- sr-users - kamailio.org

Branch Route trigger for Withindialog calls?
by Benoit Panizzon 21 Mar '24

21 Mar '24

Hi List I am just wondering... When I am sending the initial INVITE to a customer CPE, this goes throug the whole location lookup and through a branch route in which I make some last adjustments to the headers, like removing header the customer shall not get (like P-Asserted-Identity which would reveal the caller identity on a callerid restricted call). On a Re-Invite (session timer refresh) the call is being routed by loose_route() and immediately sent to RELAY. uac_replace and uac_restore seem to work fine for stuff like To and From Header. But how do I prevent unwanted header to disclose information to the customer which should ne be disclosed? Do I need to re-arm the branch trigger also for within dialog calls? 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 ______________________________________________________

2 1

loose_route false when using topos? At which point in the call flow, does topos restore the Route header?
by Benoit Panizzon 21 Mar '24

21 Mar '24

Hi https://lists.kamailio.org/pipermail/sr-users/2020-July/109801.html I have the exact same issue. When the B side is starting a new transaction (UPDATE to refresh the session in my case) without topos enabled, that transaction contains one or multiple route header and a to_tag. Therefore loose_route() is true and the call is more or less sent to route(RELAY) immediately without further checks. If topos is enabled, all record-route header are removed and only one Via sent to the CPE. So when the CPE replies, there is no Route header. Therefore loose_route returns false. This makes me wonder, at which stage is topos restoring the route headers? Shouldn't that happen before loose_route is evaluated? 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 4

No 408 generated by tm module on fr_timer when redis crashes
by Patrick Karton 21 Mar '24

21 Mar '24

1 0

Wrong documentation for module 'Userblacklist'
by Björn Klasen 20 Mar '24

20 Mar '24

Hi everybody, I'm a little bit confused about the documentation of the Userblacklist module as I think it has to be Userblocklist since version 5.x. Also some of the commands and some column names in the database have changed. I can remember that the documentation was right at some point, but the it has been changed back to Userblacklist. I wonder if nobody noticed that because the old commands do not work. BR, Björn -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Gerhard-Fröhler-Straße 12 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de https://www.tng.de Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Gunnar Peter, Sven Schade, Carsten Tolkmit, Bernd Sontheimer Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Sie unter https://www.tng.de/datenschutz/ abrufen. ______________________________________________________________________

2 1

Bug in htable in 5.8.0
by Brooks Bridges 18 Mar '24

18 Mar '24

We have some scripts that are setting values in an htable for various things, one of which is to disable options replies to take a system “out of service”. We discovered today upon deploying 5.8.0 that the “htable.seti” command appears to be broken. [root@ip-10-52-42-102 ~]# kamcmd htable.seti system_settings option_pings_off 1 error: 500 - Not enough parameters (htable name, key name and value) however if we do htable.sets it works fine (although not an integer so it’s breaking our shutdown scripts). [root@ip-10-52-42-102 ~]# kamcmd htable.sets system_settings option_pings_off 1 Ok. Key set to new value. [root@ip-10-52-42-102 ~]# I’ve looked through the commit history for htable and haven’t found anything that really stands out as a possible issue, so can the gurus please take a look? Thanks! Brooks Bridges Sr. Developer [https://files.skyetel.com/logo.png] Direct: (888) 444‑1111 Office: (561) 453‑4085 Email: bbridges(a)skyetel.com 902 Clint Moore Road Suite 206 Boca Raton, FL 33487 www.skyetel.com 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 4

Need guidance on Teams interop with Kamailio
by Shah Hussain Khattak 17 Mar '24

17 Mar '24

Hello Experts, I'm not an expert in Kamailio, but thanks to some excellent online resources, I managed to set up a basic interconnection between Teams and Kamailio. So far, outbound calls, as well as blind and attended call transfers, are working smoothly. However, I'm facing a challenge with call disconnection for outbound calls. Specifically, when an outbound call is made and the called party hangs up, the BYE message isn't being relayed properly towards the Teams side. In my routing configurations, I remove the Record-Route header received from Teams and add a new Record-Route header with Kamailio's IP before forwarding it to my IP PBX (FreeSwitch). For the BYE message to be processed correctly, Microsoft Teams requires the original contact in the Request URI (RURI) and the Route header to match the initial Record-Route included in the INVITE. Currently, my BYE message appears as follows: BYE sip:api-du-a-auea.pstnhub.microsoft.com:443;x-i=5e96d703-b1e7-449c-aebf-a87bfa628176;x-c=f61b674bc597513ab8d888dbe7c560fb/d/8/9a6c508a9cbd44ebb892b62c33ba9b67 SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKeab5.b657e3455b6d11ed6975c69882632a27.0 Via: SIP/2.0/UDP x.x.x.x;received=x.x.x.x;rport=5060;branch=z9hG4bKK3HKQBUvX42ya Max-Forwards: 69 From: <sip:+61403225xxx@kamsbc.xyzdomain.com:5061;user=phone>;tag=j5gptBSXXrXDc To: "XYZ Pty Ltd" <sip:+61291455xxx@sip.pstnhub.microsoft.com:5061;user=phone>;tag=3a5f0c9e32bc4eb1bffed29dc954118f Call-ID: f61b674bc597513ab8d888dbe7c560fb CSeq: 80555743 BYE User-Agent: Unknown/v1.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Reason: MVTSLocal;cause=10;text="BYE received" Content-Length: 0 Kamailio conducts a lookup for api-du-a-auea.pstnhub.microsoft.com but ends up sending the BYE message to an incorrect IP. Below is a snippet of my routing block configurations: ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); #check who is the sender route(INITCHECK); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # 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(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # update $du to set the destination address for proxying #$du = "sip:" + $rd + ":9"; route(RELAY); exit; } route[INITCHECK] { if(from_uri =~ ".*microsoft.com") { setflag(FROM_TEAMS); $du = "sip:" + "MY_PBX_IP" + ":" + "5060"; //added by SHK remove_hf("Record-Route"); // Remove existing Record-Route header append_hf("Record-Route: <sip:KAM_IP:5060;transport=udp>\r\n"); route(HANDLE_RTP_FROM_TEAMS); } if(from_uri =~ ".*" + "MY_PBX_IP") { setflag(FROM_PBX); $du="sip:sip.pstnhub.microsoft.com;transport=tls"; route(HANDLE_RTP_FROM_PBX); } } #Manage RTP & transcoding comming from Teams to PBX route[HANDLE_RTP_FROM_TEAMS] { t_on_reply("PBX_REPLY_TO_TEAMS"); if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection ICE=remove media-address=HOST_IP"); // record_route(); t_relay_to_udp("MY_PBX_IP","5060"); } else { rtpengine_manage(); } } #Manage RTP & transcoding comming from PBX to Teams route[HANDLE_RTP_FROM_PBX] { t_on_reply("TEAMS_REPLY_TO_PBX"); if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all ICE=force codec-transcode=PCMA replace-origin replace-session-connection media-address=ADVERTISE_IP"); $rd = "sip.pstnhub.microsoft.com"; $td = "kamsbc.xyzdomain.com"; $fd = "kamsbc.xyzdomain.com"; // record_route(); #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; t_relay(); } else { rtpengine_manage(); } } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); 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()) return; #Teams reINVITEs if(isflagset(FROM_TEAMS)) { loose_route(); t_relay(); exit; } if(isflagset(FROM_PBX)) { #Set TLS SNI (server name & server id) $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; loose_route(); t_relay(); exit; } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails #set coresponding cert on transactions if($fd == "kamsbc.xyzdomain.com") { $xavp(tls=>server_name) = "kamsbc.xyzdomain.com"; $xavp(tls=>server_id) = "kamsbc.xyzdomain.com"; } } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); } #PBX On Reply onreply_route[PBX_REPLY_TO_TEAMS] { if (has_body("application/sdp")) { rtpengine_manage("SRTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=ADVERTISE_IP"); } else { rtpengine_manage(); } //added by SHK if (status == "200" && method == "INVITE") { remove_hf("Contact"); # Remove existing Contact header append_hf("Contact: <sip:kamsbc.xyzdomain.com:5061;transport=tls>\r\n"); } if (status=="200") { remove_hf("ALLOW:"); append_hf("ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY \r\n","CONTENT-TYPE"); } } #From Teams On Reply onreply_route[TEAMS_REPLY_TO_PBX] { if (has_body("application/sdp")) { rtpengine_manage("RTP codec-mask=all codec-transcode=PCMA replace-origin replace-session-connection media-address=HOST_IP"); } else { rtpengine_manage(); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { if (t_is_canceled()) exit; } event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:kamsbc.xyzdomain.com:SBC_PORT;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } ####### Routing Logic End ######## Given these details, I suspect the issue might be related to how I'm handling the Record-Route headers or potentially a misconfiguration in directing the BYE message. I would greatly appreciate any insights, suggestions, or guidance on how to correctly relay the BYE message back to Teams, or further refine my existing configuration. Thank you in advance for your help and support! Regards, Shah Hussain

2 2

topology hiding with active dispatcher ip
by marek 15 Mar '24

15 Mar '24

hi, i have kamailio acting as SBC i need hide topology like this ds_select_dst(DSP_GRP_TRUNK, "6"); $tu = $(tu{re.subst,/PRIVATE_IP/IP_OF_CURRENT_SELECTED_DISPATCHER/g}); what is best way for IP extraction from $du? thanks Marek

2 2

SCSF always challenge
by Anthony Blandin 14 Mar '24

14 Mar '24

Hello, I have an issue with SCSF which challenge again after a successful first registration. I have an error after the second register saying that no matching auth vector found. I attach the logs and pcap file. I don't see where is the issue. Could you help me? Thanks a lot. Anthony

1 0

append_hf not working
by Calvin E. 13 Mar '24

13 Mar '24

I'm trying to add something simple like the following: append_hf("X-testheader: True\r\n", "From"); However, I don't see my X-testheader in a packet capture. Are there any common pitfalls that would prevent append_hf from working as expected?

