- sr-users - kamailio.org

Releasing v5.8.4
by Daniel-Constantin Mierla 12 Nov '24

12 Nov '24

Hello, I am considering to release Kamailio v5.8.4 (out of branch 5.8) sometime next week, most likely on Tuesday, Nov 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

6 10

fail2ban complains about Kamailio syslog messages
by Juha Heinanen 12 Nov '24

12 Nov '24

When Kamailio message parser produces this kind of syslog message: 024-11-11T21:16:40.757364+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\340#001> fail2ban complains: Consider setting logencoding to appropriate encoding for this jail. Continuing to process line ignoring invalid characters: b'2024-11-11T21:06:15.347318+02:00 box /usr/bin/sip-proxy[181398]: ERROR: <core> [core/parser/msg_parser.c:791]: parse_msg(): ERROR: parse_msg: message=<#026#003#001#006\xc0#001>\n': 1 Time(s) Error decoding line from '/var/log/syslog' with 'UTF-8'.: 1 Time(s) Unable to retrieve Watch object associated to <_RawEvent cookie=0 mask=0x2 name='' wd=13 >: 3 Time(s) Would it be possible to somehow ensure that Kamailio syslog messages only contain valid UTF-8 characters? -- Juha

1 0

Last line of onreply_route (drops reply)
by James Browne 11 Nov '24

11 Nov '24

Hi all I noticed, in my config, that if the last line of the reply_route is a function that does not succeed, then the SIP response is not relayed and kamailio throws an error. I've searched documentation to indicate why this might be, but I can't find anything, so this looks to me like a bug. I use 5.8.1 for testing. Here's a sample config that shows what goes wrong. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "textops" request_route{exit;} reply_route{remove_hf_re("^X-Custom");} #################### Here's a sample SIP message that works. When I send this to udp:203.0.113.57:5060, kamailio relays it. #################### SIP/2.0 200 Testing Via: SIP/2.0/UDP 203.0.113.57:5060;branch=z9hG4bK-d8754z Via: SIP/2.0/UDP 198.18.0.157:5060;branch=z9hG4bK-d8754z T: <sip:c@s>;tag=676fbf45 F: <sip:c@s>;tag=676fbf45 X-Custom: lkjdfl i: testing-XXX CSeq: 1 INFO l: 0 #################### If I then remove the X-Custom header field from the SIP reply and try again, the message does not get relayed and I see this in the log. 5(1356) WARNING: <core> [core/receive.c:605]: receive_msg(): error while trying onreply script If I put a log command after the remove_hf_re(), then the message gets relayed. It looks like the reply_route always fails if the last function is not successful. Is this normal or is it a bug? James

3 5

Question regarding error message "failed to add new socket to the fd list"
by Floimair Florian 11 Nov '24

11 Nov '24

Hi! We have recently had issues with one of our Production Kamailios. When those happened, the log was filled with the following message: CRITICAL: <core> [core/tcp_main.c:4528]: handle_new_connect(): failed to add new socket to the fd list Now I wonder what the best approach is to prevent this. We are using TCP/TLS only and I think this might be related to the file ulimit, but I am not sure about that. Shared memory is set to 512MB Can you give me a hint on what to look for? Thank you very much! P.S.: Sorry, I accidentally replied to a previous post of a different topic before which is totally unrelated (I think I should stop working for today 😉) FLORIAN FLOIMAIR Software Development - Symphony Cloud Services Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria [signature_2072127332] commend.com LG Salzburg / FN 178618z

2 1

apt-key deprecation
by tom＠tomlynn.com 11 Nov '24

11 Nov '24

When I look at the debian repositories maintained by the project, there are instructions to import the public key of the repository for apt using this command: wget -O http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - which results in: Warning: apt-key is deprecated. Manage keyring files in trusted.gpg.d instead (see apt-key(8)). gpg: no valid OpenPGP data found. Is there an alternate method that can be posted in the repository that will get better results? Thank you, TL

