sr-users October 2024

sr-users@lists.kamailio.org

34 participants
22 discussions

Statically Linking Multiple Modules Compile Error

by Steve Bezalel

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 month

$hdr behaviour using jansson+http_connect

by Sergio Charrua

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*

1 month

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

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

1 month, 1 week

kamailio - acc sqlite

by marek

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

1 month, 1 week

Getting the name of an already set onreply_route

by Denys Pozniak

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

1 month, 1 week

Appending Branches Post Request Processing

by Knserbrave

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?

1 month, 2 weeks

TLS module crashes with FIPS OpenSSL

by Marat Gareev

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

1 month, 2 weeks

t_append_branches() and "dead" tcp connections

by Ihor Olkhovskyi

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)

1 month, 2 weeks

kamailio sending from udp non-listening port

by David Villasmil

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

1 month, 2 weeks

relay to voip provider / sip trunk / path?

by Linux Admin

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!

1 month, 3 weeks

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sr-users October 2024