8 9

TLS requests get relayed over unencrypted TCP
by James Browne 13 Mar '24

13 Mar '24

Hi all I've set up a kamailio server on a public IP address to serve public clients. - The clients REGISTER over a mix of TCP and TLS, so kamailio has listeners on the same IP address: port 5060 for UDP/TCP and port 5061 for TLS. - Some clients REGISTER from the same public IP address, behind the same NAT. - Sometimes, the client-side NAT can use the same client-side IP address and port for two concurrent connections - one TCP and one TLS - which is normal because the server-side port is different (they're different 5-tuples). - Now when a request comes to kamailio, to be routed to a client, the the t_relay_to_tls() function sometimes can relay the request over a TCP connection instead of a TLS connection. I've enabled tcp_connection_match=1. This is helpful, but it doesn't completely resolve the problem. It helps when both connections (TCP and TLS) are open, because the TLS connection will always be used for TLS requests. However, when there's no TLS connection open, kamailio will erroneously use the TCP connection. Note that this setup depends on TCP/TLS connections being made from the client to the server, not the other way around. I've made a cut-down version of this setup to demonstrate the problem. - Client: 11.15.32.1 - Server: 11.15.32.11 - version: kamailio 5.9.0-dev0 (x86_64/linux) ################################################################# #!KAMAILIO enable_tls=yes listen=tls:11.15.32.11:5061 listen=tcp:11.15.32.11:5060 listen=udp:11.15.32.11:5060 loadmodule "tm" loadmodule "tls" loadmodule "pv" loadmodule "rr" fork=yes log_facility=LOG_LOCAL0 log_stderror=no children=2 debug=3 tcp_connection_match=1 modparam("tls", "low_mem_threshold1", -1) modparam("tls", "low_mem_threshold2", -1) modparam("tls", "certificate", "/usr/local/etc/kamailio/certs/chain") modparam("tls", "private_key", "/usr/local/etc/kamailio/certs/key") request_route { loose_route(); $fs="tls:11.15.32.11:5061"; t_relay_to_tls(); } ################################################################# This is a BYE message to be sent from internally in the network. ########################### BYE sip:user3@11.15.32.1:33333;transport=tls SIP/2.0 Via: SIP/2.0/UDP 0.0.0.0:11111;rport;branch=z9hG4bK-d8754z-cc2c63344f5218d3-1 Route: <sip:11.15.32.11:5060;transport=tcp;r2=on;lr;ftag=ZUrDU2m4Z9Zam> Route: <sip:11.15.32.11:5061;transport=tls;r2=on;lr;ftag=ZUrDU2m4Z9Zam> f: <sip:user1@server>;tag=ZUrDU2m4Z9Zam t: <sip:user3@server>;tag=gK0ea97395 i:sdfg8we790t874ujk CSeq: 102 BYE Content-Length: 0 ########################### When I send that to kamailio over UDP, kamailio relays it over a TCP connection, as seen with socat here (socat runs on the machine with address 11.15.32.1). ########################### $ socat - TCP:11.15.32.11:5060,bind=:33333 BYE sip:user3@11.15.32.1:33333;transport=tls SIP/2.0 Via: SIP/2.0/TLS 11.15.32.11:5061;branch=z9hG4bKd2bf.137932ff9a937cde941fd1790db040c9.0 Via: SIP/2.0/UDP 0.0.0.0:11111;received=11.15.32.11;rport=11111;branch=z9hG4bK-d8754z-cc2c63344f5218d3-1 f: <sip:user1@server>;tag=ZUrDU2m4Z9Zam t: <sip:user3@server>;tag=gK0ea97395 i:sdfg8we790t874ujk CSeq: 102 BYE Content-Length: 0 ########################### Note that - the RURI has transport=tls in it, - kamailio is forcing the TLS socket, - kamailio is using the t_relay_to_tls function, - the Route header field tells that TLS should be used, and - the Via header field shows that kamailio tried to send this over TLS, but - the BYE was sent over unencrypted TCP. If I use this socat command instead, then I can receive the BYE, showing that kamailio is able to use TLS if there's a connection available. - socat - OPENSSL:11.15.32.11:5061,bind=:33333,verify=no I can't find any setting to control this, so it looks to me like a kamailio bug. At that, I see this as a security flaw, because the SIP traffic should always be encrypted but now is leaked. Can anyone help? James

1 0

Lua classes in events function
by Michal Hajek 12 Mar '24

12 Mar '24

Hi, I use KEMI/Lua. I have simple lua class: myclass = {} myclass.event = function() KSR.xlog.xinfo("Event working...\n") end In kamailio.conf have: modparam("xhttp", "event_callback", "myclass.event") In debug log have: Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_api.c:549]: sr_lua_reload_script(): No need to reload [/etc/kamailio/donjon.lua] is version 0 Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_api.c:733]: app_lua_run_ex(): executing Lua function: [[myclass.event]] Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_api.c:735]: app_lua_run_ex(): lua top index is: 40 Mar 12 13:21:30 kamailio DEBUG: app_lua [app_lua_mod.c:164]: sr_kemi_config_engine_lua(): execution of route type 513 with name [myclass.event] returned 1 The problem is that the event function in the myclasses class is not called. Is there any other way or syntax to do this? Of course, if I define a function without a class, it works. Thanks Michal

1 0

5.8 acc module failed_filter error
by Juha Heinanen 12 Mar '24

12 Mar '24

My SIP proxy didn't start with Kamailio 5.8 using the same config that starts OK with 5.7. The error is: 2024-03-11T06:56:06.457107+02:00 lohi /usr/bin/sip-proxy[2358954]: DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 407 2024-03-11T06:56:06.457265+02:00 lohi /usr/bin/sip-proxy[2358954]: ERROR: acc [acc_mod.c:369]: parse_failed_filter(): response code is not followed by comma or end of string 2024-03-11T06:56:06.457436+02:00 lohi /usr/bin/sip-proxy[2358954]: ERROR: acc [acc_mod.c:439]: mod_init(): failed to parse failed_filter param The relevant params are: modparam("acc", "failed_transaction_flag", 8) modparam("acc", "failed_filter", "407") I didn't find any difference in acc_mod.c source code between 5.7 and 5.8 and parse_failed_filter function looked OK. Then I added a debug statement in the beginning of parse_failed_filter function: LM_DBG("parsing failed_filter %s\n", s); and got to syslog: cc_mod.c:341]: parse_failed_filter(): parsing failed_filter 407rmissions|pua|rtpengine)$ That explains the parse_failed_filter error, but where does that bogus param value come from? Before acc params I have: modparam("auth_db|dialplan|domain|htable|lcr|msilo|mtree|permissions|pua|rtpengine", "db_url", "mysql://xxxx/sip_proxy") modparam("registrar|nathelper", "received_avp", "$avp(received_uri)") So it from there, but why? Is there a bug somewhere or is my SIP proxy running out of memory or something? Before the above error messages, there are no other error messages. -- Juha

2 5

Kamailio Yocto package
by Olle E. Johansson 11 Mar '24

11 Mar '24

Hello Kamailians! Before I start going down this path - anyone that has written a Kamailio Yocto layer and want to share either layer or experience or both? For those that doesn’t know: Yocto is a toolkit for building your own Linux distros for embedded systems. /O

1 0

SIP Redirect Server
by Antonio 11 Mar '24

11 Mar '24

Hi everyone I have a small question: i want to set up a basic SIP Redirect Server so I install Kamailio v5.3 and comment default request route and set simple request_route { rewritehostport("1.2.3.4"); sl_send_reply("302", "Moved Temporarily"); } but for some reason, Kamailio does not care about the port and sends a reply to 5060 [image: image.png] i also tried to delete rewritehostport but it only deletes Contact: from the reply but reply itself is still sent to 5060 port [image: image.png] -- *Antony* satskiy.a(a)gmail.com +380669197533 +48727830247

4 4

5.8 compiler warning
by Juha Heinanen 10 Mar '24

10 Mar '24

Kamailio 5.8 build generates on my Debian 12 the following warning: core/mem/q_malloc.c:996:14: warning: 'qm_strnstr' defined but not used [-Wunused-function] 996 | static void *qm_strnstr(const void *b1, int l1, const void *b2, int l2) | ^~~~~~~~~~ Perhaps the definition of the function should be #ifdef'ed by DBG_QM_MALLOC. -- Juha

1 0

SSL key logger for Diffie-Hellman cipher
by Calvin E. 08 Mar '24

08 Mar '24

I've been tasked to use LD_PRELOAD to log SSL keys for TLS connections using a Diffie-Hellman cipher. The first attempt did not work, so I wanted to sanity check whether Kamailio's TLS support is built in such a way that would defeat LD_PRELOAD. The instructions from the vendor are to update /etc/init.d/kamailio like this: env SSLKEYLOG_UDP='127.0.0.1:1234' LD_PRELOAD="/usr/local/src/voipmonitor-git/tools/ssl_keylogger/sslkeylog.so /usr/lib/x86_64-linux-gnu/libssl.so.3" \ start-stop-daemon --start --quiet --pidfile $PIDFILE \ --exec $DAEMON -- $OPTIONS || log_failure_msg " already running" Is there anything special in Kamailio (5.7.3 on Debian 12) that would prevent this from working? Not necessarily something to defeat a keylogger, but maybe the way tls.so gets loaded? The only discrepancy I've noticed is the vendor docs refer to libssl.so.3 not libssl.so.1, but the vendor said that should be OK. I'd love to hear from someone already using VoIPmonitor with Diffie-Hellman ciphers and Kamailio.