5 5

Unable to get body in multipart body
by Anthony Blandin 08 Nov '24

08 Nov '24

Hello, I want to get a body for content-type "application/isup" but I have the following error: 4(16) DEBUG: textops [textops.c:3394]: ki_get_body_part_helper(): found content type application/isup 4(16) ERROR: textops [textops.c:3397]: ki_get_body_part_helper(): no CRLF found after content type Here is the INVITE sent: INVITE sip:+883390000010022@[remote_ip]:[remote_port];user=phone SIP/2.0 From: <sip:+883390000010007@[local_ip];user=phone>;tag=[call_number] To: <sip:+883390000010022@[remote_ip];user=phone> Max-Forwards: 70 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] Route: <sip:[local_ip]:[local_port];transport=udp;lr> Record-Route: <sip:[local_ip]:[local_port];transport=udp;lr> Call-ID: [call_id] CSeq: 1 INVITE P-Asserted-Identity: <sip:+883390000010007@[local_ip]:[local_port];user=phone> Accept: application/sdp,application/isup,multipart/mixed Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,INFO,PRACK,UPDATE P-Charging-Vector: icid-value=IBCF-1-161460836711652742337;icid-generated-at=BC00.QX00.ORANGE-MSS.FR;orig-ioi=ORANGE-MSS.FR Supported: 100rel Content-Type: multipart/mixed;boundary=unique-boundary-1 MIME-Version: 1.0 Contact: <sip:[local_ip]:[local_port];transport=udp> Content-Length:[len] --unique-boundary-1 Content-Type: application/sdp v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 a=sendrecv m=audio [media_port] RTP/AVP 8 96 b=RR:0 b=RS:0 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=ptime:20 a=maxptime:40 --unique-boundary-1 Content-Type: application/isup;version=spirou;base=itu-t92+ Content-Disposition: signal;handling=required \x01\x11\x20\x01\x0a\x00\x02\x0c\x0a\x04\x10\x88\x33\x09\x00\x00\x01\x20\xf2\x0a\x0a\x84\x13\x88\x33\x09\x00\x00\x01\x00\x07\x1d\x03\x80\x90\xa3\x3f\x07\x83\x97\x26\x87\x41\x00\x00\x39\x04\x34\xc0\x3f\xc0\x00 --unique-boundary-1-- CRLF is well added at the end of Content-Type header, is anybody could help me to solve this issue? Regards, Anthony

1 0

[db_unixodbc] how to specify the database with unixodbc？
by 17603019528＠163.com 08 Nov '24

08 Nov '24

Hi Dear Community, I want to use db_unixodbc to connect to the dm8 database. Although I use Opensips, I found that this module db_unixodbc, Kamailio and Opensips are the same. My question is how to specify the database with kamailio？ I add "Database = sbc" into odbc.ini, But nothing happened。 ###### cat /usr/local/etc/odbc.ini [DM8] Description = ODBC DRIVER FOR DM8 Driver = DM8 ODBC DRIVER SERVER = 172.16.7.131 TCP_PORT = 5230 odbcinst -j unixODBC 2.3.12 DRIVERS............: /usr/local/etc/odbcinst.ini SYSTEM DATA SOURCES: /usr/local/etc/odbc.ini FILE DATA SOURCES..: /usr/local/etc/ODBCDataSources USER DATA SOURCES..: /root/.odbc.ini SQLULEN Size.......: 8 SQLLEN Size........: 8 SQLSETPOSIROW Size.: 8 ####### #################### modparam("usrloc", "db_url", "unixodbc://opensips:opensips#$@172.16.7.131:5230/dm8") #################### DBG:db_unixodbc:db_unixodbc_new_connection: opening connection: unixodbc://xxxx:xxxx@172.16.7.131/dm8 Nov 7 17:05:21 [145] DBG:db_unixodbc:db_unixodbc_new_connection: connection succeeded with reply <DSN=dm8;SERVER=172.16.7.131;UID=opensips;PWD=opensips#$;TCP_PORT=5230;> DBG:core:db_do_init: connection 0xfffd5bbbaf08 inserted in pool as 0xfffd5bbbb0e0 Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= select table_version from version where table_name='dialog' Nov 7 17:05:21 [145] ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=42S02:1:-2106:Error in line: 1 ####################

