sr-users October 2024

sr-users@lists.kamailio.org

34 participants
23 discussions

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

Kamailio Developers Meeting, Nov 19-20, 2024, in Dusseldorf
by Daniel-Constantin Mierla 14 Nov '24

14 Nov '24

Hello, Kamailio SIP Server project is organizing another meeting of its developers and community members during November 19-20, 2024 (Tue-Wed), hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and contributors in order to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join, developer or user interested in helping the project. Please note we have a limited capacity of seats in the meeting room, the main policy for accepting participants being first come first server. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

2 4

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

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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

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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

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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.

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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

kamailio - acc sqlite
by marek 14 Oct '24

14 Oct '24

hi, i'm trying setup kamailio for simple CDR logging versions - kamailio 5.7, rocky 9.4 i have configured and populate sqlite with "kamdbctl create" kamailio.cfg #CDR loadmodule "db_sqlite.so" loadmodule "acc.so" modparam("acc", "cdr_enable", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 2) modparam("acc", "db_url", "sqlite:///var/log/kamailio-cdr.sqlite") modparam("acc", "time_mode", 3) but i have this error ERROR: db_sqlite [dbase.c:232]: db_sqlite_submit_query(): failed to prepare statement: table acc has no column named time_attr any ideas where can i find acc schema update for time_attr column? bash# grep -Rni time_attr /usr/share/kamailio/db_sqlite/ shows nothing Marek

2 2

Getting the name of an already set onreply_route
by Denys Pozniak 14 Oct '24

14 Oct '24

Hello! Is it possible to find out the name of the onreply_route that was set before? Something like this: t_on_reply("MANAGE_REPLY"); ... if ( t_is_set("onreply_route") ) { get_onreply_route_name(); ... } -- BR, Denys Pozniak

3 5

Appending Branches Post Request Processing
by Knserbrave 08 Oct '24

08 Oct '24

Hi Community, Is it possible to append a branch to a transaction after the call is in a ringing state? Say a call comes in and an AOR only has two contacts. Branches are created to the contacts. A third comes in why the two are in the ringing state. Can we create a branch to the third?

2 3

TLS module crashes with FIPS OpenSSL
by Marat Gareev 08 Oct '24

08 Oct '24

Hello, I encountered a problem stopping Kamailio with FIPS OpenSSL: Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3 Missing separate debuginfos, use: dnf debuginfo-install kamailio-5.7.3-4816.x86_64 (gdb) bt #0 0x00007ff7292380ac in OPENSSL_sk_pop () from /lib64/libcrypto.so.3 #1 0x00007ff72914bf5b in conf_modules_finish_int () from /lib64/libcrypto.so.3 #2 0x00007ff72914c694 in CONF_modules_unload () from /lib64/libcrypto.so.3 #3 0x00007ff7291efff9 in OPENSSL_cleanup () from /lib64/libcrypto.so.3 #4 0x00007ff72954702b in ?? () #5 0x0000000100061c08 in ?? () #6 0x00007ff7190566c8 in ?? () #7 0x00007ffccf196a20 in ?? () #8 0x000000000071da8a in futex_release (lock=0x7ff729f08b50 <syslog>) at core/mem/../mem/../futexlock.h:134 #9 0x00000000006e9448 in destroy_tls () at core/tls_hooks.c:75 #10 0x000000000041f278 in cleanup (show_status=1) at main.c:594 #11 0x0000000000420af1 in shutdown_children (sig=15, show_status=1) at main.c:721 #12 0x0000000000421717 in handle_sigs () at main.c:752 #13 0x0000000000430c88 in main_loop () at main.c:1988 #14 0x0000000000439d13 in main (argc=14, argv=0x7ffccf1973f8) at main.c:3212 (gdb) Environment: Oracle Linux Server 9.3 Kamailio 5.7.3 yum list --installed | grep ssl openssl.x86_64 10:3.0.7-24.0.3.el9_fips @tools openssl-libs.x86_64 10:3.0.7-24.0.3.el9_fips @tools openssl-pkcs11.x86_64 0.4.11-7.el9 @anaconda xmlsec1-openssl.x86_64 1.2.29-9.el9 @AppStream What can I do for further investigation? Thanks