6 16

from TCP to UDP and Kamailio doing it wrong
by Alberto Diez 08 Mar '24

08 Mar '24

Hi kamailio community, I have an issue with a Kamailio 5.7. It's listening both in TCP and UDP. In my scenario requests arrive from devices on TCP, but I want to forward to the next hops on UDP. I am avoiding using any type of DNS resolution; since I am always forwarding to predefined next hops I am using the dispatcher module (defined with the IP addresses and transport=udp) or I wrote config files using t_relay_to_udp or t_relay_to with a udp: followed by IP address. I never set up FQDNS only IP addresses and in all of them I explicitly mention UDP. In all of these scenarios I have tried Kamailio insists in trying to use TCP with the next hop and failing because the next_hop is only UDP. I guess because the message arrived using TCP Kamilio does that but I find the behavior very confusing. I nailed down that in my situation its the tm module function prepare_new_uac (in file src/modules/tm/t_fwd.c line 119) being the one that missbehaves. The documentation of the function says literraly : "* t->uac[branch].request.dst will be filled if next_hop !=0 with the result * of the DNS resolution (next_hop, fproto and fsocket). * If next_hop is 0 all the dst members except the send_flags are read-only * (send_flags it's updated) and are supposed to be pre-filled." I found out that even when next_hop is 0 the function changes the t->uac[branch].request.dst proto, socket etc. its there that the kamailio takes the wrong decision, until that function is called within add_auc, the destination proto or the fproto etc is always 1 (UDP) which is what I am trying to force from the config file or the dispatcher definition (I tried both ways). But after calling that function then it comes as a 2 (TCP) The problematic function is a monster of 500 lines and I would like to avoid having to understand it. Since I think the scenario is not so unusual I just want to ask if maybe I am missing something that I should do to avoid the Kamailio to select TCP and in order to have the tm module respecting my preference for UDP (either with dispatcher module and the transport param or with t_relay_to or t_relay_to_udp I don't care the way). Any hints are welcome. Best regards alberto

5 9

direct media between UACs
by Sergio Charrua 08 Mar '24

08 Mar '24

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine either. Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3 and Kamailio has IP .5 The Asterisk servers are used only for testing, nothing serious. However, Kamailio is setup to use RTJson requesting routes to a Routing Server on the same network. And it works fine. Both Asterisk servers have the same dialplan, which only Answers the call and plays MOH on both ends so that RTP audio streams both ways. When making a call on Asterisk Server #1 via command line to go directly to Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/123(a)10.20.0.3 application MusicOnHold() ) the Asterisk #2 receives the call, answers and plays MOH too and I can see RTP streams coming from both ends correctly. However, if I use Kamailio to proxy the call generated from Asterisk #1 to Asterisk #2, using similar command line instruction (CLI> channel originate SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received on Kamailio who then sends it to Asterisk #2, who answers the call and plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is never received, however audio from Asterisk #1 is received by Asterisk #2, which configures a typical One Way Audio issue due to NAT. This is where it gets strange, because there is no NAT, SDP on INVITE and SIP 200 messages seem OK, as far as I understand it. Also, Asterisk servers have SIP configuration with directmedia enabled and NAT disabled to make sure that media is direct. But I have also tried with directmedia disabled and NAT enabled and get identical results. I am most probably missing some tiny detail, but I have no clue.... and I bet it is simple and stupid.... Could another pair of eyes help me with this? What is wrong? Do I really need RTPEngine even when the network has no NAT? I am sure it would work that way, but it doesn't make sense... Here are some screenshots: Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without Kamailio in between: Invite from Asterisk #1 to Asterisk #2 with direct media between both ends: https://drive.google.com/file/d/1eLjT3nr_Rc-UBaf4QhIgZ95bjETOVvxo/view?usp=… Replies from Asterisk #2 to Asterisk #1 with direct media between both ends: https://drive.google.com/file/d/11lLcB-V8rWGSrVqWiit-q9WX2FfqB6BZ/view?usp=… Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to Asterisk #2, with one way audio Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details: https://drive.google.com/file/d/1Cp9xrGcwNmQ9Ks36N_oD1Dj7lxfu-tbH/view?usp=… Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk #1 identical to above: https://drive.google.com/file/d/1mi3FCNjM3luXfENEp-0088XLgcyfXRK6/view?usp=… Reply from Asterisk #2 to Kamailio with SDP details: https://drive.google.com/file/d/1TpMGe2tvpX_5SIpSbm2b3Zro9EuYwcO-/view?usp=… Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2 identical to above: https://drive.google.com/file/d/12jq5APfFwVVPc0vJ3RkcXyN1hBE51fnQ/view?usp=… As we can see, SDP details seem OK, but if I check call flow on Asterisk #1, I can only find 1 RTP channel with audio coming from Asterisk #2 https://drive.google.com/file/d/1iEfvkylZVbthHM5kxkurWh-GAYCYLytl/view?usp=… and the same on Asterisk #2 : https://drive.google.com/file/d/12rvf9Lrwp-MNZvGCwlXEEBsBRmfcinph/view?usp=… My Kamailio.cfg is as follows: #!KAMAILIO # # config file for SIPProxy # - load balancing of VoIP calls # - no TPC listening # # Kamailio (OpenSER) SIP Server v3.2 # - web: http://www.kamailio.org # - git: http://sip-router.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # #!define WITH_DEBUG ###!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_SRC_ALLOWED 8 #!define FLT_DST_INTERNAL_IP 9 #!define FLT_SRC_INTERNAL_IP 10 #!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g" #!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g" #!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g" #!ifndef HTTP_ASYNC_CLIENT_WORKERS #!define HTTP_ASYNC_CLIENT_WORKERS 8 #!endif /* add API http timeout */ #!define HTTP_API_TIMEOUT 5000 #!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route" /* DMQ SIP message sharing */ #!define DMQ_PORT 5062 #!define DMQ_LISTEN "sip:10.20.0.2:5062" #!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062" #!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062" memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " fork=yes children=8 /* comment the next line to enable TCP - all trunks are UDP only */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=udp:10.20.0.2:5062 advertised_address="10.20.0.5"; sip_warning=no; use_dns_failover = on; ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.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 "acc.so" loadmodule "usrloc.so" loadmodule "nathelper.so" #loadmodule "rtimer.so" #loadmodule "sqlops.so" # --- CPS Limiter # --- end of CPS Limiter loadmodule "ipops.so" loadmodule "textopsx.so" loadmodule "sdpops.so" loadmodule "http_async_client.so" loadmodule "rtjson.so" loadmodule "jansson.so" loadmodule "dmq.so" loadmodule "dmq_usrloc.so" loadmodule "htable.so" loadmodule "dialog.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "log_level_name", "exec") #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) modparam("tm", "remap_503_500", 0) # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) # ----- nathelper params ----- modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") #modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") #modparam("rtimer", "exec", "timer=cdr;route=CDRS") #modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:kamailiorw@10.19.139.113:3306/kamailio") #modparam("dmq", "server_socket", DMQ_SERVER_SOCKET ) modparam("dmq", "server_address", DMQ_SERVER_ADDRESS ) modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 60) modparam("dmq_usrloc", "enable", 1) # -- CPS Limiter modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;") modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "enable_dmq", 1) # ----- http_async_client params ----- modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) modparam("http_async_client", "connection_timeout", 2000) ####### Routing Logic ######## # main request routing logic route { if (is_method("KDMQ") && $Rp == 5062) { dmq_handle_message(); } xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","start|\n"); xlog("L_INFO","===================================================\n"); xlog("L_INFO","New SIP message $rm with call-ID $ci \n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO"," received $pr request $rm $ou\n"); xlog("L_INFO"," source $si:$sp\n"); xlog("L_INFO"," from $fu\n"); xlog("L_INFO"," to $tu\n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO","---------------------------------------------------\n"); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } if ($fU=="ping") { sl_send_reply("200","OK"); exit; } # extract original source ip / port from X-forwarded-For header route(HANDLE_X_FORWARDED_FOR); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm"); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO", "ROUTE - Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ t_on_failure("MANAGE_FAILURE"); xlog("L_INFO", "ROUTE - Recording Route"); record_route(); if (is_method("INVITE") && is_request()) { if (has_body("application/sdp")) { xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n"); t_on_reply("ON_REPLY"); } } } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","ROUTE - Address Incomplete"); exit; } route(TOCARRIER); xlog("L_INFO", " ********** Route END *************"); } # extract original source ip / port from X-forwarded-For header route[HANDLE_X_FORWARDED_FOR] { if (is_present_hf("X-Forwarded-For")) { $var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:}); $var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:}); } else { $var(source_ip) = $si; $var(source_port) = $sp; } $var(to_number) = $rU; } route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port) from_number $fU to_number $ru"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = HTTP_API_TIMEOUT; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)"); jansson_set("string","from_port",$var(source_port), "$var(http_routing_query)"); jansson_set("string","from_number",$fU, "$var(http_routing_query)"); jansson_set("string","to_number",$var(to_number) , "$var(http_routing_query)"); xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: $var(http_routing_query)\n"); $http_req(body) = $var(http_routing_query); t_newtran(); http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); } # Relay request using the API (response) route[RELAY_API_RESPONSE] { if ($http_ok==1 && $http_rs==200) { xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n"); if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) { xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)"); rtjson_init_routes("$var(rtjson)"); rtjson_push_routes(); # relay the message t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); return; } } send_reply(500, "API Not Available - http response = $http_rs $http_ok"); exit; } onreply_route[ON_REPLY] { xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs"); # on reply if (t_check_status("183|180|200")) { xlog("L_INFO", "ON_REPLY - Fixing Contacts"); # subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a"); //subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f"); } if (has_body("application/sdp")){ if (sdp_remove_line_by_prefix("a=maxptime")){ xlog("L_INFO", "ON_REPLY - remove maxptime "); msg_apply_changes(); } else{ xlog("L_INFO", "ON_REPLY - did not removed maxptime "); } } if (t_check_status("408")) { xlog("L_INFO", "ROUTE - Handling 408 Timeout\n"); } } route[TOCARRIER]{ #using rtjson, unsomment following line route(RELAY_API); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO", "REQINIT - Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "REQINIT - 483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { xlog("L_INFO", "NATDETECT - Entering"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO", "NATDETECT - Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()){ xlog("L_INFO", "NATDETECT - Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS); return; } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO", "WITHINDLG - Entering"); if (!has_totag()) return; if (is_present_hf("Route") && $hdrc(Route)==1) { if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f")) { xlog("L_INFO", "WITHINDLG - Removing the route to self"); remove_hf("Route"); } } # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE|CANCEL")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO", "WITHINDLG - Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { #Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } if(is_method("BYE")) xlog("L_INFO", "WITHINDLG - BYE message from $rU"); route(RELAY); exit; } if ( is_method("ACK|BYE|INVITE|UPDATE") ) { 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. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG"); exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO", "WITHINDLG - Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # Routing to foreign domains ---> NOT USED route[SIPOUT] { xlog("L_INFO", "WITHINDLG - Entering SIPOUT"); if (uri==myself){ xlog("L_INFO", "WITHINDLG - URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO", "WITHINDLG - Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO", " ******** RELAY *******"); xlog("L_INFO", "RELAY - $si $su $ru"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO", "RELAY - branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } xlog("L_INFO", "RELAY - checking method"); if (is_method("INVITE|SUBSCRIBE|UPDATE")) { xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE"); if(!t_is_set("onreply_route")) { xlog("L_INFO", "RELAY - onreply_route NOT SET!"); t_on_reply("ON_REPLY"); # MANAGE_REPLY"); } } if (is_method("INVITE")) { xlog("L_INFO", "RELAY - is INVITE"); t_on_failure("FAILED_RELAY"); if(!t_is_set("failure_route")) { xlog("L_INFO", "RELAY - failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO", "RELAY - t_relay returns FALSE"); route("MANAGE_FAILURE"); #sl_reply_error(); } xlog("L_INFO", "RELAY - exiting"); exit; } failure_route[FAILED_RELAY] { xlog("L_INFO", "FAILED_RELAY - Entering"); if (t_check_status("[4-5][0-9][0-9]")){ xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!"); if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); } } } route[NATMANAGE] { xlog("L_INFO", "NATMANAGE - Entering"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } if (is_request()) { xlog("L_INFO", "NATMANAGE - is_request - $rm from $si"); if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO", "NATMANAGE - adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si"); if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO", "NATMANAGE - Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Manage failure routing cases route[MANAGE_FAILURE] { xlog("L_INFO", "MANAGE_FAILURE - Entering "); route(NATMANAGE); xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled"); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!"); t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!"); t_reply("404","Not found"); exit; } #!endif if (t_check_status("503")){ xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination available"); t_reply("503", "Destination not available"); exit; } if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 "); rtjson_update_branch(); route(NATMANAGE); } Any help would be greatly appreciated. Thanks in advance. Sérgio Charrua