2 1

Draining TCP/TLS SIP sessions from Kamailio and media sessions from RTPEngine
by Evgenii Buchnev 07 Nov '24

07 Nov '24

Hi Dear Community, Would appreciate your input into where to look further to explore my options and would be good to learn your experience with the following. There are many active-active kamailio nodes behind tcp load balancer. On the same VM with each Kamailio there is an instance of RTPEngine to manage Media/Transcoding/Etc for that Kamailio. This scales UP easily - new nodes just added behind tcp load balancer. To be able to scale down and/or update node configuration - I'd like to be able to drain sessions/calls properly and pass any active calls from current Kamailio and RTPEngine to other nodes. DMQ is not yet in place, although it's planned. What would be proper ways of migrating RTP sessions to other RTPEngines? Caveats? Also wonder how properly a tcp/tls based SIP session can be passed to another Kamailio? Thank you in advance for your input. Evgenii Buchnev

3 2

Roadmap to v6.0.0 - the next major release
by Daniel-Constantin Mierla 07 Nov '24

07 Nov '24

Hello, about 8 months since the last major release v5.8.x, therefore it is the time to plan a bit the next steps for getting v6.0.0 out. The Kamailio Developers Meeting is schedule in Dusseldorf, Germany during November 19-20, 2024, where we expect to push out a consistent number of updates. Then maybe we can wait for another 2 weeks or so to settle down/finish those updates. After that, if no other new features are planned by developers in short term, we can enter the testing phase, so the new major version can be released during the second part of January 2025 or first part of February 2025. Therefore, if you have new features under development that might require more than 1-1.5 months from now, and you want to get them in v6.0.0, let us know to plan properly. Other time lines suggestions to get to v6.0.0 are also welcome! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 1

Kamailio 5.5.7 alias not added to ACK
by Mitesh Thakkar 06 Nov '24

06 Nov '24

Hi, I was using kamailio 5.2.5. I have recently upgraded to kamailio 5.5.7. After upgrade, the alias is stop being add to the ACK of reinvite from upstream. Before upgrade, it was working fine. I’ll appreciate any help on this. Thank you. Regards, Miteshkumar Thakkar

1 0

SNMP setup
by Sergio Charrua 05 Nov '24

05 Nov '24

Hi all! I am having difficulties setting up SNMP, and mostly to figure out if what I have setup is correctly configured. This is what I have on Kamailio (5.8.2), which is set as stateless and as a redirect server (simply put, it receives INVITE from SBC, gets a route from routing services, modifies Contact header, and replies to SBC with 300 Multi Choice with new contact header): loadmodule "db_mysql.so" loadmodule "db_cluster.so" loadmodule "http_client.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "jansson.so" loadmodule "snmpstats.so" loadmodule "file_out.so" loadmodule "ctl.so" loadmodule "permissions.so" loadmodule "xhttp.so" loadmodule "xhttp_rpc.so" # ---- SNMP Stats params ---- modparam("snmpstats", "sipEntityType", "proxyServer") modparam("snmpstats", "sipEntityType", "redirectServer") modparam("snmpstats", "sipEntityType", "other") modparam("snmpstats", "snmpgetPath", "/usr/bin/") modparam("snmpstats", "MsgQueueMinorThreshold", 1) modparam("snmpstats", "MsgQueueMajorThreshold", 1) modparam("snmpstats", "dlg_minor_threshold", 1) modparam("snmpstats", "dlg_major_threshold", 1) modparam("snmpstats", "snmpCommunity", "kamailio") I have set the *_minor_threshold values to bear minimum. Should I add anything else? The logs show the following messages: DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 60) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 64) returns address 0x7f03af3bcc50 frag. 0x7f03af3bcc10 (size=64) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:402]: qm_malloc(): qm_malloc(0x7f03af1ff010, 20) called from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:449]: qm_malloc(): qm_malloc(0x7f03af1ff010, 32) returns address 0x7f03af3bcd00 frag. 0x7f03af3bccc0 (size=32) on 1 -th hit DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcc50), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(464) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bcc10 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) DEBUG: <core> [core/mem/q_malloc.c:515]: qm_free(): qm_free(0x7f03af1ff010, 0x7f03af3bcd00), called from snmpstats: snmp_statistics.c: get_total_bytes_waiting(471) DEBUG: <core> [core/mem/q_malloc.c:562]: qm_free(): freeing frag. 0x7f03af3bccc0 alloc'ed from snmpstats: snmp_statistics.c: get_socket_list_from_proto_and_family(185) INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs INFO: snmpstats [utilities.c:116]: get_statistic(): failed to retrieve statistics for active_dialogs The SNMP daemon is set as follows: - /etc/snmp]$ cat snmptrapd.conf # Example configuration file for snmptrapd # # No traps are handled by default, you must edit this file! # # authCommunity log,execute,net public # traphandle SNMPv2-MIB::coldStart /usr/bin/bin/my_great_script cold and on /etc/snmp/snmpd.conf: ############################################################################### # Further Information # # See the snmpd.conf manual page, and the output of "snmpd -H". agentaddress udp:161 rocommunity kamailio master agentx agentXSocket tcp:localhost:705 trap2sink 10.X.Y.Z netcool 162 When running tcpdump and listening on port 162, no traffic is captured....it seems to me that Kamailio is not sending anything to SNMP Any clue, anyone? What am I doing wrong? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 3

