers to get routing information. We noticed that if at the start of Kamailio the first in the modparam list is a running Redis server (name=real), and the second in the list is not running (name=fake), then Kamailio will not even try to connect to the running one. # ----- redis server params ----- modparam("ndb_redis", "init_without_redis", 1) modparam("ndb_redis", "server", "name=real;addr=127.0.0.1;port=6379;db=1") modparam("ndb_redis", "server", "name=fake;addr=127.0.0.1;port=6479;db=2") But if the servers in the list above are reversed, then everything will work as it should. Looks the initialization order affects behavior. Syslog output: Nov 11 18:00:45 kamailio /usr/sbin/kamailio[11445]: ERROR: ndb_redis [redis_client.c:224]: redisc_init(): Failed to create REDIS returned an error: Connection refused Nov 11 18:00:45 kamailio /usr/sbin/kamailio[11445]: ERROR: ndb_redis [redis_client.c:259]: redisc_init(): error communicating with redis server [fake] (127.0.0.1:6479/2): Connection refused Nov 11 18:00:45 kamailio /usr/sbin/kamailio[11445]: WARNING: ndb_redis [redis_client.c:264]: redisc_init(): failed to initialize redis connections, but initializing module anyway. There is no established connection to the real server after Kamailio starts both through redis-cli and through ss. [root@kamailio ~]# ss -anp | grep 6379 tcp LISTEN 0 128 127.0.0.1:6379 *:* users:(("redis-server",pid=10376,fd=6)) [root@kamailio ~]# kamailio -v version: kamailio 5.6.0-dev2 (x86_64/linux) 378a2d compiled on 04:12:43 Nov 11 2021 with gcc 4.8.5 [root@kamailio ~]# cat /etc/redhat-release CentOS Linux release 7.9.2009 (Core) g -- BR, Denys Pozniak

1 0

Forcing Outbound Socket
by Ross McKillop 11 Nov '21

11 Nov '21

Hi All, Try as I may I cannot force an outbound TCP request to use a non-ephemeral source port. Looking at the documentation it seems that as long as I have a line like listen=tcp:10.30.0.55:3339 I should be able to force an outbound connection to originate from that host and port, i.e. to establish a persistent connection from a specific socket to a remote host, but having tried to set it via $fs and set_send_socket() it always uses an ephemeral port. Ideally I would like kamailio to use a specific source port for contacting a specific remote host and keep the TCP socket established, and it seems like it should be simple enough to do but it's not working as expected. Best, Ross

4 6

TCP Persistent Connections
by Ross McKillop 10 Nov '21

10 Nov '21

Hopefully the collective hive mind can help with this one, as I have little experience of using TCP in this manner and neither does a friendly Kamailio expert I know personally. I'm replacing a closed box thing that accepts connections from various nodes over TCP. They send INVITEs and REGISTERs over that TCP socket and expect to receive inbound calls over the same established TCP socket. Struggling to make Kamailio behave the same way, but it should be possible. The first time it receives a message, I add the sender to the dispatcher as follows; jsonrpc_exec('{"jsonrpc": "2.0", "method": "dispatcher.add", "params": [5, "sip:$si:$sp;transport=tcp", 0, "socket=tcp:127.0.0.1:5776"], "id": 1}'); This works great, but despite setting the socket to use there, when a call later comes in and it's routed to one of the dispatcher nodes that's been added in this way it fails because Kamailio attempts to open a new connection to the IP / PORT rather than having kept the socket open from the earlier message and reusing the same connection/. Any ideas? I get the feeling I'm missing something obvious! Ross

3 3

Re: [SR-Users] htable.get issues
by Igor Olhovskiy 10 Nov '21

10 Nov '21

Yup, this is works! Thanks! Not reflected in docs, though... ср, 10 нояб. 2021 г. в 16:13, Fred Posner <fred(a)palner.com>: > > On 11/10/21 11:10 AM, Igor Olhovskiy wrote: > > Hello, > > I'm getting issues with *htable.get* rpc command > > Running > > kamctl rpc htable.get client_to_re 65825 > > What happens when you try: > > s:65825 vs 65825 > > > > Fred Posner | palner.com > Matrix: @fred:matrix.lod.com > o: +1 (212) 937-7844 > -- Best regards, Igor

1 0

htable.get issues
by Igor Olhovskiy 10 Nov '21

10 Nov '21

Hello, I'm getting issues with *htable.get* rpc command Running kamctl rpc htable.get client_to_re 65825 { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters (htable name and key name)" }, "id": 46641 } To mention, there is such name in table and if I specify wrong name - I'm getting corresponding error Kamailio 5.4.6 Running commands to *seti*, *sets* leads to same error answer Is it something to add a bug or am I missing something obvious? -- Best regards, Igor

1 0

$var assignment broken on master ?
by Chaigneau, Nicolas 09 Nov '21

09 Nov '21

Hello, I've just built Kamailio from the master. Assigning a value to a < var > doesn't seem to work anymore... This works with Kamailio 5.5.2. E.g. : $var(my_var) = "hello"; xlog("L_WARN", "var(my_var): $var(my_var)\n"); # => This shows value 0. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 4

How to determine size (number of elements) in xavp stack?
by Benoît Panizzon 09 Nov '21

09 Nov '21

Dear Community In my previous post I mentioned my problem with sql_xquery I fear I used this wrong. $dbr is probably only populated if sql_query is used. Anyway, it is always 0 in my testings, even if I clearly get a result. Now I would like to know how many results I have. How can I determine the size of an $xavp stack (or array)? Or is there another way to get the number of rows returned by a sql_xquery without looping an index through all rows? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 3

sql_xquery return 1 on 0 rows?
by Benoît Panizzon 09 Nov '21

09 Nov '21

Dear List... kamcmd> version kamailio 5.4.7 (x86_64/linux) Strange issue found. According to: https://kamailio.org/docs/modules/5.4.x/modules/sqlops.html I should get return value 2 if no rows returned: $var(query) contains a query that does not match (number is not assigned) if (sql_xquery("impkam", "$var(query)", "assignedtn") == 1) { if ($avp(debug) > 1) { xlog("L_INFO", "$cfg(route): SQL Dump Result: $var(assignedtn) ROWS: $dbr(assignedtn=>rows) \n"); } do stuff with the assigned number } else { do stuff in case that number is not assigned } Log Output: CHECK_ASSIGNED_TN: SQL Dump Result: 0 ROWS: 0 So the return value of the query was == 1 but the result contains no rows. How can that be? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

4 4

STIR/SHAKEN public key
by David Villasmil 05 Nov '21

05 Nov '21

Hello guys, I'm testing with 2 providers right now, and one of them is asking me to include my whole certificate on the *secsipid_add_identity(origTN, destTN, attest, origID, x5u, keyPath)* like: *secsipid_add_identity("$fU", "$rU", "A", "", "https://kamailio.org/stir/$rd/cert.pem <https://kamailio.org/stir/$rd/cert.pem>", "/secsipid/$rd/key.pem");* but it is stated that: *x5u is the HTTP URL referencing to the public key that should be used to verify the signature;* One provider is asking to put the cert there, the other hasn't asked that yet. So i'm a little confused, should the x5u be the actual cert (with its intermediary?) or only the public key? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 9

Suppressing part of the xlog line
by Steve Wilkins 05 Nov '21

05 Nov '21

Hello, Is there a way to suppress the printing "date & time, host, and Kamailio execution path" at the begging of each xlog line? Example current xlog line: Nov 4 16:54:52 myserver /usr/local/sbin/kamailio[7021]: INFO: <script>: => Test xlog line I would like xlog line to be: INFO: <script>: => Test xlog line Thank you,

3 2

Is this idea even feasible?
by Antony Stone 04 Nov '21

04 Nov '21

Hi. I'm looking for a solution to the following problem, and wondering whether Kamailio (with which I have very little experience so far) may be it, or whether I'm thinking in totally the wrong direction. The problem: I have a SIP client application which runs on a standard Linux system, and is extremely basic - it can REGISTER in order to receive inbound calls, and it can send INVITEs in order to place outbound calls, and that's about it. It handles media as well - in essence, it's a simple softphone. I can't change it for a more capable one, before anyone suggests that :) Specifically, this thing cannot send REINVITEs in order to put calls on hold, nor can it handle anything to do with transfers (blind or attended). I'm looking for something which does have these SIP capabilities which I could put in between this application and the SIP server through which it is placing and receiving calls, so that I can (through some sort of API) cause calls in progress to be put on hold, resumed, blind transferred, or hold-and-attended- transferred. Does that sound like something Kamailio can do? If so, can anyone point me in a helpful direction re how to go about it? If not, can anyone suggest something else which could sit in between a very basic SIP client and a PBX server, in order to inject these sorts of commands into the path and thereby give me these sorts of extra call capabilities? Thanks, Antony. -- I'm not impossible, just highly implausible. Please reply to the list; please *don't* CC me.

8 30

STIR/SHAKEN, is a number format mandatory?
by David Villasmil 04 Nov '21

04 Nov '21

Hello guys, I'm getting failed by my provider because I'm sending to them with +1 both on the headers and on the payload. My understanding is there is no mandatory format, or is there? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

4 9

UAC reg_random_delay
by Sergiu Pojoga 03 Nov '21

03 Nov '21

Hi there, May I know what exactly is supposed to be "delayed" in the UAC registration? *modparam("uac", "reg_random_delay", 20)* I have records stored in DB with *reg_delay=0*. When doing *uac.reg_dump*, I do see the value of *reg_delay* being random for various records. However, after restarting Kamailio, the registrations are all sent out *simultaneously*, there doesn't seem to be a "delay", whatever the purpose is supposed to be. Can someone clarify please? Kamailio 5.5.0 Thanks. Regards, --Sergiu

4 9

STIR/SHAKEN
by David Villasmil 02 Nov '21

02 Nov '21

Hello guys, I'm trying to test secsipid by creating an Identity header and trying to verify it with the cli utility, but it's failing: # secsipidx -sign-full -orig-tn 123456789 -dest-tn 123456789 -attest A -x5u https://pki.my.domain.com/stir-shaken-public-key.pem -k ec256-private.pem eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9wa2kudG5jcC50ZXh0bm93LmNvbS9zdGlyLXNoYWtlbi1wdWJsaWMta2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjM0NTY3ODkiXX0sImlhdCI6MTYzNTg4OTY3Miwib3JpZyI6eyJ0biI6IjEyMzQ1Njc4OSJ9LCJvcmlnaWQiOiJjNTE4MDI4Ny05NjJmLTRlMDItYTM1NS00YmMzY2JhYzMxNjgifQ.ei1nQY8c9EQzvuBP7pQSI03T2yIfsNg9blzSdufU7tQy0tLJwoqmM9sBAk_Uq9s2jycVxseonZ1yl1kGiSv_GA;info=< https://pki.my.domain.com/stir-shaken-public-key.pem>;alg=ES256;ppt=shaken root@sip-stir-term1:/home/admin# echo "eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiaHR0cHM6Ly9wa2kudG5jcC50ZXh0bm93LmNvbS9zdGlyLXNoYWtlbi1wdWJsaWMta2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMjM0NTY3ODkiXX0sImlhdCI6MTYzNTg4OTY3Miwib3JpZyI6eyJ0biI6IjEyMzQ1Njc4OSJ9LCJvcmlnaWQiOiJjNTE4MDI4Ny05NjJmLTRlMDItYTM1NS00YmMzY2JhYzMxNjgifQ.ei1nQY8c9EQzvuBP7pQSI03T2yIfsNg9blzSdufU7tQy0tLJwoqmM9sBAk_Uq9s2jycVxseonZ1yl1kGiSv_GA;info=< https://pki.my.domain.com/stir-shaken-public-key.pem>;alg=ES256;ppt=shaken" > identity.txt root@sip-stir-term1:/home/admin# secsipidx -check -fidentity identity.txt -fpubkey /etc/kamailio/secsipid/ec256-private.pem -expire 3600 error message: asn1: structure error: tags don't match (16 vs {class:0 tag:2 length:1 isCompound:false}) {optional:false explicit:false application:false private:false defaultValue:<nil> tag:<nil> stringType:0 timeType:0 set:false omitEmpty:false} tbsCertificate @2 not-ok any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 1

how to prevent the log_prefix in syslog acc cdrs?
by Karsten Horsmann 02 Nov '21

02 Nov '21

Hi List, i thought there could be a solution to disable the log_prefix for acc module - but i didn't found it. Is there a solution to supress log_prefix for acc like acc_json did that? I use kamailio 5.5.2 with that settings: log_name="kamailio" log_facility=LOG_LOCAL4 log_prefix="{$hdr(CSeq) $ci} " the acc module then logs the "log_prefix" - plus some module/function stuff: INFO: {1 INVITE SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc [acc.c:285]: acc_log_request(): ACC: transaction answered: timestamp=1635843776;utctime=2021-11-02 09:02:56;method=INVITE;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=echo;dst_domain=172.20.120.59;user_agent=;x_t_call_id=;x_routing=;x_billing= INFO: {1 ACK SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc [acc.c:285]: acc_log_request(): ACC: request acknowledged: timestamp=1635843776;utctime=2021-11-02 09:02:56;method=ACK;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=alice;dst_domain=172.20.120.51;user_agent=;x_t_call_id=;x_routing=;x_billing= INFO: {2 BYE SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050} acc [acc.c:285]: acc_log_request(): ACC: transaction answered: timestamp=1635843833;utctime=2021-11-02 09:03:53;method=BYE;from_tag=SDkk50d01-e8bc0cda;to_tag=XBr6apQ9HN4HD;call_id=SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050;code=200;reason=OK;src_user=bob;src_domain=sip100;src_ip=172.20.120.56;dst_ouser=alice;dst_user=alice;dst_domain=172.20.120.51;user_agent=;x_t_call_id=;x_routing=;x_billing= The acc_json module is able to log acc without log_prefix: {"time": "2021/11/02 10:02:56", "method": "INVITE", "from_tag": "SDkk50d01-e8bc0cda", "to_tag": "XBr6apQ9HN4HD", "callid": "SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050", "sip_code": "200", "sip_reason": "OK", "src_user": "bob", "src_domain": "sip100", "src_ip": "172.20.120.56", "dst_ouser": "alice", "dst_user": "echo", "dst_domain": "172.20.120.59", "user_agent": "", "x_t_call_id": "", "x_routing": "", "x_billing": ""} {"time": "2021/11/02 10:03:53", "method": "BYE", "from_tag": "SDkk50d01-e8bc0cda", "to_tag": "XBr6apQ9HN4HD", "callid": "SDkk50d01-1f7cc956b0b908e42d284189a11b44e7-ct46om0050", "sip_code": "200", "sip_reason": "OK", "src_user": "bob", "src_domain": "sip100", "src_ip": "172.20.120.56", "dst_ouser": "alice", "dst_user": "alice", "dst_domain": "172.20.120.51", "user_agent": "", "x_t_call_id": "", "x_routing": "", "x_billing": ""} And just for the records - how to get rid off the rsyslog prefix time/date stuff: # /etc/rsyslog.d/kamailio.conf # drop :msg from rtpengine with "timer run time" - debugging only: :msg, contains, "timer run time" ~ # Template only msg: $template myFormat,"%msg%\n" # log only to local* and NOT to /var/log/messages local6.*;local6.!=info -/var/log/platform/voip/kamailio/rtpengine.log local6.* stop local5.* -/var/log/platform/voip/kamailio/rtpengine-cdr.log; myFormat local5.* stop local4.* -/var/log/platform/voip/kamailio/kamailio.log local4.* stop local2.* -/var/log/platform/voip/kamailio/kamailio-cdr.log; myFormat local2.* stop local1.* -/var/log/platform/voip/kamailio/kamailio-cdr-json.log; myFormat local1.* stop -- Mit freundlichen Grüßen *Karsten Horsmann*

1 0

kamailio didnt run with acc_json (without dialog module) - kamailio 5.5.2
by Karsten Horsmann 02 Nov '21

02 Nov '21

Hi List, i want to try out acc_json (to syslog). But if i set up "cdr_enable", 1 - it breaks the startup of kamailio with that error (dialog module is not listed as dependency). Any hints? Nov 01 13:13:59 siprouter2 kamailio[22397]: INFO: acc_json [acc_json_mod.c:135]: mod_init(): janson version : 2.10 Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: acc_json [acc_json_mod.c:204]: mod_init(): can't load dialog API Nov 01 13:13:59 siprouter2 kamailio[22397]: ERROR: <core> [core/sr_module.c:943]: init_mod(): Error while initializing module acc_json (/usr/lib64/kamailio/modules/acc_json.so) #!ifdef WITH_DBTEXT loadmodule "db_text.so" #!endif #!ifndef WITH_DBTEXT loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "permissions.so" loadmodule "avpops.so" #!ifdef WITH_ACCJSON loadmodule "acc_json.so" #!endif #!ifdef WITH_DIALPLAN loadmodule "dialplan.so" modparam("dialplan", "db_url", DBURL) modparam("dialplan", "fetch_rows", 1000) modparam("dialplan", "attrs_pvar", "$avp(s:dest)") #!endif #!ifdef WITH_JSONSTATS loadmodule "xhttp.so" loadmodule "jansson.so" loadmodule "janssonrpcc.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif loadmodule "pipelimit.so" loadmodule "htable.so" .... # ----- acc params ----- modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) # ----- acc params ----- #!ifdef WITH_ACCSYSLOG modparam("acc", "log_facility", "LOG_LOCAL2") modparam("acc", "log_level", 2) #!endif /* what special events should be accounted ? */ modparam("acc", "time_mode", 4) modparam("acc", "time_attr", "utctime") modparam("acc", "time_format", "%Y-%m-%d %H:%M:%S") modparam("acc", "acc_time_column", "localtime") ## --- to syslog modparam("acc", "early_media", 1) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;user_agent=$ua;x_t_call_id=$avp(x-t-call-id);x_routing=$avp(x_routing);x_billing=$avp(x_billing)") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif #!ifdef WITH_ACCJSON modparam("acc_json", "acc_missed_flag", 3) modparam("acc_json", "acc_time_mode", 1) modparam("acc_json", "cdr_output_syslog", 1) modparam("acc_json", "acc_time_format", "%Y/%m/%d %H:%M:%S") modparam("acc_json", "acc_log_facility", "LOG_LOCAL1") modparam("acc_json", "acc_log_level", 2) # Set acc_log_level to 2 (L_INFO) modparam("acc_json", "cdr_enable", 1) #!endif .... version: kamailio 5.5.2 (x86_64/linux) 55e232 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 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: 55e232 compiled on 14:20:31 Aug 25 2021 with gcc 4.8.5 -- Mit freundlichen Grüßen *Karsten Horsmann*

2 3

Strip VP8 from SDP in rtpengine
by Rahul mm 30 Oct '21

30 Oct '21

Hello, I'm facing trouble with Android 12 using webview to attempt video call. All the media is rotated from rtpengine. To workaround this problem, I wish to strip VP8 and have the calls run on VP9. For this, I added "codec-strip=VP8" but rtpengine is not able to remove it. I've even tried "codec-strip=VP8/90000" but to no avail. The video is still garbled. Am I doing anything wrong here? Thanks and Regards Rahul

4 4

Stir/Shaken Kamailio modules - stirshaken vs secsipid
by Joel Serrano 29 Oct '21

29 Oct '21

Hi everyone, As most of us, I'm starting to implement STIR/SHAKEN in our platform. I've seen there are 2 modules available in Kamailio for this: stirshaken: https://www.kamailio.org/docs/modules/5.5.x/modules/stirshaken.html secsipid: https://kamailio.org/docs/modules/5.5.x/modules/secsipid.html Suggestions on which to use? Pros/Cons of each? Do any of you have experience with any of the two? Any comments are more than welcome! Thanks, Joel.