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Kamailio v5.8.0 Released - new major version is out
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Kamailio v5.8.0 is out – it comes with 4 new modules and a large set of improvements touching again more than 50 existing modules. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-8-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.8.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. Enjoy Kamailio v5.8.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Using uac_replace_to when forwarding call to another endpoint
by Marrold 07 Mar '24

07 Mar '24

Hi all, I am using uac_replace_to to replace the TO header on requests. I understand it can only be called once per message and the recommendation is to store the updated value in a pvar and apply it in the branch route. However in my scenario I wish to first try one endpoint and then if it gets a 4XX response forward it to another UAC, and update the TO accordingly. Currently trying to call uac_replace_to twice shows an error and corrupts the TO header. Is there a work around for this? Thanks again Matthew

3 3

Roadmap to next major Kamailio release series v5.8.x
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Hello, discussed a bit during the online Kamailio devel meeting, it is time to set the milestones towards the next major Kamailio release series v5.8.x. If no other suggestions that suit more developers, I would propose to freeze by end of this month or early February, then test for about 4 weeks as usual and release by end of February or during March. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

3 12

rtimer intermittent failures
by Marrold 06 Mar '24

06 Mar '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine, with a Python Kemi based config. I am seeing some intermittent failures when accessing an instance variable within a function called by rtimer. I'm using the rtimer module with the following parameters: modparam("rtimer", "timer", "name=hello;interval=5;mode=0;") modparam("rtimer", "exec", "timer=hello;route=ksr_route_hello") I initialise the kamailio class, and instance variable like this: class kamailio: def __init__(self): self.hello = "hi" Within the class I have the following route function: def ksr_route_hello(self, msg, evname): KSR.info("Running ksr_route_hello\n") KSR.info(f"Hello? {self.hello}\n") Then in the logs I see it sometimes works, and sometimes fails: 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Running ksr_route_hello 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Hello? hi 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Running ksr_route_hello 9(15) ERROR: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_route_hello(rtimer): Unhandled exception in the Python code: TypeError: 'NoneType' object cannot be interpreted as an integer The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 1007, in ksr_route_hello KSR.info("Running ksr_route_hello\n") SystemError: <built-in function info> returned a result with an exception set Does anyone know where I'm going wrong? Thanks Matthew

2 2

Question for RFC junkies / kamailio modifying message bodies
by Valentin Christoph 06 Mar '24

06 Mar '24

Hi all, it might be a minor issue, or it might not be an issue at all, but I stumbled over following statements in RFC 3261: Chapter 16.6 Request Forwarding (of a proxy) 1. Copy request The proxy starts with a copy of the received request. The copy MUST initially contain all of the header fields from the received request. Fields not detailed in the processing described below MUST NOT be removed. The copy SHOULD maintain the ordering of the header fields as in the received request. The proxy MUST NOT reorder field values with a common field name (See Section 7.3.1<https://www.rfc-editor.org/rfc/rfc3261.html#section-7.3.1>). The proxy MUST NOT add to, modify, or remove the message body. An actual implementation need not perform a copy; the primary requirement is that the processing for each next hop begin with the same request. Chapter 16.7 Response Processing (at a proxy) 9. Forward response After performing the processing described in steps "Aggregate Authorization Header Field Values" through "Record-Route", the proxy MAY perform any feature specific manipulations on the selected response. The proxy MUST NOT add to, modify, or remove the message body. Unless otherwise specified, the proxy MUST NOT remove any header field values other than the Via header field value discussed in Section 16.7<https://www.rfc-editor.org/rfc/rfc3261.html#section-16.7> Item 3. In particular, the proxy MUST NOT remove any "received" parameter On the other hand, everywhere in the Internet (starting with stackoverflow), you can read it is OK, if a SIP proxy modifies a body, and also kamailio allows it. Might be a question for the coffee break. All the best Christoph