Function sdp_get gets extra blank line (when MIME is used)
by James Browne 05 Nov '24

05 Nov '24

The sdp_get() function from sdpops returns an extra blank line sometimes, which looks like a bug to me. When I have an SDP that's in MIME, there's an extra newline at the end of the extracted SDP. Here's a sample config. I use 5.8.1 for testing. #################### #!KAMAILIO listen=udp:203.0.113.57:5060 debug=1 loadmodule "sdpops" loadmodule "pv" loadmodule "xlog" request_route{exit;} reply_route { sdp_get("$avp(sdp)"); xlog("L_NOTICE", "[$ci] The SDP is [$avp(sdp)].\n"); } #################### Here are two sample SIP-message bodies. #################### v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 #################### #################### --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2 Content-Type: application/sdp Content-Length: 130 v=0 o=PCD 1730 1730 IN IP4 198.18.114.148 s=PCD c=IN IP4 198.18.114.148 t=0 0 m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 --e61e7a6a-16a2-46c0-adbb-a41ba86d32a2-- #################### When the first body is received, the xlog correctly ends with this. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 130 When the second body is received, the xlog incorrectly ends with this, which has an extra newline in it. m=audio 29426 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ] with length 132 According to RFC2046, "The CRLF preceding the boundary delimiter line is conceptually attached to the boundary", but it looks like kamailio has a bug that incorrectly processes this. James

2 2

Simultaneous INVITE responses from multiple branches
by Anton Yabchinskiy 04 Nov '24

04 Nov '24

Hello, In our system we've encountered the following situation. There's a Kamailio proxy and a few REGISTER'ed devices behind the proxy. When there's an INVITE to these multiple devices, two of them simultaneously answer the call. Kamailio proxy accepts the first 200 OK from one of the devices and tries to CANCEL all other branches. But the second 200 OK device have already responded OK and doesn't acknowledge the CANCEL. So, there's now one established dialog with media, and another "established" one neither having media nor being correctly terminated. Looks like the similar scenario is described in section 3.1.2 of RFC 5407 (Example Call Flows of Race Conditions in the Session Initiation Protocol (SIP)). Is it possible to mitigate such scenarios in a Kamailio proxy? E.g., is it possible to send an additional BYE to these CANCEL'led branches? In current configuration, Kamailio proxy retrieves contacts with registrar:lookup() and does parallel forking with tm:t_relay().