1 0

RTPENGINE [ice] Priority collision between candidate pairs
by keffa＠alcall.net 29 Oct '21

29 Oct '21

Hello @everyone Please can you explain me how this message is errors or not ? « Oct 29 08:34:44 ip76 rtpengine[649]: WARNING: [YGiXZ1MmIA port 30104]: [ice] Priority collision between candidate pairs 9oZd7sWNVguf6cgl:e4b19f6635447b850 » thank

5 4

htable shtcn expressions
by Brooks Bridges 29 Oct '21

29 Oct '21

Referring to the expressions outlined at http://www.kamailio.org/wiki/cookbooks/5.4.x/pseudovariables#shtcn_htable_e… , is there any additional documentation or examples floating around that further explains how the right and left prefix matching works? The rest of them are pretty clear, but I'm having troubles with these and can't get them working like I would expect them to work. Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

3 5

Configuration script behaviour with http_async_client usage
by Kris Domagalski 29 Oct '21

29 Oct '21

Hi, I'm new in Kamailio and looking for some configuration guidance. I'm using http_async_client for API requests. It's configured to use TEST2 route, which should be executed upon reception of HTTPS response. After receiving the response I would like to go back to the request_route and perform some more actions. Currently behaviour is, in the syslog I can see only entries which I push from TEST2 route (TEST 123). Cannot see the one from request route. Current syslog output: TEST123 Expected syslog output: TEST 123 TEST 789 And now my question is why kamailio is not executing remaining actions (xlog) in request_route? request_route { route(TEST1); xlog("L_INFO","TEST 789\n"); } route[TEST1] { # define API request body in JSON format jansson_set("string", "from", $fU, "$var(http_routing_query)"); # set API request parameters $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = 100; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = $var(http_routing_query); t_newtran(); # send API request http_async_query(http://local:3000/test, "TEST2"); } route[TEST2] { xlog("L_INFO","TEST 123\n"); } Thanks Kris

1 0

Re: [SR-Users] No "child_destroy" export - issue with module flatstore and flush=0
by Chaigneau, Nicolas 29 Oct '21

29 Oct '21

Hello Henning, Thanks for your reply ! For the record, I'm using Kamailio module "flatstore" with the flush deactivated. In this situation, when Kamailio is stopped, I'm losing the remaining data in files not yet flushed. So my idea was, when destroy is called, to iterate over the connection pools, and flush/close all the opened files properly. But I found out that it's not that simple :/ Do you think this is something that could be fixed in Kamailio ? Regards, Nicolas. De : Henning Westerholt <hw(a)gilawa.com> Envoyé : mardi 26 octobre 2021 20:38 À : Kamailio (SER) - Users Mailing List Cc : Chaigneau, Nicolas Objet : RE: No "child_destroy" export ? Hi Nicolas, you are right, there is no special function for children destroy. Most modules do not need it, as the data is in shared memory, processes are started over standard kamailio APIs which will be stopped from the core automatically, to give some background. Some modules do it by just iterating over an internal list of their children and then do the cleanup task. You can find an example in cdp module (destroy, diameter_peer_destroy). 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 Chaigneau, Nicolas Sent: Tuesday, October 26, 2021 7:14 PM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] No "child_destroy" export ? Hello, I have a module, for which I need to perform some processing in the children processes when Kamailio is being stopped. I've looked at the "struct module_export", in which I see "mod_init", "child_init", "mod_destroy"... but no "child_destroy". :/ Is there a way to achieve this ? Thanks for your help. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

Does failure_route for redirection covers all error cases ?
by Liviu ANDRON 29 Oct '21

29 Oct '21

Hello, We are routing calls to FreeSWITCH servers. We have a failure routing mechanism in place, which looks pretty common from my research around: route[INVITE] { ... t_on_failure("REROUTE"); and failure_route[REROUTE] { if (t_is_canceled()) { exit; } // also 6xx ? if (t_check_status("5[0-9][0-9]") or (t_branch_timeout() and !t_branch_replied())) { // re-route to another available FreeSWITCH server } } The problem is that we don't capture all the failures we would like to, one such example being 480 sent by FreeSWITCH in various cases (even default in hangup_cause_to_sip, mod_sofia.c, https://github.com/signalwire/freeswitch/blob/master/src/mod/endpoints/mod_…), like a Python script crash, but also legitimate cases like "user not registered" (USER_NOT_REGISTERED). Are there other cases? Like a timeout replied (t_branch_timeout() and t_branch_replied()) which never recovers ? Shouldn't any t_branch_timeout() re-arm the failure route ? Thanks, Liviu

3 2

xhttp_pi show dr_rules buffer 2 small Error.
by Abdirahman Osman 28 Oct '21

28 Oct '21

Hello Everyone, I am using the xhttp_pi modules to see the list of rules on the browser http://172.28.253.17:5060/pi/dr_rules/show?cmd=on and am I getting Internal Server Error I checked the logs and noticed the error "ph_build_reply_footer(): buffer 2 small" 13(20) DEBUG: db_postgres [km_dbase.c:411]: db_postgres_free_query(): PQclear(0x55e2416ae9a0) result set 13(20) ERROR: xhttp_pi [xhttp_pi_fnc.c:2470]: ph_build_reply_footer(): buffer 2 small 13(20) DEBUG: xhttp_pi [xhttp_pi.c:159]: pi_send(): pi_fault(500,"Internal Server Error" 13(20) DEBUG: xhttp_pi [xhttp_pi.c:169]: pi_send(): xhttp_api.reply(0x7ffc39eb1410, 500, Internal Server Error, text/html, Internal Server Error) 13(20) DEBUG: xhttp [xhttp_mod.c:416]: xhttp_send_reply(): response with content-type: text/html 13(20) DEBUG: xhttp [xhttp_mod.c:424]: xhttp_send_reply(): response with body: Internal Server Error 13(20) DEBUG: xhttp [xhttp_mod.c:426]: xhttp_send_reply(): sending out response: 500 Internal Server Error Which buffer do I need to increase here? tried to increase sql_buffer_size but that didn't work for me. Thanks! Abdirahman A. Osman

2 2

Info: ClueCon 2021 - Sessions touching Kamailio
by Daniel-Constantin Mierla 27 Oct '21

27 Oct '21

Hello, ClueCon conference happens this week, today I will have a session about what Kamailio offers for changing SIP message content (starting in about 25 minutes) and later today Fred Posner will present about Kamailio and FreeSwitch. You can watch the even live, more details at: * https://www.cluecon.com/ 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/

2 1

t_relay changing source IP?
by Brooks Bridges 27 Oct '21

27 Oct '21

So we have a box that's bound to all interfaces on the server, and one specific IP address (that is an HA IP), and I've run into a bit of an odd behavior that I cannot explain. For 99.9999% of our traffic, the invites come into the HA IP, and then are forwareded out the HA IP to an SBC for further processing. In a very rare instance however, I have found that there is an invite coming in the HA IP, and for some unknown reason it gets forwarded out the *interface* ip instead of the HA IP. For example: Working: 1.2.3.4 -- invite --> 9.8.7.6 (ha IP) -- invite --> 5.6.7.8 (sbc) Non working: 1.2.3.4 -- invite --> 9.8.7.6 (ha IP) *something happens here* 9.8.7.5 (interface IP) -- invite --> 5.6.7.8 (sbc) I've been through tm, tmx, and the incoming invite of a working and non working request with a fine toothed comb and I cannot for the life of me explain this behavior. Any thoughts? Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

4 11

Use constants in attrs column of the dispatcher module
by Markus Bönke 27 Oct '21

27 Oct '21

Hello all, I’ve a question regarding the dispatcher module. So far we use on every server a separate database table, where the attribute column is set to the local internal or external socket like "socket=udp:192.168.1.70:5060“ or socket=udp:80.239.xxx.xxx:5060. Is there a way to use constants here like socket=udp:IPADDR_INTERN:5060 or socket=udp:IPADDR_EXTERN:5060 which are replaced when the data is loaded from the DB? Thanks and regards Markus

3 4

No "child_destroy" export ?
by Chaigneau, Nicolas 26 Oct '21

26 Oct '21

Hello, I have a module, for which I need to perform some processing in the children processes when Kamailio is being stopped. I've looked at the "struct module_export", in which I see "mod_init", "child_init", "mod_destroy"... but no "child_destroy". :/ Is there a way to achieve this ? Thanks for your help. Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 1

Kamailio/Asterisk Realtime issues
by Hayden Ryan 26 Oct '21

26 Oct '21

Hi all, I have Kamailio and Asterisk set up in a realtime configuration, like this: https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb In my setup, Kamailio forwards all REGISTER requests to Asterisk, and Asterisk authenticates them. I have a single peer in sip.conf pointing towards the Kamailio server, as type=peer Each user in the Asterisk database is set to type=user, as when they are set to anything else (or blank), incoming calls from Kamailio to Asterisk display the name of the most recent extension to register. Is there something wrong with my setup here? What I am trying to do is to get MWI working. SUBSCRIBEs for MWI are forwarded to and handled by Asterisk correctly. Yet the MWI state on the handset is incorrect, as I believe it is sending the latest MWI update to the most recently registered extension. Any suggestions would be much appreciated. I assume I'm doing this the wrong way and shouldn't be forwarding any REGISTERs to Asterisk and should be handling authentication in Kamailio itself? But that seems incompatible with the realtime configuration guides I've seen. Thank you, Hayden

2 1

Please reivew the issue about diameter_request_async
by BOEINGNG 26 Oct '21

26 Oct '21

Hello! Please review the issue I mentioned here:https://github.com/kamailio/kamailio/issues/2900 <https://github.com/kamailio/kamailio/issues/2900> Could you provide some advice to me? Regrads HANG LI

1 0

Info: risk of failure by upgrading to MariaDB 10.3.31
by Daniel-Constantin Mierla 25 Oct '21

25 Oct '21

Hello, not strictly related to Kamailio, but I know a large number of its deployments run with MariaDB. Be aware of a (pretty high) risk of failure after upgrading MariaDB server to v10.3.31, which comes now on Debian 10 via the usual apt update+upgrade. I was hit on one the servers I upgraded recently, first one worked fine, the 2nd failed to start MariaDB service after upgrade - there is a bug report about it: * https://www.mail-archive.com/debian-bugs-dist@lists.debian.org/msg1823336.h… Downgrading MariaDB seems to be the right solution for now, specially if you didn't do a backup just before upgrade. Don't try other tricks without a fresh backup, data files can be damaged. If you have a fresh backup, there seems to be a solution by deleting database files to start with a clean set and then restore from backup with sql statements. FreeBSD has a similar bug report, so doesn't seem to be only Debian, could be other Linux distros as well. Even with fresh backups, plan for a longer maintenance time frame than usual. 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/

1 0

T_req(pv) not accessible in dialog event route in v5.5.2
by Joel Serrano 23 Oct '21

23 Oct '21

Hello, I'm in the process of updating some servers that are running Kam v5.1.4 to v5.5.2. After the upgrade I've noticed that I've lost some custom metrics that we have defined via config-script. Snippet of the relevant config: event_route[dialog:start] { # Gather metrics regarding source IP xlog("L_WARN", "[DLG] - Checking country for ($si | $T_req($si)) - START\n"); if (geoip2_match("$T_req($si)", "src")) { # Country found ... } xlog("L_WARN", "[DLG] - Checking country for ($si | $T_req($si)) - END\n"); } NOTE: xlog lines added for debugging, they normally are not there. In version 5.1.4 in the logs I see: Oct 21 15:55:47 13cn3 sbc[22606]: WARNING: <script>: [DLG] - Checking country for (A.B.C.D | 186.185.186.7) - START Oct 21 15:55:47 13cn3 sbc[22606]: WARNING: <script>: [DLG] - Checking country for (A.B.C.D | 186.185.186.7) - END The same config, on v5.5.2 is returning: Oct 21 15:57:08 sbc0504 sbc[22606]: WARNING: <script>: [DLG] - Checking country for (A.B.C.D | <null>) - START Oct 21 15:57:08 sbc0504 sbc[22606]: WARNING: <script>: [DLG] - Checking country for (A.B.C.D | <null>) - END After enabling debug logs I see: WARNING: <script>: [DLG] - Checking country for (198.1.55.124 | <null>) - START *DEBUG: tmx [t_var.c:309]: pv_get_t_var_req(): used in unsupported route block - type 64*DEBUG: geoip2 [geoip2_pv.c:488]: geoip2_update_pv(): attempt to match: <null> DEBUG: geoip2 [geoip2_pv.c:491]: geoip2_update_pv(): no match for: <null> Did something change in the tmx module that could affect $T_req(pv) being accessible in the dialog:start event_route? Any alternatives to get the source IP of the request (not the reply) in the dialog:start event_route? Let me know if you prefer that I open a ticket for this in GH. Thanks, Joel.

2 3

"EXTERN.h: No such file or directory" error when compiling Kamailio 5.5 on Centos 7
by AL RSM 22 Oct '21

22 Oct '21

Hi, While trying to install Kamailio 5.5 on Centos 7 I am getting the following error. Perl is already installed. I verified "EXTERN.h" does not exist on the VM. # make include_modules="tm app_perl" cfg # make prefix=/ all .... .... Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. Can't locate ExtUtils/Embed.pm in @INC (@INC contains: /usr/local/lib64/perl5 /usr/local/share/perl5 /usr/lib64/perl5/vendor_perl /usr/share/perl5/vendor_perl /usr/lib64/perl5 /usr/share/perl5 .). BEGIN failed--compilation aborted. CC (gcc) [M app_perl.so] perlfunc.o In file included from perlfunc.c:40:0: app_perl_mod.h:37:20: fatal error: EXTERN.h: No such file or directory #include <EXTERN.h> ^ compilation terminated. make[2]: *** [perlfunc.o] Error 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/kamailio/src' Your help would be much appreciated. Thanks, AL

2 1

tls connections
by Richard Robson 20 Oct '21

20 Oct '21

I'm trying in vain to get a tls connection to a carrier. ~Without too much detail we are exchanging certificates and they are saying we are triggering a reset of the connection because we are requesting to chage the tcp window size. there are requiring us to have a window size of 65535. there we are sending a smaller window size. as far as I understand this should be OK. I have tried to force out linux install to set the window size to 65535 but we are still sing the smallerwindow in the tcp packet. Is there any settings in Kamailio that could force the window size? we are using Kamailio 5.4.2 Thanks in advance Richard

2 2

Catching websocket / TCP errors
by Marrold 20 Oct '21

20 Oct '21

Hi, We're currently implementing WebRTC (SIP over a websocket) on our Kamailio proxies and we're experiencing a minor issue after a restart. We write all USRLOC entries to a shared database to implement HA. These are then loaded on a restart, which leaves us with entries with TCP connections that no longer exist. When a call attempt is made the proxy can't find the connection ID and immediately rejects the call with a "500 No error (2/SL)" - Oct 5 15:38:24 proxy-01 /sbin/kamailio[32087]: WARNING: <core> [core/msg_translator.c:2861]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Oct 5 15:38:24 proxy-01 /sbin/kamailio[32087]: ERROR: <core> [core/msg_translator.c:2038]: build_req_buf_from_sip_req(): could not create Via header Oct 5 15:38:24 proxy-01 /sbin/kamailio[32087]: ERROR: tm [t_fwd.c:479]: prepare_new_uac(): could not build request Oct 5 15:38:24 proxy-01 /sbin/kamailio[32087]: ERROR: tm [t_fwd.c:1759]: t_forward_nonack(): failure to add branches Oct 5 15:38:24 proxy-01 /sbin/kamailio[32087]: ERROR: sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: No error (2/SL) How can we catch this error to avoid terminating the call prematurely? Or should we flush these TCP connections on start up? Thanks Matthew

2 5

Use constants in attrs column of the dispatcher module
by Markus Bönke 19 Oct '21