3 2

Kamailio closing tcp connection on downstream responses
by Kjartan S 06 Mar '24

06 Mar '24

Trying to setup Kamailio with MS Teams Direct Routing. I am familiar with the integration steps required by Microsoft (using contact header as FQDN, certified CA for certificate) aswell as the famous guide; https://skalatan.de/en/blog/kamailio-sbc-teams + https://learn.microsoft.com/en-us/microsoftteams/direct-routing-protocols-s… Sending any SIP requests (Invite, ACK, BYE) toward MS is working fine (with provisional response received in Kamailio) - so there is obviously a working connection between Kamailio <-> MS Direct Routing environment. HOWEVER, once Kamailio is forwarding SIP responses to MS (provisional responses/200 OK received from UAS), then Kamailio is closing the TCP connection. Please see attached logs below. Any idea why Kamailio is closing the connection when trying to send provisional responses to MS? - Using t_realy on original invite from MS - tcp_reuse_port=yes, tcp_rd_buf_size = 16384, tcp_connection_lifetime=3605, children=8, socket_workers=4 - No override rules in firewall - OS; debian:11.7 - Notice that TLS connection is made on port :26627 - however, MS sends port :5061 in VIA header (on first Invite). This seems to me to be a conflict in ports. https://www.kamailio.org/wiki/cookbooks/5.4.x/core#reply_route "There is no network route that can be enforced for a SIP reply - it is sent based on Via header, according to SIP RFC3261" ----- Handshake complete for 52.114.76.76:26627 19(43) DEBUG: tls [tls_domain.c:818]: sr_ssl_ctx_info_callback(): SSL handshake done 19(43) DEBUG: tls [tls_server.c:473]: tls_accept(): TLS accept successful 19(43) DEBUG: tls [tls_server.c:476]: tls_accept(): tls_accept: new connection from 52.114.76.76:26627 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 19(43) DEBUG: tls [tls_server.c:480]: tls_accept(): tls_accept: local socket: 10.129.0.91:5061 19(43) DEBUG: tls [tls_server.c:420]: tls_dump_cert_info(): tls_accept: client certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 19(43) DEBUG: tls [tls_server.c:424]: tls_dump_cert_info(): tls_accept: client certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 04 ----- Kamailio trying to forward SIP response "183" to MS (response received from UAS). Kamailio finds active connection by ID 13 / 0x7f4ecad0be08 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 13 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2627]: tcpconn_send_put(): tcp connection found (0x7f4ecad0be08), acquiring fd 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2637]: tcpconn_send_put(): c=0x7f4ecad0be08, n=16 23(47) DEBUG: <core> [core/tcp_main.c:3982]: handle_ser_child(): read response= 7f4ecad0be08, 2, fd -1 from 20 (44) 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2665]: tcpconn_send_put(): after receive_fd: c= 0x7f4ecad0be08 n=8 fd=12 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2842]: tcpconn_do_send(): sending... 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2878]: tcpconn_do_send(): after real write: c= 0x7f4ecad0be08 n=1191 fd=12 ----- Kamailio immediately closes the connection due to EOF. 19(43) DEBUG: <core> [core/tcp_read.c:280]: tcp_read_data(): EOF on connection 0x7f4ecad0be08 (state: 0, flags: 4018) - FD 10, bytes 0, rd-flags 10000 ([52.114.76.76]:26627 -> [52.114.76.76]:5061)19(43) DEBUG: <core> [core/tcp_read.c:1544]: tcp_read_req(): EOF 19(43) DEBUG: <core> [core/io_wait.h:597]: io_watch_del(): DBG: io_watch_del (0x555b3cf20260, 10, -1, 0x10) fd_no=2 called 19(43) DEBUG: <core> [core/tcp_read.c:1927]: handle_io(): removing from list 0x7f4ecad0be08 id 13 fd 10, state 2, flags 4018, main fd 57, refcnt 2 ([52.114.76.76]:26627 -> [52.114.76.76]:5061) 19(43) DEBUG: <core> [core/tcp_read.c:1702]: release_tcpconn(): releasing con 0x7f4ecad0be08, state -1, fd=10, id=13 ([52.114.76.76]:26627 -> [52.114.76.76]:5061)

1 0

RADIUS ACC - Interim Updates
by duarterocha91＠gmail.com 06 Mar '24

06 Mar '24

Hello all, I can't seem to find a way to make interim accounting requests for RADIUS. Can you help me with this? I can't find any native way for doing this. I've thought of getting a custom timer for each dialog and run a event route similar do dialog:start or dialog:end. Is that possible? Cheers, Duarte Rocha

2 2

Add XML to INVITE
by David Villasmil 06 Mar '24

06 Mar '24

Hello guys, I'm trying to add an xml with a boundary to an outgoing INVITE. But if i do: set_body_multipart("$rb", "application/sdp", "delimiter"); msg_apply_changes(); append_body_part("$var(something)", "application/pidf+xml"); msg_apply_changes(); but kamailio adds the SDP, but NOT the second $var(something). If i switch them and add firs the var, and then the SDP. kamailio adds the contents of $var(something) but NOT the SDP.... point is it doesn't add the second append Thanks!!! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

dispatcher - sip options per gw
by marek 05 Mar '24

05 Mar '24

hi, i have kamailio gw (5.6.x) for voxbone. dispatcher module voxbone has separated inbound/outboud and prohibit sip options to "inbound" IP #traffic from voxbone 20 sip:81.201.82.45:5060;transport=udp 1 0 duid=abc;socket=udp:x.x.x.x:5060 #traffic TO voxbone 30 sip:81.201.89.110:5060;transport=udp 0 0 duid=abc;socket=udp:x.x.x.x:5060 is it possible send sip options per GW with dispatcher module? i tried flag "1 (bit at index 0 - 1 <<0) - inactive destination" https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#idm1059 Marek