2 2

Re: Dealing with flood of invalid SIP messages
by Juha Heinanen 04 Nov '24

04 Nov '24

I gave APIBAN a try. After a few days, no requests were banned by APIBAN, but during the same period, lots of request were banned by other checks in my config. That made me wonder if APIBAN really is effective. -- Juha

2 3

Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo
by matilopi17＠gmail.com 01 Nov '24

01 Nov '24

Hello, This is kamailio version 5.8.2 and we`re having this issue every now and then. The error displayed is: ** ERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo ** ERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters The problem is that the FIFO file /run/kamailio/kamailio_rpc.fifo doesn't exist, even though the "jsonrpc" module is in the Kamailio configuration, and the file is created when Kamailio service is restarted. But as mentioned, after few days the issue is repeated. I`ll appreciate any suggestion or advice of what I should check. Thank you!

4 5

Failed to parse the R-URI with topoh enabled
by Ale 29 Oct '24

29 Oct '24

Hi All, I think I have a problem with topoh, but I can't identify it. Kamailio 5.6.6, used as a stateless proxy, receives a 487 from a phone and propagates it correctly. Next comes the ack that should be forwarded back to the phone, but instead it doesn't forward it and produces the following error "pv_get_ruri(): failed to parse the R-URI" The error is generated by xlog("...$ru from $fu...\n"") as first line of request_route. If I disable topoh everything works perfectly.Topoh is only configured with mask_key only. Furthermore, not all phones generate this problem, at the moment there are some snoms. ACK sip:user@phone_ip:2048;line=kkqgarrj SIP/2.0 Via: SIP/2.0/UDP proxy_sip_ip;branch=z9hG4bK2ec.e7126a3134bef9d974f57dd43ebd4ea2.0 Route: <sip:kamailio_ip:5060;lr;received=sip:phone_ip:2048> Max-Forwards: 66 From: <sip:111111111@sip.example.com>;tag=pDc1BQ0B57Ujj To: <sip:user@sip.example.com>;tag=vslvowy4y2 Call-ID: 40a6c905-c0c8-4c20-b3f7-397b3fce58b6 CSeq: 90285584 ACK Content Length: 0 There is no difference between the ACK of the snom and other phones that work. The only difference I noticed is that the snom 487 contains Contact field in the header. Any suggestions? Thank you

3 5

How to handle e2e ACK with t_check_trans()
by Benoit Panizzon 28 Oct '24

28 Oct '24

Hi List To correctly route ACK belonging to a transaction (totag) the example code does: if ( is_method("ACK") ) { if ( t_check_trans() ) { route(RELAY); exit; } else { xlog("L_INFO", "$cfg(route): $rm NOT loose route NO transaction found DROP\n"); exit; } } Unfortunatels I fear that in a parallell forking scenario, the transaction is remove from memory as soon as the first 'negative' ACK is being processed so that the end 2 end ACK to the 200 OK of the branch that connected the call, never makes it through. I attempted to increase wt_timer to 10000 to have the transaction stay around 10 seconds but that does not solve the issue. I read that t_check_trans()'s result shall be considered more as a hint than as a confirmation of an existing transaction. Is there a clean / secure way to process e2e ACK and make sure they belong to a transaction? Or can I just safely remove the t_check_trans() call and route any ACK with a totag? Mit freundlichen Grüssen -Benoît Panizzon- -- 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

Dealing with flood of invalid SIP messages
by mayamatakeshi 27 Oct '24

27 Oct '24

Hi, I was going through some old company tickets that I am assigned to and found a case when possibly an attacker flooded our kamailio server with invalid sip messages like this: 2019-04-27T20:14:05.533554+09:00 IPX051 /usr/local/src/git/sip-router/kamailio[1732]: ERROR: <core> [parser/msg_parser.c:714]: ERROR: parse_msg: message=<[F#016sD#026Z<8D>97<F8><B5>;<A9><E7>-<D2>(<E2><F6> v;/#021k\<CC>8<B1>λ<F4>#004M<B6><BE><EC>#035#003<94><E1>=<A0><FF><E3><AF>Kwzr<8B>A#036B<D7>#027#023cu<82>Y<D4>#037<FB><AC>S_<C4>Qg<AB><DE>F<88>I#006<8C><FA><F4>~#y3G<C7>H<80>b<BC><AD>#035<89>#002<DB><C8>#001U<9E>#007<CB><F9>nT<E5><EE><8E><F1>#0144> At that time we manually banned the IP. But it would be helpful to have this done automatically by fail2ban. So I was thinking this log should include the src IP address. I looked at the latest kamailio commit and core/parser/msg_parser.c does this log the same way so I was thinking in opening an issue for this. But maybe this should be dealt with differently. Any ideas?