19 Oct '21

Hello all, I’ve a question regarding the dispatcher module. So far we use on every server a separate database table, where the attribute column is set to the local internal or external socket like "socket=udp:192.168.1.70:5060“ or socket=udp:80.239.xxx.xxx:5060. Is there a way to use constants here like socket=udp:IPADDR_INTERN:5060 or socket=udp:IPADDR_EXTERN:5060 which are replaced when the data is loaded from the DB? Thanks and regards Markus

1 0

Kamailio on AWS Kubernetes with NLB
by Cedric Thiebault 18 Oct '21

18 Oct '21

Hello, I'm trying to deploy Kamailio on Kubernetes (AWS EKS with Network Load Balancer) but I'm not able to register using TCP. I'm using official Docker image 5.5.0-buster. I did some tests with Docker Compose and it worked fine, but on K8S it does not :-( Here are my logs: Listening on udp: 127.0.0.1:5060 udp: 172.31.4.29:5060 udp: 172.31.0.221:5060 tcp: 127.0.0.1:5060 tcp: 172.31.4.29:5060 tcp: 172.31.0.221:5060 Aliases: tcp: ip-172-31-0-221.eu-central-1.compute.internal:5060 tcp: 172-31-4-29.kube-prometheus-stack-prometheus-node-exporter.infra-prometheus.svc.cluster.local:5060 tcp: localhost4.localdomain4:5060 tcp: localhost4:5060 tcp: localhost.localdomain:5060 tcp: localhost:5060 udp: ip-172-31-0-221.eu-central-1.compute.internal:5060 udp: 172-31-4-29.kube-prometheus-stack-kubelet.kube-system.svc.cluster.local:5060 udp: localhost4.localdomain4:5060 udp: localhost4:5060 udp: localhost.localdomain:5060 udp: localhost:5060 0(1) INFO: <core> [core/tcp_main.c:4983]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(1) INFO: rr [../outbound/api.h:52]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(1) INFO: rr [rr_mod.c:188]: mod_init(): outbound module not available 0(1) INFO: <core> [main.c:3030]: main(): processes (at least): 41 - shm size: 67108864 - pkg size: 8388608 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(1) INFO: <core> [core/udp_server.c:154]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(1) INFO: <core> [core/udp_server.c:206]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 29(35) INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/35 31(37) INFO: ctl [io_listener.c:213]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 35(41) INFO: <core> [core/parser/parse_fline.c:147]: parse_first_line(): method not followed by SP 35(41) ERROR: <core> [core/parser/parse_fline.c:271]: parse_first_line(): parse_first_line: bad message (offset: 0) 35(41) ERROR: <core> [core/parser/msg_parser.c:747]: parse_msg(): ERROR: parse_msg: message=<> 35(41) ERROR: <core> [core/receive.c:374]: receive_msg(): core parsing of SIP message failed (194.230.146.66:9055/2) And my service: apiVersion: v1 kind: Service metadata: name: kamailio-tcp annotations: service.beta.kubernetes.io/aws-load-balancer-nlb-target-type: ip service.beta.kubernetes.io/aws-load-balancer-proxy-protocol: '*' service.beta.kubernetes.io/aws-load-balancer-scheme: internet-facing service.beta.kubernetes.io/aws-load-balancer-target-group-attributes: preserve_client_ip.enabled=true service.beta.kubernetes.io/aws-load-balancer-type: external labels: app.kubernetes.io/instance: test-sip app.kubernetes.io/name: kamailio-tcp app.kubernetes.io/version: 5.5.0-buster spec: type: LoadBalancer externalTrafficPolicy: Local ports: - name: sip-tcp nodePort: 30567 port: 5060 protocol: TCP targetPort: 5060 selector: app.kubernetes.io/instance: test-sip app.kubernetes.io/name: kamailio I have a Netwok Load Balancer create with a listener for TCP 5060 but when I try to connect with Zoiper or Linphone, it does not work :-( I used tcpdump on my pod and it received well some SIP commands. Any ideas of what could be wrong here? Thanks in advance! Cedric

2 5

Routing dialogs (INVITE) from Lua
by Emilio Panighetti 18 Oct '21

18 Oct '21

Hello, Based on kamailio-basic-kemi.cfg and kamailio-basic-kemi-lua.lua, I’m trying to get a simple test to go through using SIPp built-in UAC and UAS scenarios, using LUA config: Commented out all the registration blocks and trying to run a simple proxy for now, but I can’t seem able to change the URI. SIPp UAC: 192.168.1.12:5060 SIPp UAS: 192.168.1.12:5062 Kamailio 5.5.2: 192.168.15.:5060 Part of the Kamailio.lua config: -- wrapper around tm relay function function ksr_route_relay() -- enable additional event routes for forwarded requests -- - serial forking, RTP relaying handling, a.s.o. if KSR.is_method_in("IBSU") then if KSR.tm.t_is_set("branch_route")<0 then KSR.dbg("EP: ksr_route_relay Set for branch_route\n"); KSR.tm.t_on_branch("ksr_branch_manage"); end end if KSR.is_method_in("ISU") then if KSR.tm.t_is_set("onreply_route")<0 then KSR.dbg("EP: ksr_route_relay Set for onreply_route\n"); KSR.tm.t_on_reply("ksr_onreply_manage"); end end if KSR.is_INVITE() then if KSR.tm.t_is_set("failure_route")<0 then KSR.dbg("EP: ksr_route_relay Set for failure\n"); KSR.tm.t_on_failure("ksr_failure_manage"); end end if KSR.tm.t_relay()<0 then KSR.dbg("EP: ksr_route_relay Set for reply_error\n"); KSR.sl.sl_reply_error(); end -- EP: Base route KSR.dbg("EP: ksr_route_relay Check for Base Route\n"); if KSR.is_INVITE() then KSR.dbg("EP: ksr_route_relay Base Route (DU)\n"); --KSR.sl.send_reply("503", "Debug"); KSR.setdsturi("sip:192.168.1.12:5062"); KSR.pv.sets("$du", "sip:192.168.1.12:5062"); end KSR.dbg("EP: ksr_route_relay exit\n"); KSR.x.exit(); end What’s happening here is that instead of forwarding the INVITE to the SIPp UAC, it creates a loop. Tried KSR.setdsturi() only (KSR.pv.sets commented out) and it’s still the same. If I comment out the 503 reply, then that’s what the UAC gets back. Here’s the log when calling ksr_route_relay: Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/msg_translator.c:162]: check_via_address(): (192.168.1.12, 192.168.1.12, 0) Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:489]: sr_lua_reload_script(): No need to reload [/etc/kamailio/kamaili o.lua] is version 0 Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:673]: app_lua_run_ex(): executing Lua function: [[ksr_onsend_route]] Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:675]: app_lua_run_ex(): lua top index is: 0 Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: tm [t_funcs.c:376]: t_relay_to(): new transaction forwarded Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_r oute_relay Check for Base Route Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_relay Check for Base Route Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_r oute_relay Base Route (DU) Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_relay Base Route (DU) Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: setdsturi is str: sip :192.168.1.12:5062 Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: sets is str: $du Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[1] for: sets is str: sip:192. 168.1.12:5062 Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/kemi.c:2906]: sr_kemi_pv_sets(): pv set: $du Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/pvapi.c:371]: pv_cache_add(): pvar [$du] added in cache Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1003]: sr_kemi_lua_exec_func_ex(): param[0] for: dbg is str: EP: ksr_r oute_relay exit Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/kemi.c:85]: sr_kemi_core_dbg(): EP: ksr_route_relay exit Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:1837]: sr_kemi_lua_exit(): script exit call Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_api.c:729]: app_lua_run_ex(): ksr error call from Lua: ~~ksr~exit~~ Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: app_lua [app_lua_mod.c:167]: sr_kemi_config_engine_lua(): execution of route type 1 with no n ame returned 1 Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) INFO: LUA {INVITE}: <core> [core/receive.c:515]: receive_msg(): request-route executed in: 1335 usec Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/usr_avp.c:637]: destroy_avp_list(): destroying list (nil) Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/xavp.c:541]: xavp_destroy_list(): destroying xavp list (nil) Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: LUA {INVITE}: <core> [core/receive.c:606]: receive_msg(): cleaning up Oct 17 23:42:48 ol-vm1 kamailio[2416]: 3(2419) DEBUG: <core> [core/udp_server.c:493]: udp_rcv_loop(): received on udp socket: (112/100/661) [[INVITE sip:service@ 192.168.1.15:5060 SIP/2.0 0D 0A Record-Route: <sip:192.168.1.15;lr=on> 0D 0A Via: SIP/2.0/U]] Any idea how we can route this message? Thanks!

2 1

dmq_usrloc writes to database on other nodes too
by Maarten Ureel 17 Oct '21

17 Oct '21

I have the idea that if you enable dmq_usrloc, the other nodes that receive the usrloc info, also try to write to database. We see this by a lot of errors indicating the index violation on postgres. What is the easiest way to tell Kamailio that it does not need to store the usrloc in database if the data came through a DMQ message? Currently I have in my config: # DMQ processing if(is_method("KDMQ") && $Rp == 5090) { dmq_handle_message(); } I have understood that no further processing happens then. So is my assumption correct that my other handling of REGISTER will not be used then? As in, can you handle this in code that it will only store the register information in memory? Like this? # Registration handling route[REGISTRAR] { # If coming from DMQ, don't save to database if(dmq_is_from_node()) { save("location", "0x01") } }

2 4

kamailio SipTrace Module
by Андрей Зюзенков 17 Oct '21

17 Oct '21

Good afternoon, I apologize for disturbing you. I have a question that I cannot find in the documentation https://www.kamailio.org/docs/modules/devel/modules/siptrace.html#siptrace.… 3.3. trace_flag parameter has some values. And what values can be and what do they mean pls, help

2 1

http_async_client "route_name" is considered a failure route?
by Brooks Bridges 16 Oct '21

16 Oct '21

So I've been working on this for a while, and I cannot seem to find a way around this. When using the http_async_client to request data from an external API, the returned data in the "route_name" target is being treated as a failure route, so most of the subsequent processing and messaging that I want to do is not working, either because the function can't be used in a failure route, or because I have to create a transaction prior to doing the request, which then throws an error by immediately triggering a 500 after I return the response I want (via 302 to the requesting client). Am I using http_async_client incorrectly somehow that's causing this behavior or is it expected that the response route is treated as a failure route? Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

2 1

Call to MS Teams
by Володимир Іванець 15 Oct '21

15 Oct '21

Hello all! I'm trying to make a call from the system with Kamailio to a MS Teams. I can answer in the Teams client, system will receive response 200 OK and audio streams will be started. The problem I have is that after a short period of time Teams sends BYE with the reason "An acknowledgement was not received for the call acceptance in the allotted time". If I try to hangup from the system side, Teams will receive BYE and respond with "481 Call Leg/Transaction Does Not Exist" - "Call leg unavailable". It looks like Teams can not match and accept ACK request to finish establishing the call and the following requests have the same problem. I was wondering if someone had this problem before or can share a SIP packet capture (of course with hidden addresses and domain names) of a correctly established and completed call to Teams? Thanks a lot! Regards, Volodymyr Ivanets

1 1

unable to receive UDP
by dem＠htsweb.com 13 Oct '21

13 Oct '21

After 4 (long) days of trying to receive any UDP SIP, I'm hoping someone can point out what I'm doing wrong. I've looked at the kamailio code. I'm beginning to think it's the 'polling' code, but it looks fine. It could be Centos or a Dell network driver, but not seeing any errors anywhere. - stock build/compile... Version: kamailio 5.5.2 (x86_64/linux) 0d53d9 Compile 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 MAX_RECV_BUFFER_SIZE=262144 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 0d53d9 Compiled with: gcc 4.8.5 Compiled architecture: x86_64 Compiled on: 08:44:19 Oct 11 2021 Running Centos 7 all updated. Dell R630. # sestatus SELinux status: disabled # firewall-cmd --list-all ports: 5666/tcp 873/tcp 873/udp 80/tcp 443/tcp 5060/tcp 5060/udp 5061/tcp 5061/udp - Just trying to send SIP INVITE from trunk provider to R630 kamailio... (SIP trunk) UDP 172.200.200.202:5060 -> (firewall) UDP 172.250.250.162:5060 -> 1:1 NAT -> (R630) UDP 10.102.88.81:5060 - I can 'see' the packet hit the em3 network... # tcpdump -i em3 -e -vv -n port 5060 11:53:33.260892 99:99:ef:3e:69:c1 > 99:99:1c:48:4e:e3, ethertype IPv4 (0x0800), length 1366: (tos 0x0, ttl 54, id 20430, offset 0, flags [none], proto UDP (17), length 1352) 172.200.200.202.sip > 10.102.88.81.sip: [udp sum ok] SIP, length: 1324 INVITE sip:+18005551234@172.250.250.162:5060 SIP/2.0 ... At this point, I just need to see a SIP INVITE received. Then I can go back to working on a real config. I've tried with TCP on/off, listening on all networks, alias on/off, strace and everything else I could think of.Truly appreciate any guidance. Tried every sample config possible, but I just need to test receiving the SIP UDP packet, so this 'should' be minimum config... #!KAMAILIO debug=9 log_stderror=no log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " children=2 disable_tcp=yes udp4_raw=on auto_aliases=no alias="sbc01.mysite.com" listen=udp:10.102.88.81:5060 loadmodule "debugger.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" modparam("debugger", "cfgtrace", 1) request_route { xlog("SIP received xxxxxxxxxxxxxxxxxxx\n"); } # kamcmd ps 11578 main process - attendant 11579 udp receiver child=0 sock=10.102.88.81:5060 11580 udp receiver child=1 sock=10.102.88.81:5060 11581 slow timer 11583 timer 11584 secondary timer 11585 JSONRPCS FIFO 11586 JSONRPCS DATAGRAM 11587 ctl handler - I get plenty of logging, but the logging just waits here... Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: INFO: ctl [io_listener.c:210]: io_listen_loop(): io_listen_loop: using epoll_lt as the io watch method (auto detected) Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7fd2a8bae010, 4800) called from core: core/io_wait.c: init_io_wait(469) Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10059]: DEBUG: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd2a8bae010, 65456) returns address 0x7fd2a8c652d8 frag. 0x7fd2a8c652a0 (size=65456) on 1 -th hit Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd2a8bae010, 4800) returns address 0x7fd2a8c652d8 frag. 0x7fd2a8c652a0 (size=4800) on 1 -th hit Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7fd2a8bae010, 2400) called from core: core/io_wait.c: init_io_wait(516) Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7fd2a8bae010, 2400) returns address 0x7fd2a8c66600 frag. 0x7fd2a8c665c8 (size=2400) on 1 -th hit Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: ctl [io_listener.c:246]: io_listen_loop(): io_listen_loop: adding socket 8, type 2, transport 3 (/var/run/kamailio//kamailio_ctl) Oct 12 11:38:04 sbc01 /usr/local/sbin/kamailio[10060]: DEBUG: ctl [../../core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0x7fd2a70a5360, 8, 2, 0x2439960), fd_no=0 - The only other thing that I see is transport 3. Isn't that a UNIXS_SOCK?

5 9

Kamailio Inbound proxy to Asterisk - ACL Filtering
by Mihai Cezar 13 Oct '21

13 Oct '21

Hello, I have an issue with filtering on the asterisk side, my requests are: UsersPhones(bria) -> Kamailio -> Asterisk -> Sip Trunk Out. The goal is to manage a new layer of protection ( IP filtering / Whitelisting ). When I try to compile a list of Whitelisted IP in sip.conf I get this error: NOTICE[205]: acl.c:748 ast_apply_acl: SIP contact ACL: Rejecting '145.72.23.45' due to a failure to pass ACL '(BASELINE)' WARNING[205]: chan_sip.c:17061 parse_register_contact: Domain '5.12.16.2:48669' disallowed by contact ACL (violating IP 145.72.23.45) WARNING[205]: chan_sip.c:17933 register_verify: Registration denied because of contact ACL The IP 145.72.23.45, is the proxy kamailio and if I added it to sip.conf it works, but so does every ip afterwards. I tried with contactpermit also with permit, the result is the same as long as I permit the proxy ip it works. Is there something that I can do on the asterisk side to activate this filtering Or there is something that I can do in Kamailio so it will forward the realip ? contactdeny=0.0.0.0/0.0.0.0 contactpermit=145.72.23.45/32 contactpermit=5.12.16.2/32 Thanks in advance,

4 7

Compile kamailio-5.4 against openssl1.1.1
by Antony Russell 13 Oct '21

13 Oct '21

Hello I see that support for compiling kamailio against openssl1.1.1 was recently added (https://github.com/kamailio/kamailio/pull/2779) and I see that this can be done in kamailio-5.5.2. Will this ability be made available in kamilio-5.4? I see it is not in the most recent 5.4 release (5.4.7) -- *Antony Russell* Chief Technical Officer Tel 087 820 8113 antony.russell(a)telviva.co.za www.telviva.co.za *Telviva (Pty) Ltd - formerly Connection Telecom (Pty) Ltd* -- https://www.telviva.co.za/legal/email-disclaimer <https://www.telviva.co.za/legal/email-disclaimer>

1 0

Fail Call from Webrct to PSTN SIP Trunk
by Abdul Rahman Reza 11 Oct '21

11 Oct '21

Hi Everyone There some log kamailio Warning: 399 03077.08860.B.005.422.228.0.15.01039.00000000.00000 "SDP body length beyond limit" how to increase maximum length sdp in kamailio, because i don't see any configuration related to sizing max length SDP on kamailio documentation please help me Thank you very much

3 3

Let's Encrypt DST Root CA X3 cert CA expiration 30th/Sept - Any issues?
by Joel Serrano 11 Oct '21

11 Oct '21

Hello, I'm wondering if anyone had any issues yesterday with the expiration of the DST Root CA X3 cert? Out of all the servers I manage, only a couple were affected (debian 8). They were production servers so we replaced the cert with a different one to solve the issue while we find the root cause. Anyone out there had any issues yesterday because of this? I'm just curious! Joel.