2 2

direct media between UACs
by Sergio Charrua 05 Mar '24

05 Mar '24

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine neither. Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3 and Kamailio has IP .5 The Asterisk servers are used only for testing, nothing serious. However, Kamailio is setup to use RTJson requesting routes to a Routing Server on the same network. And it works fine. Both Asterisk servers have the same dialplan, which only Answers the call and plays MOH on both ends so that RTP audio streams both ways. When making a call on Asterisk Server #1 via command line to go directly to Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/ 123(a)10.20.0.*3* application MusicOnHold() ) the Asterisk #2 receives the call, answers and plays MOH too and I can see RTP streams coming from both ends correctly. However, if I use Kamailio to proxy the call generated from Asterisk #1 to Asterisk #2, using similar command line instruction (CLI> channel originate SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received on Kamailio who then sends it to Asterisk #2, who answers the call and plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is never received, however audio from Asterisk #1 is received by Asterisk #2, which configures a typical One Way Audio issue due to NAT. This is where it gets strange, because there is no NAT, SDP on INVITE and SIP 200 messages seem OK, as far as I understand it. Also, Asterisk servers have SIP configuration with directmedia enabled and NAT disabled to make sure that media is direct. But I have also tried with directmedia disabled and NAT enabled and get identical results. I am most probably missing some tiny detail, but I have no clue.... and I bet it is simple and stupid.... Could another pair of eyes help me with this? What is wrong? Do I really need RTPEngine even when the network has no NAT? I am sure it would work that way, but it doesn't make sense... Here are some screenshots: Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without Kamailio in between: Invite from Asterisk #1 to Asterisk #2 with direct media between both ends: [image: image.png] Replies from Asterisk #2 to Asterisk #1 with direct media between both ends: [image: image.png] Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to Asterisk #2, with one way audio Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details: [image: image.png] Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk #1 identical to above: [image: image.png] Reply from Asterisk #2 to Kamailio with SDP details: [image: image.png] Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2 identical to above: [image: image.png] As we can see, SDP details seem OK, but if I check call flow on Asterisk #1, I can only find 1 RTP channel with audio coming from Asterisk #2 [image: image.png] and the same on Asterisk #2 : [image: image.png] My Kamailio.cfg : #!KAMAILIO # # config file for SIPProxy # - load balancing of VoIP calls # - no TPC listening # # Kamailio (OpenSER) SIP Server v3.2 # - web: http://www.kamailio.org # - git: http://sip-router.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # #!define WITH_DEBUG ###!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_SRC_ALLOWED 8 #!define FLT_DST_INTERNAL_IP 9 #!define FLT_SRC_INTERNAL_IP 10 #!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g" #!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g" #!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g" #!ifndef HTTP_ASYNC_CLIENT_WORKERS #!define HTTP_ASYNC_CLIENT_WORKERS 8 #!endif /* add API http timeout */ #!define HTTP_API_TIMEOUT 5000 #!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route" /* DMQ SIP message sharing */ #!define DMQ_PORT 5062 #!define DMQ_LISTEN "sip:10.20.0.2:5062" #!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062" #!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062" memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " fork=yes children=8 /* comment the next line to enable TCP - all trunks are UDP only */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=udp:10.20.0.2:5062 advertised_address="10.20.0.5"; sip_warning=no; use_dns_failover = on; ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.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 "acc.so" loadmodule "usrloc.so" loadmodule "nathelper.so" #loadmodule "rtimer.so" #loadmodule "sqlops.so" # --- CPS Limiter # --- end of CPS Limiter loadmodule "ipops.so" loadmodule "textopsx.so" loadmodule "sdpops.so" loadmodule "http_async_client.so" loadmodule "rtjson.so" loadmodule "jansson.so" loadmodule "dmq.so" loadmodule "dmq_usrloc.so" loadmodule "htable.so" loadmodule "dialog.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "log_level_name", "exec") #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- 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) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) modparam("tm", "remap_503_500", 0) # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) # ----- nathelper params ----- modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") #modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") #modparam("rtimer", "exec", "timer=cdr;route=CDRS") #modparam("sqlops", "sqlcon", "ca=>mysql:// kamailio:kamailiorw@10.19.139.113:3306/kamailio") #modparam("dmq", "server_socket", DMQ_SERVER_SOCKET ) modparam("dmq", "server_address", DMQ_SERVER_ADDRESS ) modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 60) modparam("dmq_usrloc", "enable", 1) # -- CPS Limiter modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;") modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "enable_dmq", 1) # ----- http_async_client params ----- modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) modparam("http_async_client", "connection_timeout", 2000) ####### Routing Logic ######## # main request routing logic route { if (is_method("KDMQ") && $Rp == 5062) { dmq_handle_message(); } xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","start|\n"); xlog("L_INFO","===================================================\n"); xlog("L_INFO","New SIP message $rm with call-ID $ci \n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO"," received $pr request $rm $ou\n"); xlog("L_INFO"," source $si:$sp\n"); xlog("L_INFO"," from $fu\n"); xlog("L_INFO"," to $tu\n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO","---------------------------------------------------\n"); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } if ($fU=="ping") { sl_send_reply("200","OK"); exit; } # extract original source ip / port from X-forwarded-For header route(HANDLE_X_FORWARDED_FOR); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm"); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO", "ROUTE - Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ t_on_failure("MANAGE_FAILURE"); xlog("L_INFO", "ROUTE - Recording Route"); record_route(); if (is_method("INVITE") && is_request()) { if (has_body("application/sdp")) { xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n"); t_on_reply("ON_REPLY"); } } } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","ROUTE - Address Incomplete"); exit; } route(TOCARRIER); xlog("L_INFO", " ********** Route END *************"); } # extract original source ip / port from X-forwarded-For header route[HANDLE_X_FORWARDED_FOR] { if (is_present_hf("X-Forwarded-For")) { $var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:}); $var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:}); } else { $var(source_ip) = $si; $var(source_port) = $sp; } $var(to_number) = $rU; } route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port) from_number $fU to_number $ru"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = HTTP_API_TIMEOUT; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)"); jansson_set("string","from_port",$var(source_port), "$var(http_routing_query)"); jansson_set("string","from_number",$fU, "$var(http_routing_query)"); jansson_set("string","to_number",$var(to_number) , "$var(http_routing_query)"); xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: $var(http_routing_query)\n"); $http_req(body) = $var(http_routing_query); t_newtran(); http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); } # Relay request using the API (response) route[RELAY_API_RESPONSE] { if ($http_ok==1 && $http_rs==200) { xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n"); if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) { xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)"); rtjson_init_routes("$var(rtjson)"); rtjson_push_routes(); # relay the message t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); return; } } send_reply(500, "API Not Available - http response = $http_rs $http_ok"); exit; } onreply_route[ON_REPLY] { xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs"); # on reply if (t_check_status("183|180|200")) { xlog("L_INFO", "ON_REPLY - Fixing Contacts"); # subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a"); //subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f"); } if (has_body("application/sdp")){ if (sdp_remove_line_by_prefix("a=maxptime")){ xlog("L_INFO", "ON_REPLY - remove maxptime "); msg_apply_changes(); } else{ xlog("L_INFO", "ON_REPLY - did not removed maxptime "); } } if (t_check_status("408")) { xlog("L_INFO", "ROUTE - Handling 408 Timeout\n"); } } route[TOCARRIER]{ #using rtjson, unsomment following line route(RELAY_API); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO", "REQINIT - Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "REQINIT - 483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { xlog("L_INFO", "NATDETECT - Entering"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO", "NATDETECT - Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()){ xlog("L_INFO", "NATDETECT - Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS); return; } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO", "WITHINDLG - Entering"); if (!has_totag()) return; if (is_present_hf("Route") && $hdrc(Route)==1) { if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f")) { xlog("L_INFO", "WITHINDLG - Removing the route to self"); remove_hf("Route"); } } # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE|CANCEL")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO", "WITHINDLG - Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { #Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } if(is_method("BYE")) xlog("L_INFO", "WITHINDLG - BYE message from $rU"); route(RELAY); exit; } if ( is_method("ACK|BYE|INVITE|UPDATE") ) { 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. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG"); exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO", "WITHINDLG - Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # Routing to foreign domains ---> NOT USED route[SIPOUT] { xlog("L_INFO", "WITHINDLG - Entering SIPOUT"); if (uri==myself){ xlog("L_INFO", "WITHINDLG - URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO", "WITHINDLG - Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO", " ******** RELAY *******"); xlog("L_INFO", "RELAY - $si $su $ru"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO", "RELAY - branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } xlog("L_INFO", "RELAY - checking method"); if (is_method("INVITE|SUBSCRIBE|UPDATE")) { xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE"); if(!t_is_set("onreply_route")) { xlog("L_INFO", "RELAY - onreply_route NOT SET!"); t_on_reply("ON_REPLY"); # MANAGE_REPLY"); } } if (is_method("INVITE")) { xlog("L_INFO", "RELAY - is INVITE"); t_on_failure("FAILED_RELAY"); if(!t_is_set("failure_route")) { xlog("L_INFO", "RELAY - failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO", "RELAY - t_relay returns FALSE"); route("MANAGE_FAILURE"); #sl_reply_error(); } xlog("L_INFO", "RELAY - exiting"); exit; } failure_route[FAILED_RELAY] { xlog("L_INFO", "FAILED_RELAY - Entering"); if (t_check_status("[4-5][0-9][0-9]")){ xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!"); if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); } } } route[NATMANAGE] { xlog("L_INFO", "NATMANAGE - Entering"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } if (is_request()) { xlog("L_INFO", "NATMANAGE - is_request - $rm from $si"); if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO", "NATMANAGE - adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si"); if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO", "NATMANAGE - Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Manage failure routing cases route[MANAGE_FAILURE] { xlog("L_INFO", "MANAGE_FAILURE - Entering "); route(NATMANAGE); xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled"); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!"); t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!"); t_reply("404","Not found"); exit; } #!endif if (t_check_status("503")){ xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination available"); t_reply("503", "Destination not available"); exit; } if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 "); rtjson_update_branch(); route(NATMANAGE); } Any help would be greatly appreciated. Thanks in advance. *Sérgio *

1 0

Question syntax check / plausi check kamailio.cfg
by Valentin Christoph 04 Mar '24

04 Mar '24

Dear all, Is there any other possibility to (automatically) check the plausibility/syntax of an existing kamailio.cfg, besides kamailio -c -f kamailio.cfg ? Thanks, Christoph

3 2

DMQ module failure
by Sergio Charrua 01 Mar '24

01 Mar '24

Hi all! I'm testing the DMQ module, but having difficulties setting it up. [...] My kamailio.cfg is as follows : [...] listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 [...] modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS modparam("dmq", "notification_address", "sip:10.20.0.4:5062 ")#DMQ_NOTIFICATION_ADDRESS modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) modparam("dmq_usrloc", "enable", 1) If I check syntax (using -c parameter) all is ok. Once I start Kamailio, the following error appears on system log: INFO: dmq [dmq.c:164]: make_socket_str_from_uri(): unknown transport protocol - fall back to udp ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is listening on ERROR: <core> [core/sr_module.c:1030]: init_mod(): Error while initializing module dmq (/usr/local/lib64/kamailio/modules/dmq.so) I have used UDP as well as SIP as protocols for Server Address and Notification Address, used port 5060 and 5062 (and others too), even added a new listen=udp:0.0.0.0:5062, as well as listen=tcp:0.0.0.0:5062 but the result is always the same. Using Kamailio 5.7.4 on RHEL 9 with SELINUX set to enforcing (if relevant). No firewall active. local IP address is 10.20.0.1 second kamailio node IP address is 10.20.0.4 Any ideas? At first sight, it seems to be all set as per documentation. I even found other configurations online, and apart from the IP address, all is similar. Thanks in advance, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

2 2

Answer OPTIONS from an Asterisk
by Ludovic ELUECQUE 28 Feb '24

28 Feb '24

Hello, I've been using Kamailio to replace a Telcobridge ProSBC for several days now, but I've been stuck for a whole day on a simple concept My Kamaillio is connected to an Asterisk to which it redirects requests. However, this Asterisk sends OPTIONS requests in order to ping the phone. On the Kamailio side, I don't know how to respond to these requests. I've done tons of research but I can't find anything similar, maybe there's something very obvious I'm missing.... If I t_relay the message it never receives a reply via the phone's public ip/port. Thanks in advance if anyone could help me with this. Please apologize if it's a dumb thing. SIP OPTIONS: 2024/02/28 23:28:25.181663 100.66.138.5:5060 -> 100.66.138.7:5060 OPTIONS sip:110@100.66.138.7:5060 SIP/2.0 Via: SIP/2.0/UDP 100.66.138.5:5060 ;rport;branch=z9hG4bKPje213d53f-fd1d-47f2-b0f8-fd3a8bf5188c From: <sip:110@100.66.138.5>;tag=46e51407-77b0-47f1-af73-cd8bec95b179 To: <sip:110@100.66.138.7> Contact: <sip:110@100.66.138.5:5060> Call-ID: 74401738-d874-4c5c-a4b5-b74514361043 CSeq: 50224 OPTIONS Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 ============= kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_RTPENGINE # define WITH_ANTIFLOOD #!define WITH_WEBSOCKETS #!define WITH_TLS #!define WITH_NAT ... #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### DEFINE VARIABLE ######## #!substdef "!MY_IP_ADDR!X.X.X.X!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_PORT!4444!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 # FLB_ - per branch flags #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:X.X.X.X:5060 listen=udp:100.66.138.7:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!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 = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.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 "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "rtpengine.so" #loadmodule "rtpproxy.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" # loadmodule "nathelper.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE #!else #!loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) #rtp modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "force_send_interface", "100.66.138.7") #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # 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) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* 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=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* 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", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "load_backends", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!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 #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- #modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- #!modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- 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@sip.super.fr") #modparam("nathelper", "sipping_method", "OPTIONS") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 40) 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) modparam("debugger", "log_level_name", "exec") #!endif ######## websocket ############ #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR tcp_accept_no_cl=yes # tcp_rd_buf_size=65536 #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif ######## SECURITY ####### server_signature=no sip_warning=0 user_agent_header="User-Agent: SIP Server" ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { route(REQINIT); route(NATDETECT); route(REGISTER); route(INVITE); route(ACK); route(BYE); return; } # Wrapper for relaying requests #Check initial detail route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent|pplsip") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { 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("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT 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); } #!endif return; } route[REGISTER] { if (is_method("REGISTER")) { t_on_reply("1"); force_send_socket(100.66.138.7:5060); t_relay_to_udp("100.66.138.5", "5060"); exit(); } } route[INVITE] { if (( is_method("REFER")) || is_method("INVITE") ||is_method("NOTIFY")) { record_route(); if (src_ip == "100.66.138.5") { if(lookup("location") && !is_method("OPTIONS")) { xlog("L_NOTICE", "user enregistrax"); force_send_socket(X.X.X.X:5060); rtpengine_manage("replace-origin replace-session-connection media-address=X.X.X.X direction=priv direction=pub"); t_relay(); } else { sl_send_reply("403","Not registred"); } } else { force_send_socket(100.66.138.7:5060); rtpengine_manage("replace-origin replace-session-connection media-address=100.66.138.7 direction=pub direction=priv"); t_relay_to_udp("100.66.138.5", "5060"); } } } route[ACK] { if (is_method("ACK")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } route[BYE] { if (is_method("BYE|CANCEL")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } onreply_route { if (status=~ "180|183") { rtpengine_manage("replace-origin replace-session-connection"); } if (status=~ "200") { if(has_body("application/sdp")) { rtpengine_manage("replace-origin replace-session-connection"); } } } onreply_route[1] { if (status=~ "2[0-9][0-9]"){ save("location"); xlog("L_NOTICE", "enregistrement"); } } ====================