4 6

Issue with Retrieving Kamailio Statistics: Missing FIFO File and Errors
by satyaprakash ch 25 Oct '24

25 Oct '24

Hi, I'm encountering an issue while trying to retrieve Kamailio statistics on the server using the command kamctl stats. Instead of the expected output, I am receiving the following errors: kamctl stats: -e \E[37;31mERROR: Error opening Kamailio's FIFO /run/kamailio/kamailio_rpc.fifo -e \E[37;31mERROR: Make sure you have loaded the jsonrpcs module and set FIFO transport parameters Additionally, the file /run/kamailio/kamailio_rpc.fifo does not exist. Could you please provide suggestions ?

3 4

Handeling error returned by rtpengine_manage(...)
by Wolter Eldering 23 Oct '24

23 Oct '24

Hello, I have a question about the rtpengine module and how to use it via KEMI/python. Will processing of the native script stop when rtpengine_manage returns with an error? Or will it continue and will the SDP be forwarded unchanged? #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); } else { rtpengine_manage("replace-origin replace-session-connection"); } #!else Regards, Wolter

3 4

Kamailio not deleteing rtp sessions
by Tihomir Culjaga 22 Oct '24

22 Oct '24

Hello, I'm using Kamailio + RTPEngine. Everything works well except one thing. I noticed rtp sessions are not being deleted after call termination. Instead, they remain lingering on RTPEngine waiting for a timeout. kamailio --version 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, 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: 55e232 compiled on 14:13:04 Aug 25 2021 with gcc 8.4.1 rtpengine --version Version: 9.5.1.1-1.el8 rtpengine-ctl list totals tells me every session went to timeout :) --- snip --- Total statistics (does not include current running sessions): Uptime of rtpengine :914 seconds Total managed sessions :51 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :51 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :0 Total forced terminated sessions :0 --- snip --- I'm using rtpengine_manage in two routes "route[RELAY]" and onreply_route[MANAGE_REPLY] . Do I need to add a specific rtpengine_delete somewhere ? route[RELAY] { ... $xavp(r=>$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 (!has_totag()) { xlog("L_INFO", "NEXIOS: proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "####################### FROM ?? to ?? $ru ########################\n"); } } # we engage RTPEngine if (has_body("application/sdp")) { xlog("L_INFO", "NEXIOS: # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[MANAGE_REPLY] { ... $xavp(r=>$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 (has_totag()) { xlog("L_INFO", "onreply_route proto = [$proto], ru = [$ru]"); if ($proto =~ "ws" && !($ru =~ "transport=ws")) { xlog("L_INFO", "onreply_route ####################### FROM WS to any ########################\n"); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP direction=external direction=internal"; } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { xlog("L_INFO", "onreply_route ####################### FROM any to WS ########################\n"); setbflag(FLB_TOWEBRTC); $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=active SDES-off ICE=force RTP/SAVPF direction=internal direction=external"; } else if (($proto =~ "udp") && $ru =~ "transport=udp" && $fs =~ "MY_IP2_ADDR") { xlog("L_INFO", "onreply_route ####################### FROM ?? to ?? $ru ########################\n"); } } if (has_body("application/sdp")) { xlog("L_INFO", "onreply_route # $xavp(r=>$T_branch_idx) #\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); } } Any ideas are appreciated. Regards, Tihomir.