10 12

upgrade kamailio 4.4.x-to-5.0.0 - PostgreSQL migrations
by Giovanni Tommasini - timenet srl 11 Oct '21

11 Oct '21

Hi Dears, I'm working on upgrading Kamailio from 4.4.6 to 5.4.0 in an installation where I use PostgreSQL I understand that I have to upgrade in steps by following these links: - http://www.kamailio.org/wiki/install/upgrade/4.4.x-to-5.0.0 - http://www.kamailio.org/wiki/install/upgrade/5.0.x-to-5.1.0 - https://www.kamailio.org/wiki/install/upgrade/5.1.x-to-5.2.0 - https://www.kamailio.org/wiki/install/upgrade/5.2.x-to-5.3.0 - https://www.kamailio.org/wiki/install/upgrade/5.3.x-to-5.4.0 in most of these links, in the section "Database", I find only the subsection "MySQL Database Structure", but I don't find the indications for DB migrations for PostgreSQL Database. Only here https://www.kamailio.org/wiki/install/upgrade/5.2.x-to-5.3.0 I found both the subsection "MySQL Database Structure" and "PostgreSQL Database Structure". What does this mean? - that I'm getting lost in a glass of water? - or that I don't have to make any changes to PostgreSQL? - or that the queries to be launched on PostgreSQL are the same as those for MySQL? I would discard this hypothesis - or that the wiki sections for PostgreSQL are missing? If this were the case, I could proceed with the update and then share the queries I launched to update the wiki. thanks a lot Giovanni Tommasini

3 5

Critical log when $mbu pseudovariable is accessed
by Alberto Izquierdo 11 Oct '21

11 Oct '21

Hi, I'm getting some CRITICAL logs when accessing the content of the $mbu pseudovariable in a "onsend_route" function. The log lines look like this: ``` CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:893 FIXME: null send_sock CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:1479 FIXME: null bind_address CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:493 null send socket CRITICAL core 35847:core/msg_translator.c:493 null send socket ``` I've checked the msg_translator.c source file and there are many of `LM_CRIT("FIXME: {message}\n");` calls, and I wonder if the "FIXME" tag relates to the location of the error: a Kamailio error or there is an issue in the user code. Could I have more information about the "FIXME" tag? Thanks in advance!

2 7

rtpengine module dependencies
by Juha Heinanen 10 Oct '21

10 Oct '21

rtpengine module README tells this about module dependencies: 3.1. Kamailio Modules The following modules must be loaded before this module: tm module - (optional) if you want to have rtpengine_manage() fully functional Looks like this is not correct. If Kamailio connects to rtpengine over websocket protocol, then there is dependency on lwsc module. I haven't checked if lwsc module works without tls module when wss is used. -- Juha

1 0

Kamailio Inbound proxy to Asterisk - ACL Filtering
by Mihai 09 Oct '21

09 Oct '21

Hello, I have an issue with filtering on the asterisk side, my requests are: UsersPhones(bria) -> Kamailio -> Asterisk -> Sip Trunk Out. The goal is to manage a new layer of protection ( IP filtering / Whitelisting ). When I try to compile a list of Whitelisted IP in sip.conf I get this error: NOTICE[205]: acl.c:748 ast_apply_acl: SIP contact ACL: Rejecting '145.72.23.45' due to a failure to pass ACL '(BASELINE)' WARNING[205]: chan_sip.c:17061 parse_register_contact: Domain '5.12.16.2:48669' disallowed by contact ACL (violating IP 145.72.23.45) WARNING[205]: chan_sip.c:17933 register_verify: Registration denied because of contact ACL The IP 145.72.23.45, is the proxy kamailio and if I added it to sip.conf it works, but so does every ip afterwards. I tried with contactpermit also with permit, the result is the same as long as I permit the proxy ip it works. Is there something that I can do on the asterisk side to activate this filtering Or there is something that I can do in Kamailio so it will forward the realip ? contactdeny=0.0.0.0/0.0.0.0 contactpermit=145.72.23.45/32 contactpermit=5.12.16.2/32 Thanks in advance,

1 0

Nagios NRPE Plugin for Kamailio 5.x
by Angelo Sipper 08 Oct '21

08 Oct '21

Hi, I would like to monitor kamailio via our nagios. Is there any working plugin available for nagios nrpe to monitor shared memory and some other config variables for kamailio 5.x that someone has used and could suggest? Kind regards, Angelo

2 1

update max_while_loops in config
by Juha Heinanen 08 Oct '21

08 Oct '21

It is possible to update value of max_while_loops dynamically in config file? For example, there could be a need to increase the max in some while loop, but after the loop is done, decrease it again for other loops. There exists an rpc command to do that, but I haven't found how to do it in config file. I tried @cfg_seti.core.max_while_loops = 200; but got syntax error. -- Juha

3 6

Suspend INVITE between ip4 and ip6 users
by Péter Barabás 08 Oct '21

08 Oct '21

Hi, I try to modify the kamailio configuration to work between ipv4 and ipv6 clients. Clients are mobile sofphones, so they are mostly in background without registration. 1. When an INVITE arrives from IPV4 address, transaction is suspended and a push notification is sent to callee. 2. Callee gets push notification and registers to kamailio. 3. INVITE transaction continues. My kamailio config is based on this article: https://kb.asipto.com/kamailio:kamailio-mixed-ipv4-ipv6 using rtpengine. If clients are in foreground it works fine. But, when a client is in background, kamailio knows nothing about the callee’s IP address (ipv4 or ipv6). The problematic case is the call from IPV4 to IPV6. In this case the FLB_V4V6 flag is missing at continue, since in IPV4V6 route it cannot be set and the rtpengine tries the IPV4->IPV4 case. From the other side IPV6->IPV6. It is understandable, since FLB_V4V6 flag is missing in this case. My question is: where and when should I set this flag? Thanks. Peter

1 0

Missing line in line.count transformation
by Daniel Donoghue 08 Oct '21

08 Oct '21

I think I’ve found an edge case with the string transformations for line.count: In pv_trans.c, the following code does not appear to consider the last line if there are multiple lines and the last one is not terminated with \n. case TR_LINE_COUNT: n=0; for(i=0; i<val->rs.len; i++) if(val->rs.s[i]=='\n') n++; if(n==0 && val->rs.len>0) n = 1; val->flags = PV_TYPE_INT|PV_VAL_INT|PV_VAL_STR; val->ri = n; val->rs.s = int2str(val->ri, &val->rs.len); break; This means that my count is always one off unless I force a blank line at the end of my string. Also, this code does not cater for cases where there is a different line terminator, such as CR or CRLF as is the case when the string comes from a Mac or windows system, respectively; it might be useful to have a transformation for doing that as I’ve had to hack it using s.replace with hexadecimal escape codes (\r and \n do not work) in a variable. With every blessing, — Daniel Donoghue

4 5

Parsing Error in Timer Module of kamailio
by Surabhi Garg 07 Oct '21

07 Oct '21

Hi Team, Greetings! I am trying to use Timer Module of Kamailio to keep calling particular route at particular time interval. Here is the snippet of my config: loadmodule "timer.so" modparam("timer", "declare_timer", "MYTIMER=SHUTDOWN,5000,slow,enable"); route[SHUTDOWN] { xlog("system shutdown: $sht(graceful_shutdown=>shutdown)\n"); } But after restarting the Kamailio, I keep on getting parsing errors in every 5 sec and my route "SHUTDOWN" never get called. Which means timer is getting called in every 5 sec but not able to call the route. Could you please help me with this why these parsing errors coming? am I missing some arguments here? Kamailio Version: 5.5.X. Errors: Sep 23 15:29:31 N1VL-PA-SIP01 /usr/local/kamailio-5.5/sbin/kamailio[9907]: ERROR: <core> [core/parser/msg_parser.c:748]: parse_msg(): ERROR: parse_msg: message=<GET /timer HTTP/0.9#012#012> Sep 23 15:29:31 N1VL-PA-SIP01 /usr/local/kamailio-5.5/sbin/kamailio[9907]: ERROR: timer [timer.c:204]: timer_handler(): parse msg failed Sep 23 15:29:36 N1VL-PA-SIP01 /usr/local/kamailio-5.5/sbin/kamailio[9907]: ERROR: <core> [core/parser/msg_parser.c:748]: parse_msg(): ERROR: parse_msg: message=<GET /timer HTTP/0.9#012#012> Sep 23 15:29:36 N1VL-PA-SIP01 /usr/local/kamailio-5.5/sbin/kamailio[9907]: ERROR: timer [timer.c:204]: timer_handler(): parse msg failed Also, in place of timer module I tried using rtimer module as well which is also giving me similar results as per my requirement. But Could you please help me with the pros and cons of both the modules timer and rtimer. Any thoughts on this would be appreciated Surabhi Garg Mobile - 91 783 888 7523

2 2

[kamailio/kamailio] SIP Preconditions Settings?
by Luis Joel Borrego Soto 07 Oct '21

07 Oct '21

Hi Team, According with the RFC3312 (https://datatracker.ietf.org/doc/html/rfc3312) The SDP parameters are the following We define the following media level SDP attributes: current-status = "a=curr:" precondition-type SP status-type SP direction-tag desired-status = "a=des:" precondition-type SP strength-tag SP status-type SP direction-tag confirm-status = "a=conf:" precondition-type SP status-type SP direction-tag **precondition-type = "qos" | token strength-tag = ("mandatory" | "optional" | "none" = | "failure" | "unknown")** status-type = ("e2e" | "local" | "remote") direction-tag = ("none" | "send" | "recv" | "sendrecv") Which one should be the equivalent in the Kamailio?, is the a specific module? Br, Joel Borrego (joel.borrego)

3 3

Need kamailio to use advertise public IP only in cases of SIP communication over internet.
by Voip support 06 Oct '21

06 Oct '21

Dear Community, I have a need of communicating with other servers using private IP with private IP in Via and SDP connection information but at same time sometime need connect to public IP SIP server and on outgoing INVITE's i need advertise public IP in Record-Route and SDP Connection information. Currently if i work on listen=udp:privateip:5060 i get working the LAN part but cannot work over internet (no audio) but when i add advertise i get SDP public IP also over local network. Is it possible to solve this case using one NIC adapter? I am using RTPProxy. Please help me out. I tried to add second network adapter with advertise and mhomed=1 something started to work but i think i would need work on the call direction and different rtpproxy_manage handling on outside INVITEs and incoming 200 OK responses. I am not sure if this is the right way to do it - maybe there is some easier method of achieving what i need using one interface ? Best regards, Tom

4 3

Understanding regex format for subst_hf
by Joel Serrano 06 Oct '21

06 Oct '21

Hey guys, I'm trying to do a replacement regex using subst_hf but I can't seem to get the format right and the docs are confusing. *subst_hf: *docs say it uses perl-like regex (as I understand it, that would mean PCRE). Docs also say in the 'subexp' param that it uses the same format as *subst.* *subst: *at the beginning it says it uses perl or sed format, but in the '/re/repl/flags' param it says that sed format. https://www.kamailio.org/docs/modules/devel/modules/textops.html#textops.f.… 4.20. subst_hf(hf, subexp, flags) Perl-like substitutions in the body of a header field. Meaning of the parameters is as follows: - *hf* - header field name. - *subexp* - substitution expression in the same format as of the 'subst' function parameter. - *flags* - control flags - it has to be one of: a - all headers matching the name; f - only first header matching the name; l - only the last header matching the name. 4.16. subst('/re/repl/flags') Replaces re with repl (sed or perl like). Meaning of the parameters is as follows: - *'/re/repl/flags'* - sed like regular expression. flags can be a combination of i (case insensitive), g (global) or s (match newline don't treat it as end of line). 're' - is regular expression 'repl' - is replacement string - may contain pseudo-variables 'flags' - substitution flags (i - ignore case, g - global) I'm trying to use this: subst_hf("X-myheader", "/^(?!string1 |example2-|text3 )(.*)/myprefix \1/", "a") In an online tester using PCRE, the format is correct, but Kamailio doesn't like it: Oct 4 09:29:16 cops-n1 cops[15957]: ERROR: <core> [core/re.c:285]: subst_parser(): bad regular expression ^(?!string1 |example2-|text3 )(.*) in /^(?!string1 |example2-|text3 )(.*)/myprefix \1/ If I remove the negative lookahead chars (?!) then kamailio doesn't complain but the regex doesn't do what I want. Does this mean that Kamailio won't support negative lookahead for subst regex replacements? What I'm trying to achieve is something along: "If the value of the header Myheader doesn't start with ABC, DEF or HIJ, then prefix it with XYZ" Any suggestions/ideas on how to achieve this? Thanks, Joel.

3 7

seas module
by Daniel Paape 05 Oct '21

05 Oct '21

I am having some difficulties with the seas module under Kam 5.5.2. If my application server is not connected then the config file script below never returns if (!as_relay_t ("myserver")) { Xlog ("LM_NOTICE", "myserver not connected\n"); } A diagnostic in the "seas.c" file does report that the app server is not connected, but the one above is never printed. Any help here would be appreciated. This email is intended only for the use of the party to which it is addressed and may contain information that is privileged, confidential, or protected by law. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this email or its contents is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer.

2 5

Kamailio/RTPengine as a proxy for FreePBX/Asterisk...
by Micah Quinn 03 Oct '21

03 Oct '21

Hello all, I'm new to Kamailio, so bear with me as I stumble through this. First, I'll describe what I'm trying to achieve at a high level and then perhaps somebody can advise me on whether Kamailio is a good fit for this solution or not. I'd like to be able to deploy a small appliance type server to our customer's sites that just runs Kamailio and a VPN connection back to our datacenter. At our datacenter, we run virtualized instances of Asterisk for each of our customers. The idea is that Kamailio would act as a transparent proxy through to the Asterisk instance under nominal conditions and as a basic SIP router in the case that the Asterisk instance is unavailable. This degraded functionality would then at least allow extension to extension calling even if the Internet or Asterisk instance is down. I'm currently using dispatcher with a single entry in preparation for a time when we might want to failover to another Asterisk instance. I'm forwarding all REGISTER and INVITE messages to the server chosen from ds_select_dst. Initially this all seems to work as I can register with a softphone and pjsip show endpoints shows my softphone connected. However, when I attempt to call any extension (my own or another) Asterisk responds to the INVITE message with a "401 Unauthorized" message and the typical "The person at extension XXXX is unavailable...". I know that more details might be necessary to troubleshoot this, but I didn't want to include everything in one post and risk cluttering it up with unnecessary information. If anyone can confirm that this is a reasonable way to approach the problem, I can then provide whatever relevant data is necessary to get deeper into it. (I've used sngrep, logging, asterisk cli, etc.) Thanks in advance for any help.

6 22

Kamailio 4.4 + rtpproxy behind the NAT (debian stretch 9.13)
by Youssef Boujraf 02 Oct '21

02 Oct '21

Dear, I am testing the kakamilio 4.4 with rtpproxy on Debian stretch 9.13 I have installed the packages - apt-get install kamailio and rtpproxy. - kamailio 4.4 - rtpproxy Basic version: 20040107 *My version of RTPPROXY doesn't have the -A parameter so that means without advertising there is no way to work behind NAT? That is what I have understood after spending one week on forums. Thanks if somebody could confirm otherwise, which way I need to go to figure out my issue?* # rtpproxy -v > Basic version: 20040107 > Extension 20050322: Support for multiple RTP streams and MOH > Extension 20060704: Support for extra parameter in the V command > Extension 20071116: Support for RTP re-packetization > Extension 20071218: Support for forking (copying) RTP stream > Extension 20080403: Support for RTP statistics querying > Extension 20081102: Support for setting codecs in the update/lookup command > Extension 20081224: Support for session timeout notifications > > # rtpproxy -help > rtpproxy: invalid option -- 'h' > usage: rtpproxy [-2fvFiPa] [-l addr1[/addr2]] [-6 addr1[/addr2]] [-s path] > [-t tos] [-r rdir [-S sdir]] [-T ttl] [-L nfiles] [-m port_min] > [-M port_max] [-u uname[:gname]] [-n timeout_socket] [-d > log_level[:log_facility]] > # > /etc/hosts 127.0.0.1 localhost > 127.0.1.1 sip.dom.xx sip > 192.168.1.19 sip.dom.xx sip > PUBLIC_IP sip.dom.xx sip > # The following lines are desirable for IPv6 capable hosts > ::1 localhost ip6-localhost ip6-loopback > ff02::1 ip6-allnodes > ff02::2 ip6-allrouters > */etc/kamailio/kamailio.cfg* kamailio.config > #!define PUBLIC_IP "xxx.xxx.xxx.xxx" > #!define PRIVATE_IP "192.168.1.19" > #!define PRIVATE_NET "192.168.1.0/24“ > listen=udp:PRIVATE_IP:5060 advertise PUBLIC_IP:5060 > > #!ifdef WITH_NAT > > # ----- rtpproxy params ----- > > modparam("rtpproxy", "rtpproxy_sock", "udp:PRIVATE_IP:7722") > > > # ----- nathelper params ----- > > modparam("nathelper", "natping_interval", 30) > > modparam("nathelper", "ping_nated_only", 1) > > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > *Etc/default/rtpproxy* PRIVATE_IP=192.168.1.19 > > PUBLIC_IP=xxx.xxx.xxx.xxx > > rtpport_min= 10000 > > rtpport_max=20000 > > syslog=DBUG:LOG_LOCAL5 > > #Defaults for rtpproxy > #The control socket. > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > #To listen on an UDP socket, uncomment this line: > #CONTROL_SOCK=udp:127.0.0.1:7722 > #Additional options that are passed to the daemon. > EXTRA_OPTS="-l $PRIVATE_IP/$PUBLIC_IP ="-s $PRIVATE_IP:7222 -m $ > rtpport_min -M $rtpport_max -d $syslog"` > Thanks if you could route me to the right way. Best regards Yboujraf Email: yboujraf(a)gmail.com