2 1

best approach to setup SIP Session sharing between servers
by Sergio Charrua 28 Feb '24

28 Feb '24

Hi all! I am having some difficulties/doubts on what would be the best approach to let multiple Kamailio servers share their SIP sessions between each other. The goal is to have HA on Kamailio cluster such that if a call is received and initiated on Kamailio #1, and during any moment of the call (before or while established) if the server #1 goes down or Kamailio stops working for any reason, the call can be processed by Kamailio server #n available in the cluster. I do not want to mess with Virtual IPs or DNS, that is not the point, but instead need to get a proper way to share SIP sessions between Kamailio servers such that any server could continue the call without issue. I have checked the Dialog and DMQ module, but I am not sure if that is the way to go.... Could any one share a solution for this? Greatly appreciated. Cheers, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

8 11

Clarification on pv_www_authenticate
by James Lipski 28 Feb '24

28 Feb '24

Hello, I would like get further information on the "pv_www_authenticate" function from auth module. For the response code "-1", which is listed as "generic error", what exactly is covered when this function returns this? Is it just everything else that isn't listed reason (invalid password, nonce expired, no credentials, etc...). On my production system, I'm seeing this as a response code on some registration requests, and just want to know what is causing this (enabling full verbose logging is a bit difficult on my production system, and want to try to replicate it on my test environment, however just need clarification on what this covers). Thank you.

2 1

Topos methods_nocontact parameter
by Benoit Panizzon 27 Feb '24

27 Feb '24

Hi I'm having another go at topos trying to solve an issue of sip messages getting to big for certain clients to process. Unfortunately some methods (ACK/PRACK/BYE probably CANCEL) do not contain a contact header, therefore the topology is still revealed (and potentially a lot of via header inserted increasing message size). Is it safe to set methods_nocontact to an empty string to force creation of a Contact header for every SIP method? Is there another way to completely hide topology aka keeping header small? 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 7

How to keep SIP messages small, possibly below fragmentation limit?
by Benoît Panizzon 26 Feb '24

26 Feb '24

Hi In the process of getting rid of our commercial B2Bua SBC and handling all clients directly via Kamailio... We have come across some (unfortunately many we have in use) CPE which have an issue with fragmented packets or which simply do not allocate enough memory to parse or create messages of a certain size. Their SIP stack just crashes with errors like 'message to large for send buffer' and similar. This affects messages with containing as little as 5 via and 4 record-route header and a couple of codec in the SDP. Is there a recipe for trying to keep packets as small as possible? I would like to avoid using the topos module as I issues with this module completely failing in certain situations. -- 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 ______________________________________________________

2 2

Development is open for future release of v5.9.x series
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 was created, therefore the master branch is open for adding new features, to be part of future release series v5.9.x (or whatever version is decided for next series). Any bug fix committed to master that applies to 5.8.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.8 or 5.7. Expect that v5.8.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.9.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2024 or the beginning of 2025. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Branch 5.8 has been created
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 has been created, to be used for releasing v5.8.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.8 cd kamailio-5.8 git checkout -b 5.8 origin/5.8 Pushing commits in this branch: git push origin 5.8:5.8 Note that 5.8 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.8 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.8 will be out, respectively Kamailio v5.8.0. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Building Debian packages for further modules
by Christian Berger 22 Feb '24

22 Feb '24

Hi, for our workflow we build our own Debian packages for kamailio. We are currently trying and failing to create a package for the kafka modules. What we have tried so far: Adding the "Build-Depends:" entries for librdkafka-dev in debian/control Adding a "Package" in debian/control Adding kafka to the PACKAGE_GROUPS in debian/rules So far it seems to build the module and a package for it, but the package only contains metadata, but no .so file. We are currently running out of ideas. What else can we do? Best wishes Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 2

Kemi questions
by Stefan-Cristian Mititelu 21 Feb '24

21 Feb '24

Hello, When writing kemi cfg (python in my case), is there any way to get rid of standard cfg shared variables and use kemi cfg variables instead? E.g. Instead of $shv(blabla) (in standard config) can I use some global python var called 'blabla'? Is it seen and shared by all kamailio processes? Or I still need to use KSR.pv.get/set("$shv(blabla)") ? Thank you, Stefan

2 3

Freeswitch and Kamailio Integration.
by jayapal＠ring4.com 21 Feb '24

21 Feb '24

Greetings, I am new to Kamailio and seeking assistance. I have installed FreeSWITCH version 1.10.11 and Kamailio version 5.6.5. I have two inquiries: I aim to configure FreeSWITCH with Kamailio. Could someone kindly guide me on achieving this integration? I am curious about the storage location of Kamailio's log files. Could someone please provide information regarding this? Thank you in advance for your help and guidance.

4 4

htable write access state
by Patrick Karton 21 Feb '24

21 Feb '24

Hello are htable protected for multiple write access? can i safely update htable from one process and be sure another process will access it for write operation ? if yes is the lock on all the htable or only on the bucket where the key belongs ? Thanks.

2 1

rpm kamailio git master version
by Henning Westerholt 20 Feb '24

20 Feb '24

Hello, just a question about the https://rpm.kamailio.org/ repository, especially regarding git master builds. It seems that the git master build there was updated last autumn. It would be great if it could be updated, now as we want to test for the coming release. Thank you, Henning Westerholt -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

dialog:event unknown and kemi callback error
by Duarte Rocha 20 Feb '24

20 Feb '24

Hello all, I'm currently developing some kamailio scripts with app_jsdt by migrating a proxy of mine from Kamailio routing script to Js. I've migrated some parts of my request_route and everything seems to be working fine. I am now trying to migrate my event routes to JS and I started with dialog. dialog parameters : modparam("dialog", "event_callback", "dialogEvent") Code : function dialogEvent(event) { KSR.info("Dialog event " + event + " in JS") } After this i try to make a call and in my logs i have : "Dialog event unknown in JS" "Dialog event dialog:start in JS" Is this normal behaviour that I should ignore and only try to process "start,end,failure" or is this a signal of something wrong in my code? I also have this error in my code : "sr_kemi_cbname_lookup_idx(): index 1 is out of range". Does that mean that i have a JS function that calls a missing function in kamailio script or the opposite? How can I identify the root cause? Thanks in advance, Cheers

1 0

Variable specific to branch reply?
by Benoit Panizzon 20 Feb '24

20 Feb '24

Hi List avp stacking is starting to drive me crazy... Somehow I don't understand what is going on in this situation: I have a main branch, and add two more branches with append_branch() I use a on_branch_route trigger: branch_route[BR_TO_CPE] { [Here, some code setting $var(needbh)] if ("yes" == $var(needbh)) { xlog("L_INFO", "$cfg(route): BR_idx: $T_branch_idx BACKHAUL: YES to $nh(u) \n"); add_rr_param(";rtp=yes"); record_route(); $avp(backhaul) = "yes"; $avp(nexthop) = $nh(u); rtpengine_manage($avp(rtp_to_cpe)); } else { xlog("L_INFO", "$cfg(route): BR_idx: $T_branch_idx BACKHAUL: NO to $nh(u) \n"); record_route(); $avp(backhaul) = "yes"; $avp(nexthop) = $nh(u); } } In the Logs, I see Branches Index 0,1,2 with the desired information. So I have 3 values stacked in the avp backhaul and nexthop, right? When I get a reply, I would like to determine if backhauling is required or not. onreply_route[MANAGE_REPLY] { xlog("L_INFO", "MANAGE_REPLY: for BR_idx: $T_branch_idx\n"); while ($(avp(nexthop)[$var(i)]) != $null) { xlog("L_INFO", "MANAGE_REPLY: DEBUG Branch: $T_branch_idx VAR IDX: $var(i) NH: $(avp(nexthop)[$var(i)]) \n"); $var(i) = $var(i) + 1; } } So on EVERY reply, I would expect to get three "DEBUG" lines as I stacked 3 AVP.. Bug I get only 2! Why I am missing one of the stacked AVP? (the last one as it looks like) Or is there any other way to access a arbitrary variable specific to a branch on the reply to that branch? PS: I don't need $avp(nexthop). It's just an easy way to make sure I get the correct avp index for the branch I am interested in. 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 ______________________________________________________