4 6

parallel forking: how to fix 183 g711 then 200 ok g722?
by Benoit Panizzon 21 Oct '24

21 Oct '24

Hi Gang Parallel forking situation. A => B A => C A invites with g711,g722 which is forked to B and C B reports ringing by sending 183 + SDP g711 C report 180 ringing. C establishes connection 200 OK + SDP g722 https://www.rfc-editor.org/rfc/rfc3261.html#page-78 o If the initial offer is in an INVITE, the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that INVITE. For this specification, that is only the final 2xx response to that INVITE. That same exact answer MAY also be placed in any provisional responses sent prior to the answer. The UAC MUST treat the first session description it receives as the answer, and MUST ignore any session descriptions in subsequent responses to the initial INVITE. Now we run into the situation in which A received the 183 g711 and being RFC compliant, dropped the 2nd SDP from the 200 OK. => No audio! Has anyone encountered this situation and found a nice solution? Like generating an UPDATE or RE-INVITE towards A to change the codec in an RFC compliant way after the call is established? Or is the only solution to remove all codecs except the required g711a one when more than one contact are registered on an AOR? Mit freundlichen Grüssen -Benoît Panizzon- -- 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 5

Statically Linking Multiple Modules Compile Error
by Steve Bezalel 21 Oct '24

21 Oct '24

Hello, I am trying to compile Kamailio with all its modules statically linked. How should I go about this? I followed through the installation documentation but could not find any information. I tried passing all the modules I want to statically link in the "static_modules" argument of `make` but it seems that definitions of modules clash as such. ``` make static_modules="siputils rr" ``` ``` /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_version'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_flags'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:69: multiple definition of `module_interface_ver'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:78: first defined here /usr/bin/ld: modules/rr/rr_mod.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:166: multiple definition of `exports'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:241: first defined here /usr/bin/ld: modules/rr/rr_mod.o: in function `mod_register': /playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/rr/rr_mod.c:926: multiple definition of `mod_register'; modules/siputils/siputils.o:/playground/fuzzing-targets/sip/kamailio-dfsan/src/modules/siputils/siputils.c:817: first defined here ``` Any help would be appreciated. Thank you.