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t_append_branches() and "dead" tcp connections
by Ihor Olkhovskyi 04 Oct '24

04 Oct '24

Hello, Recently encountered an issue with tsilo (that is using t_append_branches), but not sure found a root cause. So, I have a call, but aparently there are some "dead" TCP connections in a "location" table modparam("usrloc", "handle_lost_tcp", 0) On call initialtion, INVITE's are sent to "last seen" location, but aparently there is no answer and I assume it's in a state "trying to establish TCP connection" Obviously, there is no "100 Trying" answer on any outgoing INVITE. Than, a REGISTER comes from a contact, that is supposed to be called. Usual ts_append("location", "$tu"); is called, but in the logs I see [REGISTER] tm [t_append_branches.c:208]: t_append_branches(): failure to add branches (-1) If there is an active TCP connection, branches are added as expected. I'm getting this only in a case of trying to establish a call to a "dead" TCP connection. What am I missing here or it sounds like a bug? -- Best regards, Ihor (Igor)

2 9

kamailio sending from udp non-listening port
by David Villasmil 04 Oct '24

04 Oct '24

Hello guys, i'm seeing something weird, and i'm working if you can let me know. I have a kamailio in AWS with a private IP listening on several sockets: Listening on udp: 10.1.2.36:5070 udp: 10.1.2.36:5080 udp: 10.1.2.36:5160 advertise 4.3.2.1:5160 udp: 0.0.0.0:5066 tls: 10.1.2.36:443 advertise sip.something.com:443 tls: 10.1.2.36:444 advertise sip.something.com:444 tls: 10.1.2.36:5061 When forwarding a udp invite received on 10.1.2.36:5080 to a public ip provider say on 8.8.8.8:5060, i'm forcing the outgoing socket with force_socket via 10.1.2.36:5160. But the outgoing invite does NOT use 5160, it uses some random port... Anybody knows why this might be? my problem is, that call goes to freeswitch... call is setup properly and connects fine. But 15 minutes later the end provider sends back a reINVITE, which freeswitch then sends TO THE RANDOM PORT kamailio used to send the INVITE... but by this time kamailio doesn't seem to even see the packet... help is greatly appreciated! David Regards, David Villasmil email: david.villasmil.work(a)gmail.com

4 9

relay to voip provider / sip trunk / path?
by Linux Admin 03 Oct '24

03 Oct '24

Hi. I am using https://github.com/havfo/WEBRTC-to-SIP to have a web sipcaller. it uses kamailio + rtpengine + web sipcaller (jssip) right now it is working using internal kamailo account and can call to another sip. But I need to use websipcaller to call to voip provider. I tried to register on my voip provider, and use just outbound websocket proxy, but it not working giving 401 the config I am using is https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Please help adopt it to relay anything to external sip provider. I don't know what I need, relay, sip trunk or path module. kamailio configs are very hard for me to understand. So I need it just as websocket proxy to my voip provider, that doesn't support websocket. thanks!

2 2

app_lua parse multiple files
by iliusha.md＠gmail.com 03 Oct '24

03 Oct '24

I have a Lua script which is split into multiple files, like: /etc/kamailio/routing/ ├── main.lua ├── logger.lua └── db.lua On restart, Kamailio fails to start with an error: main.lua:2: module 'logger' not found:\n\tno field package.preload If I set multiple files in the modparam parameter: modparam("app_lua", "load", "/etc/kamailio/routing/main.lua") modparam("app_lua", "load", "/etc/kamailio/routing/logger.lua") I'm getting the same error. Is it possible to load multiple files, or is this a limit to a single file currently?

4 3

Heads-up: Clean-up of legacy modules in app_python3
by Richard Chan 01 Oct '24

01 Oct '24

HI kamailio users, There is a WIP PR on master (and targeting 6.0) to clean-up legacy modules on app_python3. https://github.com/kamailio/kamailio/pull/3986/commits/a9eb4b35b67b58893772… If you rely on pre-KEMI features (Core / Router / Logger / Ranks) can you chime in here. This work is also in preparation to enable the Kamailio module to use free-threading builds of Python. Regards Richard

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sr-users October 2024