3 6

Fail call from Webrct to PSTN SIP trunk
by Abdul Rahman Reza 01 Oct '21

01 Oct '21

Hi, good morning there is something I want to ask, I'm really stuck looking for a solution, haven't found it yet, So I want to make Webrct able to call SIP and PSTN numbers via SIP trunk, but it keeps failing For testing as follows: webrct => SIP OK webrct => PSTN NOK webrct => webrct OK SIP => webrct OK SIP => PSTN OK SIP => SIP OK Thank you very much in advance this is error log from kamailio webrct => PSTN gagal dan error muncul seperti ini : Aug 16 11:25:16 esbdev01 /usr/sbin/kamailio[28300]: INFO: : START WEBSOCKET : INVITE from sip:500@domain.name (IP:127.0.0.1:43402) websocket id 12 Aug 16 11:25:16 esbdev01 /usr/sbin/kamailio[28300]: INFO: : MANAGE_BRANCH: New branch [0] to sip:+6282311XXXXXX@IP_SIP_TRUNK:5060 Aug 16 11:25:16 esbdev01 /usr/sbin/kamailio[28300]: INFO: : NATMANAGE branch_id:0 ruri: sip:+6282311XXXXXX@IP_SIP_TRUNK:5060, method:INVITE, status:, extra_id: z9hG4bK46635730, rtpengine_manage: replace-origin replace-session-connection via-branch=extra rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP Aug 16 11:25:16 esbdev01 rtpengine[16670]: INFO: [eq07esfe0a3mcjvpait6]: [control] Received command ‘offer’ from 127.0.0.1:38616 Aug 16 11:25:16 esbdev01 rtpengine[16670]: NOTICE: [eq07esfe0a3mcjvpait6]: [core] Creating new call Aug 16 11:25:16 esbdev01 rtpengine[16670]: WARNING: [eq07esfe0a3mcjvpait6]: [core] Ignoring updated remote endpoint as the local socket is IPv4 Aug 16 11:25:16 esbdev01 rtpengine[16670]: WARNING: [eq07esfe0a3mcjvpait6]: [core] Ignoring updated remote endpoint as the local socket is IPv4 Aug 16 11:25:16 esbdev01 rtpengine[16670]: INFO: [eq07esfe0a3mcjvpait6]: [control] Replying to ‘offer’ from 127.0.0.1:38616 (elapsed time 0.005891 sec) Aug 16 11:25:16 esbdev01 /usr/sbin/kamailio[28280]: INFO: : BRANCH FAILED: z9hG4bK4663573 + 0 Aug 16 11:25:16 esbdev01 rtpengine[16670]: INFO: [eq07esfe0a3mcjvpait6]: [control] Received command ‘delete’ from 127.0.0.1:44664 Aug 16 11:25:16 esbdev01 rtpengine[16670]: INFO: [eq07esfe0a3mcjvpait6]: [core] Scheduling deletion of call branch ‘’ (via-branch ‘z9hG4bK46635730’) in 30 seconds Aug 16 11:25:16 esbdev01 rtpengine[16670]: INFO: [eq07esfe0a3mcjvpait6]: [control] Replying to ‘delete’ from 127.0.0.1:44664 (elapsed time 0.000210 sec) Aug 16 11:25:16 esbdev01 /usr/sbin/kamailio[28280]: INFO: : Failure: and this is log from webrct browser using jssip LoggerFactory.js:100 Mon Aug 16 2021 10:42:54 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | SessionDescriptionHandlerOptions: {"peerConnectionOptions":{"rtcConfiguration":{"iceServers":[{"urls":"turn:domain_name:443?transport=tcp","username":"websip","credential":"websip"},{"urls":"turns:domain_name:80?transport=tcp","username":"websip","credential":"websip"}]},"iceCheckingTimeout":5000}} LoggerFactory.js:100 Mon Aug 16 2021 10:42:54 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | initPeerConnection LoggerFactory.js:100 Mon Aug 16 2021 10:42:54 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | New peer connection created LoggerFactory.js:100 Mon Aug 16 2021 10:42:54 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | acquiring local media LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | acquired local media streams LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | createOffer LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | Setting local sdp. LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | sdp is v=0 o=- 5499752371154858348 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 a=extmap-allow-mixed a=msid-semantic: WMS 2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w m=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:3068753223 cname:vdFflh5e6U+kdnYw a=ssrc:3068753223 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=ssrc:3068753223 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:3068753223 label:f8941c64-0252-4ab7-b1c5-3731f47c9fd7 m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:1 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:13 urn:3gpp:video-orientation a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 goog-remb a=rtcp-fb:96 transport-cc a=rtcp-fb:96 ccm fir a=rtcp-fb:96 nack a=rtcp-fb:96 nack pli a=rtpmap:97 rtx/90000 a=fmtp:97 apt=96 a=rtpmap:98 VP9/90000 a=rtcp-fb:98 goog-remb a=rtcp-fb:98 transport-cc a=rtcp-fb:98 ccm fir a=rtcp-fb:98 nack a=rtcp-fb:98 nack pli a=fmtp:98 profile-id=0 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=98 a=rtpmap:100 VP9/90000 a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=fmtp:100 profile-id=2 a=rtpmap:101 rtx/90000 a=fmtp:101 apt=100 a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc a=rtcp-fb:102 ccm fir a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:121 rtx/90000 a=fmtp:121 apt=102 a=rtpmap:127 H264/90000 a=rtcp-fb:127 goog-remb a=rtcp-fb:127 transport-cc a=rtcp-fb:127 ccm fir a=rtcp-fb:127 nack a=rtcp-fb:127 nack pli a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f a=rtpmap:120 rtx/90000 a=fmtp:120 apt=127 a=rtpmap:125 H264/90000 a=rtcp-fb:125 goog-remb a=rtcp-fb:125 transport-cc a=rtcp-fb:125 ccm fir a=rtcp-fb:125 nack a=rtcp-fb:125 nack pli a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f a=rtpmap:107 rtx/90000 a=fmtp:107 apt=125 a=rtpmap:108 H264/90000 a=rtcp-fb:108 goog-remb a=rtcp-fb:108 transport-cc a=rtcp-fb:108 ccm fir a=rtcp-fb:108 nack a=rtcp-fb:108 nack pli a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 a=rtpmap:35 AV1X/90000 a=rtcp-fb:35 goog-remb a=rtcp-fb:35 transport-cc a=rtcp-fb:35 ccm fir a=rtcp-fb:35 nack a=rtcp-fb:35 nack pli a=rtpmap:36 rtx/90000 a=fmtp:36 apt=35 a=rtpmap:124 H264/90000 a=rtcp-fb:124 goog-remb a=rtcp-fb:124 transport-cc a=rtcp-fb:124 ccm fir a=rtcp-fb:124 nack a=rtcp-fb:124 nack pli a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:119 rtx/90000 a=fmtp:119 apt=124 a=rtpmap:123 H264/90000 a=rtcp-fb:123 goog-remb a=rtcp-fb:123 transport-cc a=rtcp-fb:123 ccm fir a=rtcp-fb:123 nack a=rtcp-fb:123 nack pli a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:118 rtx/90000 a=fmtp:118 apt=123 a=rtpmap:114 red/90000 a=rtpmap:115 rtx/90000 a=fmtp:115 apt=114 a=rtpmap:116 ulpfec/90000 a=ssrc-group:FID 1593038176 4012956172 a=ssrc:1593038176 cname:vdFflh5e6U+kdnYw a=ssrc:1593038176 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:1593038176 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:1593038176 label:1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 cname:vdFflh5e6U+kdnYw a=ssrc:4012956172 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:4012956172 label:1e978692-178f-46cb-8c18-8f130343f637 LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | waitForIceGatheringComplete LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE is not complete. Returning promise LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: gathering LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 63469 typ host generation 0 ufrag c/YU network-id 2 network-cost 50 LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:211156821 1 udp 2122194687 192.XXX.X.X 50637 typ host generation 0 ufrag c/YU network-id 1 network-cost 10 LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 55429 typ host generation 0 ufrag c/YU network-id 2 network-cost 50 LoggerFactory.js:100 Mon Aug 16 2021 10:42:55 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:211156821 1 udp 2122194687 192.XXX.X.X 50147 typ host generation 0 ufrag c/YU network-id 1 network-cost 10 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 ufrag c/YU network-id 2 network-cost 50 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 ufrag c/YU network-id 1 network-cost 10 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 ufrag c/YU network-id 2 network-cost 50 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate received: candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 ufrag c/YU network-id 1 network-cost 10 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | RTCIceGatheringState changed: complete LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | sending WebSocket message: INVITE sip:+6282311XXXXXX@domain_name SIP/2.0 Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK4035596 Max-Forwards: 70 To: <sip:+6282311XXXXXX@domain_name> From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8516 INVITE Contact: <sip:tfqbf6kb@asv5il0mh6sc.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.13.6 Content-Type: application/sdp Content-Length: 6891 v=0 o=- 5499752371154858348 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 a=extmap-allow-mixed a=msid-semantic: WMS 2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w m=audio 63469 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 10.XX.XX.XX a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 63469 typ host generation 0 network-id 2 network-cost 50 a=candidate:211156821 1 udp 2122194687 192.XXX.X.X 50637 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:3068753223 cname:vdFflh5e6U+kdnYw a=ssrc:3068753223 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=ssrc:3068753223 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:3068753223 label:f8941c64-0252-4ab7-b1c5-3731f47c9fd7 m=video 55429 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 c=IN IP4 10.XX.XX.XX a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 55429 typ host generation 0 network-id 2 network-cost 50 a=candidate:211156821 1 udp 2122194687 192.XXX.X.X 50147 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:1 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:13 urn:3gpp:video-orientation a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 goog-remb a=rtcp-fb:96 transport-cc a=rtcp-fb:96 ccm fir a=rtcp-fb:96 nack a=rtcp-fb:96 nack pli a=rtpmap:97 rtx/90000 a=fmtp:97 apt=96 a=rtpmap:98 VP9/90000 a=rtcp-fb:98 goog-remb a=rtcp-fb:98 transport-cc a=rtcp-fb:98 ccm fir a=rtcp-fb:98 nack a=rtcp-fb:98 nack pli a=fmtp:98 profile-id=0 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=98 a=rtpmap:100 VP9/90000 a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=fmtp:100 profile-id=2 a=rtpmap:101 rtx/90000 a=fmtp:101 apt=100 a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc a=rtcp-fb:102 ccm fir a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:121 rtx/90000 a=fmtp:121 apt=102 a=rtpmap:127 H264/90000 a=rtcp-fb:127 goog-remb a=rtcp-fb:127 transport-cc a=rtcp-fb:127 ccm fir a=rtcp-fb:127 nack a=rtcp-fb:127 nack pli a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f a=rtpmap:120 rtx/90000 a=fmtp:120 apt=127 a=rtpmap:125 H264/90000 a=rtcp-fb:125 goog-remb a=rtcp-fb:125 transport-cc a=rtcp-fb:125 ccm fir a=rtcp-fb:125 nack a=rtcp-fb:125 nack pli a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f a=rtpmap:107 rtx/90000 a=fmtp:107 apt=125 a=rtpmap:108 H264/90000 a=rtcp-fb:108 goog-remb a=rtcp-fb:108 transport-cc a=rtcp-fb:108 ccm fir a=rtcp-fb:108 nack a=rtcp-fb:108 nack pli a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 a=rtpmap:35 AV1X/90000 a=rtcp-fb:35 goog-remb a=rtcp-fb:35 transport-cc a=rtcp-fb:35 ccm fir a=rtcp-fb:35 nack a=rtcp-fb:35 nack pli a=rtpmap:36 rtx/90000 a=fmtp:36 apt=35 a=rtpmap:124 H264/90000 a=rtcp-fb:124 goog-remb a=rtcp-fb:124 transport-cc a=rtcp-fb:124 ccm fir a=rtcp-fb:124 nack a=rtcp-fb:124 nack pli a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:119 rtx/90000 a=fmtp:119 apt=124 a=rtpmap:123 H264/90000 a=rtcp-fb:123 goog-remb a=rtcp-fb:123 transport-cc a=rtcp-fb:123 ccm fir a=rtcp-fb:123 nack a=rtcp-fb:123 nack pli a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:118 rtx/90000 a=fmtp:118 apt=123 a=rtpmap:114 red/90000 a=rtpmap:115 rtx/90000 a=fmtp:115 apt=114 a=rtpmap:116 ulpfec/90000 a=ssrc-group:FID 1593038176 4012956172 a=ssrc:1593038176 cname:vdFflh5e6U+kdnYw a=ssrc:1593038176 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:1593038176 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:1593038176 label:1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 cname:vdFflh5e6U+kdnYw a=ssrc:4012956172 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:4012956172 label:1e978692-178f-46cb-8c18-8f130343f637 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE candidate gathering complete LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | received WebSocket text message: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK4035596;rport=43402;received=127.0.0.1 To: <sip:+6282311XXXXXX@domain_name >;tag=a2d3384ccaf3d0890b1d994026562a34.429c0000 From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8516 INVITE Proxy-Authenticate: Digest realm="domain_name", nonce="YRnf9GEZ3shCY5HMSWfzE6ctJ5ig1Kd7" Server: kamailio (5.5.0 (x86_64/linux)) Content-Length: 0 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | sending WebSocket message: ACK sip:+6282311XXXXXX@domain_name SIP/2.0 Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK7163476 Max-Forwards: 70 To: <sip:+6282311XXXXXX@domain_name >;tag=a2d3384ccaf3d0890b1d994026562a34.429c0000 From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8516 ACK Supported: outbound User-Agent: SIP.js/0.13.6 Content-Length: 0 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | sending WebSocket message: INVITE sip:+6282311XXXXXX@domain_name SIP/2.0 Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK6711062 Max-Forwards: 70 To: <sip:+6282311XXXXXX@domain_name> From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8517 INVITE Proxy-Authorization: Digest algorithm=MD5, username="500", realm="domain_name", nonce="YRnf9GEZ3shCY5HMSWfzE6ctJ5ig1Kd7", uri="sip:+6282311XXXXXX@domain_name", response="da7c3cf9c21d8aa85f72c5e723f8d42c" Contact: <sip:tfqbf6kb@asv5il0mh6sc.invalid;transport=ws;ob> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: SIP.js/0.13.6 Content-Type: application/sdp Content-Length: 6891 v=0 o=- 5499752371154858348 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 1 a=extmap-allow-mixed a=msid-semantic: WMS 2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w m=audio 63469 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126 c=IN IP4 10.XX.XX.XX a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 63469 typ host generation 0 network-id 2 network-cost 50 a=candidate:211156821 1 udp 2122194687 192.XXX.X.X 50637 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:112 telephone-event/32000 a=rtpmap:113 telephone-event/16000 a=rtpmap:126 telephone-event/8000 a=ssrc:3068753223 cname:vdFflh5e6U+kdnYw a=ssrc:3068753223 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w f8941c64-0252-4ab7-b1c5-3731f47c9fd7 a=ssrc:3068753223 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:3068753223 label:f8941c64-0252-4ab7-b1c5-3731f47c9fd7 m=video 55429 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 121 127 120 125 107 108 109 35 36 124 119 123 118 114 115 116 c=IN IP4 10.XX.XX.XX a=rtcp:9 IN IP4 0.0.0.0 a=candidate:3250795810 1 udp 2122260223 10.XX.XX.XX 55429 typ host generation 0 network-id 2 network-cost 50 a=candidate:211156821 1 udp 2122194687 192.XXX.X.X 50147 typ host generation 0 network-id 1 network-cost 10 a=candidate:2403387858 1 tcp 1518280447 10.XX.XX.XX 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=candidate:1108738981 1 tcp 1518214911 192.XXX.X.X 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=ice-ufrag:c/YU a=ice-pwd:7VB8/SEvBOYMyPpKgYKzOpPK a=ice-options:trickle a=fingerprint:sha-256 92:73:F2:E2:70:7C:43:BD:59:3A:6C:F3:7D:B9:9F:75:19:81:4C:40:0E:DA:63:D9:FF:90:60:41:49:E6:B0:9C a=setup:actpass a=mid:1 a=extmap:14 urn:ietf:params:rtp-hdrext:toffset a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:13 urn:3gpp:video-orientation a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:12 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay a=extmap:11 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:6 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=rtcp-mux a=rtcp-rsize a=rtpmap:96 VP8/90000 a=rtcp-fb:96 goog-remb a=rtcp-fb:96 transport-cc a=rtcp-fb:96 ccm fir a=rtcp-fb:96 nack a=rtcp-fb:96 nack pli a=rtpmap:97 rtx/90000 a=fmtp:97 apt=96 a=rtpmap:98 VP9/90000 a=rtcp-fb:98 goog-remb a=rtcp-fb:98 transport-cc a=rtcp-fb:98 ccm fir a=rtcp-fb:98 nack a=rtcp-fb:98 nack pli a=fmtp:98 profile-id=0 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=98 a=rtpmap:100 VP9/90000 a=rtcp-fb:100 goog-remb a=rtcp-fb:100 transport-cc a=rtcp-fb:100 ccm fir a=rtcp-fb:100 nack a=rtcp-fb:100 nack pli a=fmtp:100 profile-id=2 a=rtpmap:101 rtx/90000 a=fmtp:101 apt=100 a=rtpmap:102 H264/90000 a=rtcp-fb:102 goog-remb a=rtcp-fb:102 transport-cc a=rtcp-fb:102 ccm fir a=rtcp-fb:102 nack a=rtcp-fb:102 nack pli a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f a=rtpmap:121 rtx/90000 a=fmtp:121 apt=102 a=rtpmap:127 H264/90000 a=rtcp-fb:127 goog-remb a=rtcp-fb:127 transport-cc a=rtcp-fb:127 ccm fir a=rtcp-fb:127 nack a=rtcp-fb:127 nack pli a=fmtp:127 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42001f a=rtpmap:120 rtx/90000 a=fmtp:120 apt=127 a=rtpmap:125 H264/90000 a=rtcp-fb:125 goog-remb a=rtcp-fb:125 transport-cc a=rtcp-fb:125 ccm fir a=rtcp-fb:125 nack a=rtcp-fb:125 nack pli a=fmtp:125 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42e01f a=rtpmap:107 rtx/90000 a=fmtp:107 apt=125 a=rtpmap:108 H264/90000 a=rtcp-fb:108 goog-remb a=rtcp-fb:108 transport-cc a=rtcp-fb:108 ccm fir a=rtcp-fb:108 nack a=rtcp-fb:108 nack pli a=fmtp:108 level-asymmetry-allowed=1;packetization-mode=0;profile-level-id=42e01f a=rtpmap:109 rtx/90000 a=fmtp:109 apt=108 a=rtpmap:35 AV1X/90000 a=rtcp-fb:35 goog-remb a=rtcp-fb:35 transport-cc a=rtcp-fb:35 ccm fir a=rtcp-fb:35 nack a=rtcp-fb:35 nack pli a=rtpmap:36 rtx/90000 a=fmtp:36 apt=35 a=rtpmap:124 H264/90000 a=rtcp-fb:124 goog-remb a=rtcp-fb:124 transport-cc a=rtcp-fb:124 ccm fir a=rtcp-fb:124 nack a=rtcp-fb:124 nack pli a=fmtp:124 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032 a=rtpmap:119 rtx/90000 a=fmtp:119 apt=124 a=rtpmap:123 H264/90000 a=rtcp-fb:123 goog-remb a=rtcp-fb:123 transport-cc a=rtcp-fb:123 ccm fir a=rtcp-fb:123 nack a=rtcp-fb:123 nack pli a=fmtp:123 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032 a=rtpmap:118 rtx/90000 a=fmtp:118 apt=123 a=rtpmap:114 red/90000 a=rtpmap:115 rtx/90000 a=fmtp:115 apt=114 a=rtpmap:116 ulpfec/90000 a=ssrc-group:FID 1593038176 4012956172 a=ssrc:1593038176 cname:vdFflh5e6U+kdnYw a=ssrc:1593038176 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:1593038176 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:1593038176 label:1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 cname:vdFflh5e6U+kdnYw a=ssrc:4012956172 msid:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w 1e978692-178f-46cb-8c18-8f130343f637 a=ssrc:4012956172 mslabel:2yMfiToO0B0g0rBtLDd7vEaYmQ9PrZ0QUC9w a=ssrc:4012956172 label:1e978692-178f-46cb-8c18-8f130343f637 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | received WebSocket text message: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK6711062;rport=43402;received=127.0.0.1 To: <sip:+6282311XXXXXX@domain_name> From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8517 INVITE Server: kamailio (5.5.0 (x86_64/linux)) Content-Length: 0 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | ICE Connection State changed to iceConnectionChecking LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | received WebSocket text message: SIP/2.0 500 Server Internal Error Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK6711062;received=127.0.0.1;rport=43402 Call-ID: eq07e6in98hmjs1uulas From: "webrtc"<sip:500@domain_name>;tag=t9cuh03rp3 To: <sip:+6282311XXXXXX@domain_name>;tag=23r36zzp CSeq: 8517 INVITE Warning: 399 03077.08860.B.005.422.228.0.15.01039.00000000.00000 "SDP body length beyond limit" Content-Length: 0 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.transport | sending WebSocket message: ACK sip:+6282311XXXXXX@domain_name SIP/2.0 Via: SIP/2.0/WSS asv5il0mh6sc.invalid;branch=z9hG4bK2860365 Max-Forwards: 70 To: <sip:+6282311XXXXXX@domain_name>;tag=23r36zzp From: "webrtc" <sip:500@domain_name>;tag=t9cuh03rp3 Call-ID: eq07e6in98hmjs1uulas CSeq: 8517 ACK Supported: outbound User-Agent: SIP.js/0.13.6 Content-Length: 0 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.inviteclientcontext | closing INVITE session eq07e6in98hmjs1uulast9cuh03rp3 LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | closing PeerConnection LoggerFactory.js:100 Mon Aug 16 2021 10:42:56 GMT+0700 (Western Indonesia Time) | sip.invitecontext.sessionDescriptionHandler | resetIceGatheringComplete and this is config of kamailio.cfg #!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD ##!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!define WITH_PSTN #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!IP-SERVER!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![IPv6-SERVER]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!DOMAIN-NAME!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "IP-SIP-TRUNK-SERVER" pstn.gw_port = "5060" #!endif # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ##!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@DOMAIN-NAME") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if ($proto =~ "ws") { xlog("L_INFO", "START WEBSOCKET : $rm from $fu (IP:$si:$sp) websocket id $ws_conid\n"); } if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($sht(ipban=>$si) != $null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK")) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri != myself && uri != myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else if (is_first_hop()) { set_contact_alias(); } setflag(FLT_NATS); } return; } # NAT handling route[NATMANAGE] { if (is_request()) { if (has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif )) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # SIP --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (status =~ "[12][0-9][0-9]") { route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; }