1 0

$hdr behaviour using jansson+http_connect
by Sergio Charrua 19 Oct '24

19 Oct '24

Hi all! My team and I are finalizing implementation of STIR/SHAKEN. The flow has been simplified and is something like: Call comes from SBC to Kamailio, Kamailio makes an HTTP request to a REST API, response is received by Kamailio (5.8.2) and depending on the response, it replies back to SBC with a SIP 300 Multiple Choice of a SIP 4xx error, denying the call. The INVITEs received by SBC may or may not contain the Identity header. On Kamailio side, the following code gets the required headers to build the JSON object sent to REST API: route[HANDLE_STIRSHAKEN] { if (!is_method("INVITE")) { return; } $var(requestTime) = $utimef(%s); $var(ruri) = "$ru"; #request URI $var(from) = "$hdr(From)"; #$fU From header $var(to) = "$hdr(To)"; #$tU; To header $var(contact) = "$hdr(Contact)"; #Contact header $var(callID) = "$ci"; # Call ID $var(identity) = "$hdr(Identity)"; # Identity header jansson_set("string","from" , $var(from) , "$var(post)"); jansson_set("string","to" , $var(to) , "$var(post)"); jansson_set("string","r-uri" , $var(ruri) , "$var(post)"); jansson_set("string","contact" , $var(contact) , "$var(post)"); jansson_set("string","call-id" , $var(callID) , "$var(post)"); jansson_set("string","identity" , $var(identity) , "$var(post)"); http_connect("api1", "/stsh", "application/json","$var(post)","$var(result)"); xlog("L_DEBUG", "HANDLE_STIRSHAKEN - Result = $var(result)\n"); [... do some other irrelevant stuff ...] } The usual INVITE is similar to this: INVITE sip:+33142XXXXXX@10.242.XXX.YYY:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.240.AAA.BBB:5064;branch=z9hG4bK+30f178b47ee25dd589f986ed7402b4eb3+sip+6+a64eb889 From: "+493XXXXXXX" <sip:+493XXXXXXX@10.240.AAA.BBB ;user=phone>;tag=10.240.AAA.BBB+6+ffe686f8+614d17a3 To: +33142XXXXXX <sip:+33142XXXXXX@10.242.XXX.YYY> CSeq: 1 INVITE Expires: 180 Content-Length: 234 Call-Info: <sip:10.240.AAA.BBB:5064>;method="NOTIFY;Event=telephone-event;Duration=2000" Supported: timer, 100rel Contact: <sip:+493XXXXXXX@10.240.AAA.BBB:5064;user=phone> Content-Type: application/sdp Call-ID: 0gQAAC8WAAACBAAALxYAAHRfFdrMLbnN/iMYu2NTT6kvhuhZAFl7MG1V2ozJLUcXg9dxbyQZ/PtoswtdHtcv9g--(a)10.240.AAA.BBB Min-SE: 90 Session-Expires: 90;refresher=uac X-dtmf: 5 P-Asserted-Identity: "+493XXXXXXX" <sip:+493XXXXXXX@10.240.AAA.BBB> Accept: application/sdp Accept: application/dtmf-relay Max-Forwards: 66 Allow: INVITE,CANCEL,BYE,ACK, INFO Identity: dGhpcyBpcyBzb21lIGR1bW15IGJhc2U2NCBlbmNvZGVkIElkZW50aXR5IGhlYWRlcg==.eW91IHdvdWxkIG5vdCBleHBlY3QgbWUgdG8gbGVhdmUgdGhlIHJlYWwgdmFsdWUgaGVyZSByaWdodD8=.dGhhbmtzIGZvciB5b3VyIGNvbXByZWhlbnNpb24=;info=< https://someURL.com/with/some/parameter;sc697295b.cer>;alg=ES256;ppt=shaken P-Origination-Id: 23A16F8B-F9AB-4536-9884-85C96EF3B964 P-Attestation-Indicator: C has-sdp: present [SDP stuff] (I have replaced some of the values for obvious reasons). - SBC is sending every 70 secs 2 automatically generated INVITEs to kamailio - 1st call contains the Identity header, the other call does not contain Identity header. - Both calls have different Call-IDs and different FROM and TO values, so they are different calls and can be perfectly found in logs as separated calls - Kamailio receives each call and extracts headers, as shown in code above, and sends the JSON request to HTTP REST API Here is the weird part.... every now and then, completely random, the JSON object request sent to the HTTP REST API contains an Identity value, supposedly extracted from the SIP header **but** the SIP message captured in SNGrep doesn't have **any* I*dentity header!! In fact, in the kamailio logs, where I sent the header values extracted from INVITE, the identity is empty! BUT the JSON object has the identity set with value. And the worst part is that, when this issue happens, the value of the Identity set in JSON belongs to a completely different call that was processed a few minutes before ! So I suspected about some Caching behaviour on the http_connect function and added the following lines before sending the JSON to REST API: if ( is_present_hf("Identity") ) { jansson_set("string","identity",* $var(identity),* "$var(post)"); } else jansson_set("string","identity", "", "$var(post)"); forcing the identity property of the JSON to an empty string, but this did not work either! I even added a new property which is the current timestamp making sure every request was unique, and that did not work either! The only way I could make it work was replacing the above lines using these lines: if ( is_present_hf("Identity") ) { jansson_set("string","identity", *(a)hf_value2.Identity*, "$var(post)"); } else jansson_set("string","identity", "", "$var(post)"); So, instead of using $hdr(Identity) , Kamailio extracts the Identity header using @hf_value2.Identity Question: why is this behaviour? Could this be a bug somewhere? Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 3

Re: Dispatcher ping: "binding to source address X.X.X.X:XXXX failed: Address already in use" for TCP and TLS destinations
by Mirko Brankovic 15 Oct '24

15 Oct '24

Hi, Has anyone found how to safely suppress the warning? We see it on 5.7.4 Thanks, Mirko

2 2

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users