2 2

ACK not accepted
by Angelo Sipper 01 Oct '21

01 Oct '21

Hi, I am having a problem with some inbound calls that all have the below behavior. My setup has: Carrier(UDP) -> (UDP)Kamailio1(UDP) -> (UDP)Kamailio2(TLS) -> (TLS)UAC Kamailio1 has Topos Module. UAC is behind NAT with LAN IP 192.168.1.105 and Public 200.200.200.30 Carrier IP 50.50.50.19, Kamailio1 IP 100.100.100.243, Kamailio2 IP 100.100.100.210 The Carrier sends in a new call and the call is answered from the UAC with 200 OK SDP. Carrier sends ACK and it looks like it reaches the UAC but, UAC still resends several times 200 OK and carrier sends server ACK that arrives or not ??? to UAC until finally UAC sends a BYE. On the SIP flow you can see that Kamailio2 sends the ACK to 192.168.1.105 and not to 200.200.200.30. Is this ok?? Another thing to mention is that Kamailio2 that has the UAC connected via TLS, can be used fine from the UAC to make an outbound call following the reverse path, but on the inbound we have the below. Any help will be much appreciated, ACK from Kamailio2 100.100.100.210 to UAC Session Initiation Protocol (ACK) Request-Line: ACK sip:btpsh-f3-6153392a-5fa-5@192.168.1.105:5061;transport=TLS SIP/2.0 Message Header Via: SIP/2.0/TLS 100.100.100.210:5061 ;branch=z9hG4bKfc5f.67a77810bcdafe811ea3672a027450d8.0 Via: SIP/2.0/UDP 100.100.100.243;rport=5060;branch=z9hG4bKfc5f.e2b515e9c8c33d2bd05682f26df24e39.0 Call-ID: isbcaqhye0rozjgeghorzav1vtytoqztkheq(a)B.5.108.ims.forthnet.gr [Generated Call-ID: isbcaqhye0rozjgeghorzav1vtytoqztkheq(a)B.5.108.ims.incarrier.com] From: <sip:987654321@ims.carrier.com ;user=phone>;tag=ZZ58CDf_CZ.Yj2U5 To: <sip:23456789@ims.carrier.com ;user=phone>;tag=0120779a-edf4-464a-9c43-2ddf54459083 CSeq: 1 ACK Max-Forwards: 64 Request-Disposition: no-fork Content-Length: 0 Contact: <sip:btpsh-f3-6153392a-5fa-5@100.100.100.243> |Time | 50.50.50.19 | 100.100.100.210 | 192.168.1.105 | | | | 100.100.100.243 | | 200.200.200.30 | |0.000000 | INVITE SDP (g711A te | | | |SIP INVITE From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> Call-ID:isbcaqhye0rozjgeghorzav1vtytoqztkheq@B.5.108.ims.incarrier.gr CSeq:1 | |(5060) ------------------> (5060) | | | | |0.001737 | 100 trying -- your c | | | |SIP Status 100 trying -- your call is important to us | |(5060) <------------------ (5060) | | | | |0.074622 | | INVITE SDP (g711A te | | |SIP INVITE From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |0.074797 | | INVITE SDP (g711A te | | |SIP INVITE From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |0.081110 | | 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | | |(5060) <------------------ (5060) | | | |0.082070 | | 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | | |(5060) <------------------ (5060) | | | |0.082833 | | | INVITE SDP (g711A te | |SIP INVITE From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) ------------------> (37275) | | |0.163871 | | | 100 Trying| | |SIP Status 100 Trying | | | |(5061) <------------------ (37275) | | |0.447621 | | | 180 Ringing | |SIP Status 180 Ringing | | | |(5061) <------------------ (37275) | | |0.448004 | | 180 Ringing | | |SIP Status 180 Ringing | | |(5060) <------------------ (5060) | | | |0.448765 | | 180 Ringing | | |SIP Status 180 Ringing | | |(5060) <------------------ (5060) | | | |0.465033 | 180 Ringing | | | |SIP Status 180 Ringing | |(5060) <------------------ (5060) | | | | |2.529599 | | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |2.733556 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |2.734596 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |2.864311 | 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |3.024789 | | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |3.025005 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |3.025854 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |3.034979 | 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |3.106784 | ACK | | | | |SIP Request INVITE ACK 200 CSeq:1 | |(5060) ------------------> (5060) | | | | |3.126287 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |3.126542 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |3.145821 | | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |4.025449 | | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |4.025743 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |4.026548 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |4.033198 | 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |4.094832 | ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |4.102992 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |4.103316 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |4.106301 | | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |6.025167 | | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |6.025543 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |6.026377 | | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |6.034778 | 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |6.096896 | ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |6.104675 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |6.104910 | | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |6.108907 | | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |10.024538| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |10.024866| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |10.025750| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |10.033461| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |10.095387| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |10.102265| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |10.102449| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |10.105760| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |14.025310| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |14.025585| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |14.026528| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |14.037027| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |14.099479| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |14.111540| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |14.111708| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |14.116089| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |18.025311| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |18.025659| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |18.026572| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |18.034736| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |18.096101| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |18.106560| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |18.106687| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |18.111007| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |22.024673| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |22.024953| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |22.026120| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |22.033514| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |22.095104| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |22.105921| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |22.106043| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |22.110298| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |26.031501| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |26.031903| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |26.033136| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |26.040343| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |26.101967| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |26.111938| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |26.112432| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |26.118306| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |30.035446| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |30.035693| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |30.036501| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |30.044170| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |30.106395| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |30.116960| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |30.117412| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |30.129913| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |34.026492| | | 200 OK SDP (g711A) | |SIP Status 200 OK | | | |(5061) <------------------ (37275) | | |34.026807| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |34.027775| | 200 OK SDP (g711A) | | |SIP Status 200 OK | | |(5060) <------------------ (5060) | | | |34.035453| 200 OK SDP (g711A) | | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | | | |34.097385| ACK | | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | |(5060) ------------------> (5060) | | | | |34.106319| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |34.106681| | ACK | | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | |(5060) ------------------> (5060) | | | |34.110714| | | ACK | | |SIP ACK From: < sip:987654321@ims.carrier.com;user=phone> To:<sip:123456789@ims.carrier.com;user=phone> CSeq:1 | | | |(5061) --------------------------------------> (5061) | |34.526688| | | BYE | | |SIP Request BYE CSeq:15322 | | | |(5061) <------------------ (37275) | | |34.528722| | BYE | | | |SIP Request BYE CSeq:15322 | | |(5060) <------------------ (5060) | | | |34.529704| | BYE | | | |SIP Request BYE CSeq:15322 | | |(5060) <------------------ (5060) | | | |34.668418| BYE | | | | |SIP Request BYE CSeq:15322 | |(5060) <------------------ (5060) | | | | |34.788963| 200 OK | | | | |SIP Status 200 OK | |(5060) ------------------> (5060) | | | | |34.796051| | 200 OK | | | |SIP Status 200 OK | | |(5060) ------------------> (5060) | | | |34.796205| | 200 OK | | | |SIP Status 200 OK | | |(5060) ------------------> (5060) | | | |34.796604| | | 200 OK | | |SIP Status 200 OK | | | |(5061) ------------------> (37275) | | Kind regards, Angelo

1 3

Querying presence state from config file
by Igor Olhovskiy 01 Oct '21

01 Oct '21

Hello! Is there any method of querying presence (dialog) state for particular endpoint from config file other than that direct query to presence tables in database? Idea is to reflect server-side DND state as "confirmed" (talking) , so I need to skip certain PUBLISH messages not to reset this state. -- Best regards, Igor

2 2

kamailio 4.4 + RTPPROXY - virtualbox - setup - callee or caller sends INVITE during the conversation and at REGISTER EXPIRES stop the connection
by Youssef Boujraf 01 Oct '21

01 Oct '21

Dear Thanks for your quick reply and sorry for spamming your mailbox. I am very confused about the setup of kamailio.cfg and rtpproxy. I have an existing kamailio 4.4 - rtpproxy running and I get the config files from both and setup the new vm. Just private connection works but the idea is to have connection from outside too. I am sure there is a mismatch with public ip, natting and/or setup of the vm hosts file. - architecture: *- 1x virtualbox* + 1x vm with bridge mode for network (get private ip address from the router). I suppose this is the PRIVATE_IP *+ 1x desktop (windows)* hosting a sipphone here Microsip ( https://www.microsip.org/) *- 1x modem/router* from the telecom provider - bbox-3V+ from PROXIMUS + forced in DMZ host the Kamailio server PRIVATE_IP [image: 211001 - kamailio architecture and settings.png] - the hosts file from Kamailio + rtpproxy is ( Maybe something is wrong or missing) /etc/hosts 127.0.0.1 localhost > 127.0.1.1 sipsecure.by-systems.be sipsecure > 192.168.1.19 sipsecure.by-systems.be sipsecure > xxx.XXX.xxx.XXX sipsecure.by-systems.be sipsecure > # The following lines are desirable for IPv6 capable hosts > ::1 localhost ip6-localhost ip6-loopback > ff02::1 ip6-allnodes > ff02::2 ip6-allrouters > */etc/kamailio/kamailio.cfg* kamailio.config > #!define PUBLIC_IP "xxx.xxx.xxx.xxx" > #!define PRIVATE_IP "192.168.1.19" > #!define PRIVATE_NET "192.168.1.0/24“ > > #!ifdef WITH_NAT > > # ----- rtpproxy params ----- > > modparam("rtpproxy", "rtpproxy_sock", "udp:PRIVATE_IP:7722") > > > # ----- nathelper params ----- > > modparam("nathelper", "natping_interval", 30) > > modparam("nathelper", "ping_nated_only", 1) > > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > *Etc/default/rtpproxy* PRIVATE_IP=192.168.1.19 > > PUBLIC_IP=xxx.xxx.xxx.xxx > > rtpport_min= 10000 > > rtpport_max=20000 > > syslog=DBUG:LOG_LOCAL5 > > #Defaults for rtpproxy > #The control socket. > #CONTROL_SOCK="unix:/var/run/rtpproxy/rtpproxy.sock" > #To listen on an UDP socket, uncomment this line: > #CONTROL_SOCK=udp:127.0.0.1:7722 > #Additional options that are passed to the daemon. > EXTRA_OPTS="-l $PRIVATE_IP/$PUBLIC_IP ="-s $PRIVATE_IP:7222 -m $ > rtpport_min -M $rtpport_max -d $syslog"` > - *Misconfiguring the VM* That means do I need to use setup 1 or 2 or another 1. 2x NIC interfaces (one with private ip, second from dmz ip + NAT1:1 public ip) 2. 1x NIC interface (one private ip and natted 1:1 with public address) - *test* 1. *use case 1 : *connect two (2x) sipphone from private network (included kamailio) = OK 2. *use case 2 :* connect one (1x) codec from private ip and one (1x) codec from public network (like mobile) = = issue - disconnect after 30 s (register expires) start a conversation with audio but the caller or the callee continues to send and INVITE message. 3. *use case 3 :* connect two (2x) sipphone from public network = issue - disconnect after 30 s (register expires) start a conversation with audio but the caller or the callee continue to send an INVITE message. Best Regards, Youssef BOUJRAF

1 0

RTPEngine + Kamailio state question
by Alex Balashov 30 Sep '21

30 Sep '21

Hi, Sorry if I missed any developments on this, but: Kamailio does keep a runtime association of Call-ID+Ftag => RTPEngine instance, which is significant if an RTPEngine set with multiple members is used, or if a non-default set is used for a given call. When Kamailio is restarted, this runtime association goes away, so any subsequent offer/answer cycles or any other commands will (maybe) go to the wrong RTPEngine and/or the wrong set. Last time I checked a few years ago, there was no way to persist this association across restarts. Has this changed? Thanks! — Alex -- 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/

3 6

Sipwise SEMS
by David Villasmil 30 Sep '21

30 Sep '21

Hello guys, I know SEMS can provide conference, voicemail, and other services. In theory it’s also a B2BUA. Could I use a python script to provide a simple routing service? I.e.: receive an invite and send it somewhere else based on some routing logic? Thanks all David -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

10 22

using LDAP as database ?
by Volker Denneberg 28 Sep '21

28 Sep '21

Hi, I am searching for a setup where phones register at a central kamailio and kamailio shares these contact infos with multiple asterisks. So they all need to share a database. For redundancy reasons, the database should work across 2 datacenters. As far as 1 understood there are some options * use mysql - good chances to find example code, Siremis API, seems to be the standard solution * use redis - more performant than mysql * use ldap - good chances to integrate with existing IT management but most difficult to configure Maybe I missed a better approach? Kamailio's ldap driver (http://www.kamailio.net/docs/modules/5.3.x/modules/db2_ldap.html) seems to be from 2008 - is it still recommended ? Kind regards, Volker

2 1

siptrace & topos modules on 5.3
by Angelo Sipper 24 Sep '21

24 Sep '21

Hi, We are using kamailio 5.3 latest version with the topos module. We have added Homer 7 latest version to consolidate traces from all kamailios and we have enabled siptrace module on kamailio to send data to homer. What we have seen up today is that traces on homer have sip headers before topos changes. So if I look at the same call on sngrep on the kamailio machine, I see the correct topos changed headers. If I check the same call on Homer I see headers without topos changes. Also we get duplicate headers on ACKs on homer. I was wondering what we have done wrong in the config and we have these results on homer. Any help will be much appreciated! Our config has: ===========- #!substdef "!MY_HOMER_CAPTURE!sip:MY_HOMER_IP_ADDR:MY_HOMER_PORT!g" #!substdef "!MY_LOCAL_SENDSOCK!sip:MY_KAMAILIO_IP:MY_LOCAL_SEND_PORT!g" loadmodule "topos.so" loadmodule "siptrace.so" modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK") modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_mode", 0) modparam("siptrace", "hep_capture_id", NODE_ID) modparam("topos", "db_url", DBURL) modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "branch_expire", 130) # 3 Min modparam("topos", "dialog_expire", 10800) # 3 Hours modparam("topos", "clean_interval", 60) modparam("topos", "event_mode", 3) request_route { sip_trace_mode("t"); ........ } onsend_route { if (is_method("ACK")) { sip_trace(); } } ============== Thank you! Kind regards, Angelo

3 5

Info: APT Debian/Ubuntu Packages Archive
by Daniel-Constantin Mierla 24 Sep '21

24 Sep '21

Hello, a new server is available to offer APT repositories for each released version of Kamailio: * https://deb-archive.kamailio.org/ It aims to offer a way to downgrade to an older version, if someone wants it for various reasons. At this moment it offers dedicated APT repositories for Kamailio 5.3.8, 5.4.6, 5.4.7, 5.5.1 and 5.5.2. More details were posted in a news article at: * https://www.kamailio.org/w/2021/09/apt-debian-packages-archive/ 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/

1 0

PRACK with UDP to TCP translation send over wrong protocol
by Angelo Sipper 24 Sep '21

24 Sep '21

Hello, We have the following setup and I am having trouble passing the PRACK with correct protocol to the provider. UAC (OverTLS) ->(UDP)Kamailio1 (UDP)-> (UDP)Kamailio2 (TCP)-> (TCP)Provider UAC : 10.2.1.160 NATED to 100.100.100.61 <http://50.50.50.50:5065/> (TLS) Kamailio1 Sockets: 50.50.50.210:5061 <http://170.170.170.210:5061/> (TLS), 50.50.50.210:5060 <http://170.170.170.210:5060/> (UDP) Kamailio2 Sockets: 50.50.50.182:5060 <http://170.170.170.170:5060/> (UDP) 50.50.50.182:5080 <http://170.170.170.170:5060/> (TCP) Provider: 200.200.200.209:5060 <http://200.200.200.193:5060/> (TCP) We use dispatcher on Kamailio1 & Kamailio2 The invite goes fine from UCA up to the provider and the provider responses arrive to UAC fine. The PRACK from UAC to kamailio1 goes fine to Kamaili02 but, Kamailio2 uses the UDP socket to send to the provider so the PRACK does not arrive to the provider and the call is canceled. If the provider is on UDP everything works fine. Below I have the trace. What should I look for in the config to resolve this wrong protocol/socket on the PRACK? Kamailio2 Details: Sockets ====== listen=udp:50.50.50.182:5060 listen=tcp: 50.50.50.182:5080 On the rr module I have ================== modparam("rr", "enable_double_rr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_full_lr", 1) CONFIG REQINIT & WITHINDLG PARTS ================================ route[REQINIT] { set_reply_no_connect(); force_rport(); if (!mf_process_maxfwd_header("10")) { xlog("L_ERROR","Too Many Hops Rejecting!"); sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp sanity 1511\n"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp sanity 17895\n"); exit; } } route[WITHINDLG] { if (!has_totag()) return; if (loose_route()) { if (is_method("BYE")) { rtpengine_query(); rtpengine_delete(); } route(RELAY_OUT); } if ( is_method("ACK") ) { #if (t_lookup_request()) { if (t_check_trans()) { route(RELAY_OUT); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not Allowed"); exit; } TRACE ====== 50.50.50.210:5060 50.50.50.182:5060 50.50.50.182:32818 200.200.200.209:5060 qqqqqqqqqqwqqqqqqq qqqqqqqqqqwqqqqqqq qqqqqqqqqqwqqqqqqqq qqqqqqqqqqqwqqqqqqqqq a 15:31:11.787490 x INVITE (SDP) x x x a +0.003099 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x x a 15:31:11.790589 x 100 trying -- your call is x x x a +0.002220 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x x a 15:31:11.792809 x x x INVITE (SDP) x a +0.020725 x x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x a 15:31:11.813534 x x x 100 Trying x a +2.628884 x x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x a 15:31:14.442418 x x x 183 Session Progress (SDP) x a +0.001572 x x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x a 15:31:14.443990 x 183 Session Progress (SDP) x x x a +0.299221 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x x a 15:31:14.743211 x PRACK x x x a +0.043114 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x x a 15:31:14.786325 x x PRACK x a +0.448104 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq> x a 15:31:15.234429 x PRACK x x x a +0.045697 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x x a 15:31:15.280126 x x PRACK x a +0.954153 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x a 15:31:16.234279 x PRACK x x x a +0.045832 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x x a 15:31:16.280111 x x PRACK x a +1.954170 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x a 15:31:18.234281 x PRACK x x x a +0.045839 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x x a 15:31:18.280120 x x PRACK x a +1.829676 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x a 15:31:20.109796 x x x 183 Session Progress (SDP) x a +0.001460 x x x <<<qqqqqqqqqqqqqqqqqqqqqqqq x a 15:31:20.111256 x 183 Session Progress (SDP) x x x a +2.123075 x <<<qqqqqqqqqqqqqqqqqqqqqqqq x x x a 15:31:22.234331 x PRACK x x x a +0.045825 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x x a 15:31:22.280156 x x PRACK x a +2.239640 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x a 15:31:24.519796 x x x 486 Busy Here x a +0.000555 x x x <qqqqqqqqqqqqqqqqqqqqqqqqqq x a 15:31:24.520351 x x x ACK x a +0.008965 x x x qqqqqqqqqqqqqqqqqqqqqqqqqq> x a 15:31:24.529316 x 486 Busy Here x x x a +0.000835 x <qqqqqqqqqqqqqqqqqqqqqqqqqq x x x a 15:31:24.530151 x ACK x x x a +1.704184 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x x a 15:31:26.234335 x PRACK x x x a +0.045856 x qqqqqqqqqqqqqqqqqqqqqqqqqq> x x x a 15:31:26.280191 x x PRACK x a +3.954734 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x a 15:31:30.234925 x PRACK x x x x +0.045279 x qqqqqqqqqqqqqqqqqqqqqqqq>>> x x x x 15:31:30.280204 x x PRACK x x +3.954116 x x qqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq>>> x x 15:31:34.234320 x PRACK x x x 2021/09/21 15:31:11.787490 50.50.50.210:5060 -> 50.50.50.182:5060 INVITE sip:123456789@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:50.50.50.210;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Record-Route: <sip:50.50.50.210:5061 ;transport=tls;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias Max-Forwards: 69 From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com> Contact: "MyUserNAme" <sip:mycustomerid@10.2.1.160:58568 ;transport=TLS;ob;alias=100.100.100.61~48572~3> Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.20.5 Content-Type: application/sdp Content-Length: 326 P-Asserted-Identity: <sip:987654321@sip.mydomain.com> v=0 o=- 3841227071 3841227071 IN IP4 50.50.50.215 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 36448 RTP/AVP 8 0 101 c=IN IP4 50.50.50.215 b=TIAS:64000 a=ssrc:2024148033 cname:4f2468fd611910fa a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:36449 2021/09/21 15:31:11.790589 50.50.50.182:5060 -> 50.50.50.210:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3;rport=5060 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com> Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Server: myserver Content-Length: 0 2021/09/21 15:31:11.792809 50.50.50.182:32818 -> 200.200.200.209:5060 INVITE sip:123456789@sip.mydomain.com:5060 SIP/2.0 Record-Route: <sip:50.50.50.182:5080 ;transport=tcp;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.182;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.210;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Record-Route: <sip:50.50.50.210:5061 ;transport=tls;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Via: SIP/2.0/TCP 50.50.50.182:5080 ;branch=z9hG4bK37fe.254ecab744d678154779037580c59731.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias Max-Forwards: 68 From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com> Contact: "MyUserNAme" <sip:mycustomerid@10.2.1.160:58568 ;transport=TLS;ob;alias=100.100.100.61~48572~3> Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 326 P-Asserted-Identity: <sip:987654321@sip.mydomain.com> v=0 o=- 3841227071 3841227071 IN IP4 50.50.50.183 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 29308 RTP/AVP 8 0 101 c=IN IP4 50.50.50.183 b=TIAS:64000 a=ssrc:2024148033 cname:4f2468fd611910fa a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:29309 2021/09/21 15:31:11.813534 200.200.200.209:5060 -> 50.50.50.182:32818 SIP/2.0 100 Trying Via: SIP/2.0/TCP 50.50.50.182:5080 ;branch=z9hG4bK37fe.254ecab744d678154779037580c59731.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Content-Length: 0 2021/09/21 15:31:14.442418 200.200.200.209:5060 -> 50.50.50.182:32818 SIP/2.0 183 Session Progress Via: SIP/2.0/TCP 50.50.50.182:5080 ;branch=z9hG4bK37fe.254ecab744d678154779037580c59731.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Record-Route: <sip:50.50.50.182:5080 ;transport=tcp;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.182:5060 ;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.210:5060 ;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Record-Route: <sip:50.50.50.210:5061 ;transport=tls;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Contact: <sip:123456789@200.200.200.209:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Require: 100rel RSeq: 1509 Content-Length: 241 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 12038 7929 IN IP4 200.200.200.209 s=SIP Media Capabilities c=IN IP4 200.200.200.212 t=0 0 m=audio 17366 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 2021/09/21 15:31:14.443990 50.50.50.182:5060 -> 50.50.50.210:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK37fe.f05cee0feff15d488a12d3758cece7ae.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj472f07568b924214ada83ddfb0fb12b6;alias From: 987654321 <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14320 INVITE Record-Route: <sip:50.50.50.182:5080 ;transport=tcp;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.182:5060 ;r2=on;lr=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Record-Route: <sip:50.50.50.210:5060 ;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Record-Route: <sip:50.50.50.210:5061 ;transport=tls;r2=on;lr;ftag=73622588a6084d8ba1bca54781e5094d;vsf=AAAAAFpUAgECBwAABnEBB3ZdWEIAQkVFS0BHX2ljZS5ldQ--;vst=AAAAAAAAAAAKCQ8ABAkHCAYKAgdxQFpGbkJCRktDREFYU1AAV0Qw;did=e8e.bb7;nat=yes> Contact: <sip:123456789@200.200.200.209:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Require: 100rel RSeq: 1509 Content-Length: 253 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 12038 7929 IN IP4 50.50.50.183 s=SIP Media Capabilities c=IN IP4 50.50.50.183 t=0 0 m=audio 29320 RTP/AVP 8 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:29321 2021/09/21 15:31:14.743211 50.50.50.210:5060 -> 50.50.50.182:5060 PRACK sip:123456789@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 69 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK Route: <sip:50.50.50.182:5060 ;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Route: <sip:50.50.50.182:5080 ;transport=tcp;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:14.786325 50.50.50.182:5060 -> 200.200.200.209:5060 PRACK sip:123456789transport=tcp@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.182;branch=z9hG4bK47fe.3406fe45bb75c3e955b6aa67e53aff58.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 68 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:15.234429 50.50.50.210:5060 -> 50.50.50.182:5060 PRACK sip:123456789@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 69 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK Route: <sip:50.50.50.182:5060 ;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Route: <sip:50.50.50.182:5080 ;transport=tcp;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:15.280126 50.50.50.182:5060 -> 200.200.200.209:5060 PRACK sip:123456789transport=tcp@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.182;branch=z9hG4bK47fe.3406fe45bb75c3e955b6aa67e53aff58.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 68 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:16.234279 50.50.50.210:5060 -> 50.50.50.182:5060 PRACK sip:123456789@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 69 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK Route: <sip:50.50.50.182:5060 ;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Route: <sip:50.50.50.182:5080 ;transport=tcp;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:16.280111 50.50.50.182:5060 -> 200.200.200.209:5060 PRACK sip:123456789transport=tcp@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.182;branch=z9hG4bK47fe.3406fe45bb75c3e955b6aa67e53aff58.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 68 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK RAck: 1509 14320 INVITE Content-Length: 0 2021/09/21 15:31:18.234281 50.50.50.210:5060 -> 50.50.50.182:5060 PRACK sip:123456789@200.200.200.209:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.210;branch=z9hG4bK47fe.067d9577e9ab3e30e01745a2cf73b7a4.0;i=baa3 Via: SIP/2.0/TLS 10.2.1.160:58568 ;received=100.100.100.61;rport=48572;branch=z9hG4bKPj9ab94c339144428bb7ada63b133f2c12;alias Max-Forwards: 69 From: "MyUserNAme" <sip:987654321@sip.mydomain.com >;tag=73622588a6084d8ba1bca54781e5094d To: <sip:123456789@sip.mydomain.com>;tag=gK0d8c9073 Call-ID: 846ffb2f076848c5b8edc876fa1c0d9b CSeq: 14321 PRACK Route: <sip:50.50.50.182:5060 ;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> Route: <sip:50.50.50.182:5080 ;transport=tcp;lr;r2=on;ftag=73622588a6084d8ba1bca54781e5094d;did=e8e.c4d1> RAck: 1509 14320 INVITE Content-Length: 0 Any help will be much appreciated. Kind regards, Angelo

2 7

Releasing Kamailio 5.3.9
by Henning Westerholt 23 Sep '21

23 Sep '21

Hello, I am planning to release in two weeks (likely on Wednesday 7th October) the next minor version out of branch 5.3, version 5.3.9. Backports of fixed issues from other stable branches will be integrated as usual. If you are aware of new issues which were not yet reported on the issue tracker or are missing back-ports, report them. Please note that this will be the last minor release created from the 5.3.x stable branch. The stable branches are usually maintained from the Kamailio development team for a period of two years. The other two stable branches of Kamailio will still be maintained (5.4.x and 5.5.x). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Kamailio v5.4.7 Released
by Daniel-Constantin Mierla 22 Sep '21

22 Sep '21

Hello, Kamailio SIP Server v5.4.7 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.7. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.2 being release out of it. For more details about version 5.4.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/09/kamailio-v5-4-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 0

Add a new trunk
by sondes boughdiri 22 Sep '21

22 Sep '21

Good morning , I have a question , how to add a trunk from operators in kamailio. Best regards .

1 0

Re: [SR-Users] How to get $fU and $tU from MSRP module?
by Henning Westerholt 22 Sep '21

22 Sep '21

(please keep the list in CC) Hello, I see – its kind of pseudo-SIP message that is presented in the cfg in this route. And it seems that you want to access the cpim body. You already checked that the $msrp(crhop), $msrp(nexthop) and $msrp(prevop) etc.. are not working for you? It’s probably not the best way, but what about using $msrp(body) to get the body and then using a regular expression (like a transformation) on this to get the From/To? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: BOEINGNG <lihang567890(a)126.com> Sent: Wednesday, September 22, 2021 2:41 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] How to get $fU and $tU from MSRP module? Yes, I logged, but only got <null> for both $fU and $tU. 2021年9月20日下午4:03，Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> 写道： Hello, I am not that familiar with the MSRP module, but have you tried to just log/access the $fU and $tU in the event_route[msrp:frame-in]? 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 BOEINGNG Sent: Friday, September 17, 2021 10:57 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] How to get $fU and $tU from MSRP module? Hello! I want to get $fU and $tU from event_route[msrp:frame-in] in order to filter out MSRP messages sent by a specific sender and received by a specific receiver, I know I can get message content from $msrp(buf), the message content is like this: MSRP 7a57f552155926c0 SEND To-Path: msrp://172.18.0.14:6060/9d8da4a34dc2102959e7;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-84-1;tcp msrp://172.18.0.2:34862/2a67b54ae0750b7d82aa;tcp From-Path: msrp://172.18.0.14:6060/c71ae3675e114d08216b;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-83-1;tcp msrp://172.18.0.2:34858/8baea7c52ba8755322fd;tcp Byte-Range: 1-197/197 Failure-Report: yes Success-Report: yes Message-ID: c792c94039b8a969 Content-Type: message/cpim From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 MIME-Version: 1.0 Content-Type: text/plain hello -------7a57f552155926c0$ I want to get alice and bob from the content below： From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 someone can suggest some method. In other route like request_route we can use $fU and $tU to access them. Regrads HANG LI

1 0

Unable to pull enviornment variables
by Adarsh Chauhan 22 Sep '21

22 Sep '21

Hi Team, Need some help here. Want to pull environment variable however kamailio just doesnt pick it. I exported the environment variable *export STAGE="dev"* Then i tried following 1) *#!defenv DEVSTAGE* gives error 0(3809) ERROR: <core> [core/cfg.lex:2085]: pp_define_env(): env variable not defined [DEVSTAGE] 2) *#!substdef "/DEVSTAGE/$env(DEVSTAGE)/";* 0(3962) DEBUG: <core> [core/ppcfg.c:87]: pp_subst_add(): ### added subst expression: [/DEVSTAGE/$env(DEVSTAGE)/] 0(3962) DEBUG: <core> [core/cfg.lex:1964]: pp_define(): defining id: DEVSTAGE 0(3962) DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 0(3962) DEBUG: <core> [core/parser/msg_parser.c:677]: parse_msg(): SIP Request: 0(3962) DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): method: <OPTIONS> 0(3962) DEBUG: <core> [core/parser/msg_parser.c:681]: parse_msg(): uri: <sip:you@kamailio.org> 0(3962) DEBUG: <core> [core/parser/msg_parser.c:683]: parse_msg(): version: <SIP/2.0> 0(3962) DEBUG: <core> [core/parser/parse_hname2.c:293]: parse_sip_header_name(): parsed header name [Via] type 1 0(3962) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=2 0(3962) DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 0(3962) DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via 0(3962) DEBUG: <core> [core/pvapi.c:497]: pv_spec_lookup(): PV <$env(DEVSTAGE)> is not in cache 0(3962) DEBUG: <core> [core/pvapi.c:335]: pv_cache_add(): PV cache not initialized, doing it now 0(3962) DEBUG: <core> [core/pvapi.c:371]: pv_cache_add(): pvar [$env(DEVSTAGE)] added in cache 0(3962) DEBUG: <core> [core/cfg.lex:2053]: pp_define_set(): ### setting define ID [DEVSTAGE] value [<null>] (mode: 1) 0(3962) DEBUG: <core> [core/ppcfg.c:188]: pp_substdef_add(): ### added substdef: [DEVSTAGE]=[<null>] (0) 0(3962) DEBUG: <core> [core/re.c:436]: subst_run(): running. r=1 0(3962) DEBUG: <core> [core/re.c:504]: subst_str(): no match 0(3962) DEBUG: <core> [core/re.c:436]: subst_run(): running. r=1 0(3962) DEBUG: <core> [core/re.c:504]: subst_str(): no match my end goal is to compare the directive and load modules accordingly. *#!ifdef DEVSTAGE == "dev"*

3 4

Releasing Kamailio v.5.4.7
by Daniel-Constantin Mierla 22 Sep '21

22 Sep '21

Hello, I am planning to release a new version from branch 5.4, respectively 5.4.7, sometime next week, likely on Wednesday, Sep 22, 2021. If anyone is aware of issues not yet reported to bug tracker or missing backports, report them in order to try to get the fixes in this release. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

JSONRPC ERROR in siremis GUI.
by 岩間理佐 22 Sep '21

22 Sep '21

Hi, I have a question regarding the jsonRPC commands in the siremis GUI. 【Environment】・Centos7 ・Kamailio 5.5.2 ・siremis 5.3 (installed by git clone https://github.com/asipto/siremis siremis-5.3.x) ・php php-mysqlnd-7.1.33-16.el7.remi.x86_64 php-json-7.1.33-16.el7.remi.x86_64 php-mbstring-7.1.33-16.el7.remi.x86_64 php-pdo-7.1.33-16.el7.remi.x86_64 php-gd-7.1.33-16.el7.remi.x86_64 php-xml-7.1.33-16.el7.remi.x86_64 php-pecl-zip-1.19.4-1.el7.remi.7.1.x86_64 php-xmlrpc-7.1.33-16.el7.remi.x86_64 php-cli-7.1.33-16.el7.remi.x86_64 php-process-7.1.33-16.el7.remi.x86_64 php-pear-1.10.13-1.el7.remi.noarch php-common-7.1.33-16.el7.remi.x86_64 php-7.1.33-16.el7.remi.x86_64 When I execute commands in command service →JSONRPC commands from siremis GUI, it always shows error page. I tried some approaches <https://eow.alc.co.jp/search?q=approaches&ref=awlj> referring to this page, https://github.com/asipto/siremis/issues/26 but I still have this error message in /var/log/httpd/error_log > [Tue Sep 21 16:43:26.396967 2021] [php7:warn] [pid 6181] [client > 152.165.116.138:55822] PHP Warning: strstr() expects parameter 1 to be > string, resource given in > /var/www/siremis-5.3.x/openbiz/bin/ErrorHandler.php on line 158, referer: > http://kamailio.xcallyjp.com/siremis/sipadmin/jrcmds > [Tue Sep 21 16:43:26.401139 2021] [php7:notice] [pid 6181] [client > 152.165.116.138:55822] u socket send failed: Permission denied, referer: > http://kamailio.xcallyjp.com/siremis/sipadmin/jrcmds I have tried my best to do some researches about it, but still stuck. Any help is appreciated. Thank you! Lisa Iwama

2 1

Question regarding parser error for REGISTER request
by Floimair Florian 20 Sep '21

20 Sep '21

Hello List! We have a certain user agent that somehow triggers a parser error log entry, everytime this type of user agent registers. Sep 20 12:03:49 kamailio-dev-1 /usr/sbin/kamailio[1803]: ERROR: <core> [core/parser/parse_param.c:517]: parse_param2(): Invalid character, ; expected The odd thing now is that this register still succeeds. And we do not see that error for other methods like e.g. INVITE. The user agent in question is an Android app based on linphone which we develop inhouse. We also use the same for iOS where the error does not occur. The REGISTER request contains a push-token in the Contact header and I think this is where the parser somehow generates this error. Since this is the only line apart from the User-Agent header where the content of the two is different. We did have a look and diffed the differences between our iOS and Android app variants and didn’t see any obvious problem or invalid character (compared even with hex editor). So my questions regarding this I hope that someone of you can help me with: * Is there any way to make the parser error more verbose (e.g. which character in which header it is complaining about)? Right now it only says that there was an error and that it expected ‘;’ but not where and what character was found instead. * Why does the parser throw the error but the registration still succeeds? It does not quite make sense to me. Best Regards FLORIAN FLOIMAIR Development Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria Phone: +43 662 85 62 25 Mail: f.floimair(a)commend.com<mailto:f.floimair@commend.com> [signature_1687123261] commend.com LG Salzburg / FN 178618z

2 1

Question on the SL Module
by Alessio Garzi 20 Sep '21

20 Sep '21

Hello Kama ML, today I stumbled on function "update_sl_stats()" inside sl_stats.c and noticed that some response codes are treated differently. For example here: https://github.com/kamailio/kamailio/blob/a9ef78b7f43616f2ceaa26df97eff68eb… since we have a "break;" statement, when a 500 code is received, the 5XX counter is not incremented even if a 500 code belongs to the 5XX family. We have the same behaviour for other response codes and it really seems that it's not a bug and the author really wanted it. The docs are not so helpful and they don't explain much in detail how counters are built. I just want to have confirmation this is how it must work and I wonder if it's the case to write something about it into the docs. Thanks Alessio

2 1

Kamilio Presence
by Jerry Kendall 20 Sep '21

20 Sep '21

Hey Bill, Just checking to see if you managed to get the BLFs to work on the Yealinks... I've Grandstream/Yealink/Polycom devices and can't get BLFs to work - the NOTIFY is not being sent. regards, Jerry __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 2

Kamailio - FreeSWITCH - RTPproxy
by Jerry Kendall 20 Sep '21

20 Sep '21

Hello all, I am new to the list but have been reading plenty. Loads of good tips and help. I am playing with Freeswitch and would like to put Kamailio/RTPProxy in front for the user endpoints. Freeswitch will talk directly with the service provider. I am wondering *if* i can do this... 1) have the endpoints register with kamailio 2) have the media proxy via RTPProxy as it is needed for NAT 3) have the actual work (voicemail/ring groups/ivr/conference/etc...) done on Freeswitch 4) have all the SUBSCRIBE/PUBLISH/PRESENCE done on Kamailio Is this doable? Are there any guides to setting this sort of thing up? regards, Jerry

3 3

How to get $fU and $tU from MSRP module?
by BOEINGNG 20 Sep '21

20 Sep '21

Hello! I want to get $fU and $tU from event_route[msrp:frame-in] in order to filter out MSRP messages sent by a specific sender and received by a specific receiver, I know I can get message content from $msrp(buf), the message content is like this: MSRP 7a57f552155926c0 SEND To-Path: msrp://172.18.0.14:6060/9d8da4a34dc2102959e7;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-84-1;tcp msrp://172.18.0.2:34862/2a67b54ae0750b7d82aa;tcp From-Path: msrp://172.18.0.14:6060/c71ae3675e114d08216b;tcp msrp://172.18.0.12:5060/msrp-6142fc2b-83-1;tcp msrp://172.18.0.2:34858/8baea7c52ba8755322fd;tcp Byte-Range: 1-197/197 Failure-Report: yes Success-Report: yes Message-ID: c792c94039b8a969 Content-Type: message/cpim From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 MIME-Version: 1.0 Content-Type: text/plain hello -------7a57f552155926c0$ I want to get alice and bob from the content below： From: sip:alice@ims.mnc001.mcc001.3gppnetwork.org To: sip:bob@ims.mnc001.mcc001.3gppnetwork.org DateTime: 2021-09-16T09:23:49.339805+00:00 someone can suggest some method. In other route like request_route we can use $fU and $tU to access them. Regrads HANG LI

2 1

Reinite breaks the call
by Angelo Sipper 20 Sep '21

20 Sep '21

Hello, I am trying to understand a problem we have with reinvites. When there is a reinvite on the call our provider refuses to receive the reinvite and the call is dropped. The setup is the bellow: UA-> Kamailio1 -> Kamailio2 -> Provider On Kamailio2 we have Topos enabled. Looking on the trace I can see that on the second invite, Kamailio does send out the original contact and not the one that was originally send out to the provider. Is this correct? Is this breaking the call? Bellow the trace and the flow from Kamailio2 until the second reinvite. All reinvites are the same as per the last Thank you for your help! 2021/09/15 16:29:59.420162 50.50.50.181:5060 -> 200.200.200.193:5060 INVITE sip:1234567890@sip.mydomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 Max-Forwards: 67 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com> Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 481 Contact: <sip:btpsh-b5-6141f513-11f69-4@50.50.50.181> v=0 o=FreeSWITCH 1631683257 1631683258 IN IP4 50.50.50.183 s=FreeSWITCH c=IN IP4 50.50.50.183 t=0 0 m=audio 26472 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:26473 a=ptime:20 m=video 26510 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=rtcp-fb:103 ccm fir a=rtcp-fb:103 ccm tmmbr a=rtcp-fb:103 nack a=rtcp-fb:103 nack pli a=sendrecv a=rtcp:26511 ====================================================================================== 2021/09/15 16:29:59.439009 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Content-Length: 0 ====================================================================================== 2021/09/15 16:30:02.596613 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Contact: <sip:1234567890@200.200.200.193:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Content-Length: 313 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 27823 25478 IN IP4 200.200.200.193 s=SIP Media Capabilities c=IN IP4 200.200.200.196 t=0 0 m=audio 19958 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 m=video 0 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=inactive ====================================================================================== 2021/09/15 16:30:06.163783 200.200.200.193:5060 -> 50.50.50.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.b5141bfea33b57c5223aff14ab3b5466.0 From: 987654321 <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 INVITE Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Contact: <sip:1234567890@200.200.200.193:5060> Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS P-Asserted-Identity: <sip:6979111336@MOBILE.MOBILEPROVIDER.COM;user=phone> Require: timer Supported: timer Session-Expires: 300;refresher=uac Content-Length: 313 Content-Disposition: session; handling=required Content-Type: application/sdp v=0 o=Sonus_UAC 27823 25478 IN IP4 200.200.200.193 s=SIP Media Capabilities c=IN IP4 200.200.200.196 t=0 0 m=audio 19958 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 m=video 0 RTP/AVP 103 b=AS:1024 a=rtpmap:103 H264/90000 a=inactive ====================================================================================== 2021/09/15 16:30:06.405147 50.50.50.181:5060 -> 200.200.200.193:5060 ACK sip:1234567890@200.200.200.193:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK3db6.58e9d8820221b191b49fe0c91299e56f.0 Max-Forwards: 68 From: "987654321" <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302510 ACK Proxy-Authorization: Digest username="mycustomerid", realm=" sbc1.mydomain.com", nonce="YUH2g2FB9VcCxZ3l2WSvJCj7aUokLhAKksYQ1zfOfyQRWm90h2vJgH8I/B+A", algorithm=MD5, uri="sip:1234567890@sbc1.mydomain.com", response="dc4d3d7e9281c04ecc3e3 63ea162fb" Content-Length: 0 Contact: <sip:btpsh-b5-6141f513-11f69-4@50.50.50.181> ============================================= 2021/09/15 16:32:37.685497 50.50.50.181:5060 -> 200.200.200.193:5060 INVITE sip:1234567890@200.200.200.193:5060 SIP/2.0 Via: SIP/2.0/UDP 50.50.50.181;branch=z9hG4bK4db6.691137b194a14953af928c339e03d60e.0 Via: SIP/2.0/UDP 50.50.50.210;rport=5060;branch=z9hG4bK4db6.4619cd1b550ba7e45a3fbd0321a5c0b0.0 Via: SIP/2.0/UDP 62.1.2.62;received=62.1.2.62;rport=5060;branch=z9hG4bKXDZQF19XUj3QB Max-Forwards: 67 From: "987654321" <sip:987654321@sip.mydomain.com>;tag=2QK2mpeH5Sy4g To: <sip:1234567890@sip.mydomain.com>;tag=gK0dc15c15 Call-ID: dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq: 41302511 INVITE Contact: <sip:mycustomerid@62.1.2.62:5060 ;transport=udp;gw=1606bb6a-6563-43be-b4dd-4890d524d0e9> User-Agent: Supervoice VPBX Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Proxy-Authorization: Digest username="mycustomerid", realm=" sbc1.mydomain.com", nonce="YUH2g2FB9VcCxZ3l2WSvJCj7aUokLhAKksYQ1zfOfyQRWm90h2vJgH8I/B+A", algorithm=MD5, uri="sip:atpsh-b5-6141f513-11f69-4@50.50.50.181", response="50ed75ca9002 ca76fbfa306fb8a599" Session-Expires: 300;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 348 Remote-Party-ID: "987654321" <sip:987654321@sbc1.mydomain.com >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1631683257 1631683258 IN IP4 50.50.50.183 s=FreeSWITCH c=IN IP4 50.50.50.183 t=0 0 m=audio 26472 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=rtcp:26473 a=ptime:20 m=video 0 RTP/AVP 103 a=rtpmap:103 H264/90000 |Time | 50.50.50.210 | 200.200.200.193 | | | | 50.50.50.181 | |0.000024 | INVITE SDP (g711A g7 | |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> Call-ID:dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq:41302510 | |(5060) ------------------> (5060) | | |0.001523 | 100 trying -- your c | |SIP Status 100 trying -- your call is important to us | |(5060) <------------------ (5060) | | |0.011760 | | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302510 | | |(5060) ------------------> (5060) | |0.030607 | | 100 Trying| |SIP Status 100 Trying | | |(5060) <------------------ (5060) | |2.690501 | | 183 Session Progress |SIP Status 183 Session Progress | | |(5060) <------------------ (5060) | |2.708868 | 183 Session Progress | |SIP Status 183 Session Progress | |(5060) <------------------ (5060) | | |3.188211 | | 183 Session Progress |SIP Status 183 Session Progress | | |(5060) <------------------ (5060) | |3.199099 | 183 Session Progress | |SIP Status 183 Session Progress | |(5060) <------------------ (5060) | | |6.755381 | | 200 OK SDP (g711A te |SIP Status 200 OK | | |(5060) <------------------ (5060) | |6.819017 | 200 OK SDP (g711A te | |SIP Status 200 OK | |(5060) <------------------ (5060) | | |6.973845 | ACK | | |SIP Request INVITE ACK 200 CSeq:41302510 | |(5060) ------------------> (5060) | | |6.996745 | | ACK | |SIP ACK From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302510 | | |(5060) ------------------> (5060) | |158.269723| INVITE SDP (g711A g7 | |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> Call-ID:dfb297d1-90cb-123a-d4a1-00155d01bf11 CSeq:41302511 | |(5060) ------------------> (5060) | | |158.275396| 100 trying -- your c | |SIP Status 100 trying -- your call is important to us | |(5060) <------------------ (5060) | | |158.277105| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |158.753781| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |159.753493| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |161.753502| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |165.753630| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |169.754980| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |173.753719| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |177.753676| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |181.753657| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |185.753503| | INVITE SDP (g711A g7 |SIP INVITE From: 1234567890 <sip:1234567890@sip.mydomain.com> To:< sip:987654321@sip.mydomain.com> CSeq:41302511 | | |(5060) ------------------> (5060) | |188.254337| 503 Service not avai | |SIP Status 503 Service not available | |(5060) <------------------ (5060) | | |188.255085| ACK | | |SIP Request INVITE ACK 200 CSeq:41302511 | |(5060) ------------------> (5060) | | |188.326552| BYE | | |SIP Request BYE CSeq:41302512 | |(5060) ------------------> (5060) | | |188.342106| | BYE | |SIP Request BYE CSeq:41302512 | | |(5060) ------------------> (5060) | |188.373697| | 200 OK | |SIP Status 200 OK | | |(5060) <------------------ (5060) | |188.379670| 200 OK | | |SIP Status 200 OK | |(5060) <------------------ (5060) | | Kind Regards, Angelo

4 4

Re: [SR-Users] Error in ul_insert_ucontact() function of usrloc module
by hamed r 20 Sep '21

20 Sep '21

hello Ben I was using development functions and the C code of this function did not exist. I needed to manually add a contact to kamailio but this does not work and I used the HTable module to solve the problem. Thanks for the tips

1 0

Slack channel
by Sergio Carvalho 20 Sep '21

20 Sep '21

Hi everyone, How can I join the Kamailio slack channel? Is it open to the community? Best regards, Sergio Carvalho

3 2

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