1 1

mem_join_free flag in kamailio 5.6 deb
by Arsen Semenov 20 Feb '24

20 Feb '24

Hello! I have an issue with kamailio 5.6.5 after some time of flying there are errors in logs: <core> [core/mem/q_malloc.c:297]: qm_find_free(): > qm_find_free(0x7fc79c286010, 536); Free fragment not found! mem_join=1 by default, but it seems that MEM_JOIN_FREE flag is not defined in this build kamailio -I > Print out of kamailio internals > Version: kamailio 5.6.5 (x86_64/linux) > Default config: /etc/kamailio/kamailio.cfg > Default paths to modules: /usr/lib/x86_64-linux-gnu/kamailio/modules > 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, TLS_PTHREAD_MUTEX_SHARED > 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: unknown > Compiled with: gcc 10.2.1 > Compiled architecture: x86_64 > Compiled on: > OS Debian 11 bullseye However I noticed the flag exists in 5.7 installed from deb Print out of kamailio internals > Version: kamailio 5.7.4 (x86_64/linux) > Default config: /etc/kamailio/kamailio.cfg > Default paths to modules: /usr/lib/x86_64-linux-gnu/kamailio/modules > Compile flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, > *MEM_JOIN_FREE*, 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 > 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: unknown > Compiled with: gcc 10.2.1 > Compiled architecture: x86_64 > Compiled on: OS Debian 11 bullseye and now, what got me bit confused is that when I build 5.6.5 from source "make printcdefs" gives me this flag, alongside with other flags which I do not see later in "kamailio -v" output. So I am not sure whether this flag was defined during 5.6.5 deb build time or it has been somehow lost? So I need your advice. Regards, Arsen

1 0

How to query, if callid is handled by rtpengine module already?
by Benoît Panizzon 20 Feb '24

20 Feb '24

Hi I have come across this challenge: Initial invite (no to_tag) has is authenticated and has credentials. So a lookup for the username is made to determine if the customer needs backhauling, thus rtpengine needs to be engaged. On a subsequent invite (re-invite to change codec or to refresh session timer) there is a to_tag so it is being routed without sending a challenge and has no auth username. I don't use dialog on the registrar. This is another transaction. So I don't know, if I need to engage rtpengine on that DSP or not. What is the best way to find out, if rtpengine is already handling that callid? if(rtpengine_query_v("j", "$var(dummy)")) { rtpengine_manage(); } basically works, but generates a couple of ugly error messages in the log as rtpengine_query_v does not find the call. Is there a nicer way? -- 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 ______________________________________________________

2 3

TCP timeout occurs when the uac_req_send sets $ouri = Received (sip:10.11.40.XX:57646;transport=tls) but The Received connection normal
by riffs＠sina.cn 20 Feb '24

20 Feb '24

The connection registered by the client is normal, and the client can send messages and renew the registration normally, but the push message to the client using uac_req_send() from kamailio fails. Location information after the client is registered: ]#kamcmd ul.lookup location test01 Received: sip:10.11.40.XX:57646;transport=tls ]#netstat -anp | grep 57646 "tcp 0 0 10.11.55.xx:5061 10.11.40.XX:57646 ESTABLISHED 3467/kamailio" kamailio.lua KSR.pv.sets("$uac_req(ouri)", rows.received) In this way, most users can receive messages normally, but occasionally some users cannot receive messages but can renew their registration and send messages normally. Once this is the case for this user, the use of uac_req_send() will continue to fail until the client establishes a new connection. If a problem occurs, the following logs are displayed on the server: 2024-02-20T10:01:25.329595+08:00 10.11.55.XX (72819) ERROR: <core> [core/tcp_main.c:4692]: tcpconn_main_timeout(): connect 10.11.40.XX:57646 failed (timeout) The connection already exists and should be normal, why does "tcpconn_main_timeout(): connect 10.11.40.XX:57646 failed (timeout)" occur.

1 0

Dispatcher ping: "binding to source address X.X.X.X:XXXX failed: Address already in use" for TCP and TLS destinations
by Jardar Leira 17 Feb '24

17 Feb '24

We have recently upgraded from 5.6.4 to 5.7.4. No change in the KAMAILIO configuration. On all upgraded servers we are now getting this message in the logs for every destination when the dispatcher sends out OPTIONS to destinations listed in the dispatcher list: WARNING: <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address X.X.X.X:YYYY failed: Address already in use [98] The X.X.X.X being the servers IP address and YYYY being the servers listening port for tcp and/or tls. It's not consistent. For some servers it's every time OPTIONS is sent. For others it's just now and then. udp destinations have no issues. There are no other processes occupying these ports. Changing the listening port on the KAMAILIO also results in the new port reported as in use. Dispatcher OPTIONS still seems to work for both tcp and tls. Unsurprisingly using a dynamic, unprivileged port > 1023 and not the server port as the source port, as it shouldn't. The packages are sendt and a reply is registered. Inbound OPTIONS are processed as expected by KAMAILIO on the listening port. Have I missed something transitioning from 5.6 -> 5.7, or is this a bug? Relevant config: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 2) modparam("dispatcher", "ds_ping_interval", 120) modparam("dispatcher", "ds_ping_from", "sip:ping@xxxxxx.xxx") modparam("dispatcher", "ds_ping_latency_stats", 1) OS: Debian Bullseye Kamailio installed from https://deb.kamailio.org/kamailio57 bullseye Best regards, Jardar Leira

3 2

Forward subscriber username via dispatcher
by s.czech＠nowatel.com 16 Feb '24

16 Feb '24

Hi, I'm trying to forward the original username of the peer via Kamailio dispatcher to the Asterisk and I can't get it working. The scenario looks like this: Client Phone (username 1) -> Asterisk Client ---trunk(test-trunk)--> Kamailio Dispatcher/RTPEngine --trunk--> Asterisk Final Client is dialing via Asterisk Client with a simple extension: exten => _X.,1,Set(CALLERID(all)= 587555611 <587555611>) exten => _X.,n,Dial(SIP/${EXTEN}@test-trunk) Asterisk Client is registering to the Kamailio with its own credentials and Kamailio is registering in the Asterisk Final The problem is that I can't find a way to extract the original trunk username (like test-trunk) in the outgoing connection to the Asterisk Final (via Kamailio): Since i'm using the CALLERID on the Asterisk Client all I can see on the Asterisk Final is: INVITE sip:587322591@192.168.20.74 SIP/2.0 Record-Route: <sip:587322591@192.168.20.74;lr;ftag=as52b874a1;nat=yes> Via: SIP/2.0/UDP 192.168.20.74;branch=z9hG4bKc6e3.a77db1f002191ae3ee1a414b63d34a72.0 Via: SIP/2.0/UDP 192.168.21.228:5060;received=192.168.21.228;branch=z9hG4bK705ec94a;rport=5060 Max-Forwards: 69 From: "587555611" <sip:587555611@192.168.21.228>;tag=as52b874a1 To: <sip:587322591@192.168.20.74> Contact: <sip:587555611@192.168.21.228:5060;alias=192.168.21.228~5060~1> Call-ID: 433f9ff84cdb437f0d383797003c232f@192.168.21.228:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 16.29.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer P-Asserted-Identity: "587555611" <sip:587555611@192.168.21.228> Content-Type: application/sdp Content-Length: 268 There is no way I can forward the 'test-trunk' in the SIP header. I tried to add append_hf to ROUTE[relay] like append_hf("X-Trunk $au r\n"); but if the trunk is already registered then $au is null or the CALLERID value. How to extract the original username (test-trunk) on the Asterisk Final? Do you have any ideas? Thank you in advance ;)

1 0

Latest point releases seem to break early dialog UPDATEs from B-side when using topos (again)
by George Diamantopoulos 16 Feb '24

16 Feb '24

Hello all, I've noticed that there seems to be a regression with the topos module, more specifically the redis flavour, but I'm assuming the storage backend shouldn't make a difference. I have confirmed this affects both 5.6.5 and 5.7-nightly, so I'm assuming some backported commit is to blame. Kamailio 5.6.4 used previously, to the best of my memory, was not affected. Early dialog UPDATEs sent from the callee seem to be somehow malformed, since the 200 OK the UAS produces goes directly to the UAC despite having executed record_route() for the original INVITE. I can't imagine how this is possible, I thought responses always honour the Via header and record route shouldn't play a role in this case, right? Simply turning topos off restores the intended behaviour, however, so I can't help but think this is somehow related. Here's two call flows demonstrating the behaviour: * Topos on, 200 OK to UPDATE missing: https://pastebin.com/raw/J0zQeM5g * Topos off, 200 OK to UPDATE routed correctly: https://pastebin.com/raw/49yErezb I was wondering if anyone is aware of any commits that might be responsible for this. Additionally, is there an archive of debian packages with previous point releases so that I can confirm this regression with the latest versions? Thank you! Best regards, George

2 3

Reject TCP SYN
by David Villasmil 15 Feb '24

15 Feb '24

Hello all, Is there any way of actually rejecting (RST) NEW tcp connection attempts, while allowing the ongoing ones to finish naturally? I’m thinking maybe we can add this feature? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 4

uacreg - memory vs database
by Aymeric Moizard 15 Feb '24

15 Feb '24

Hi! I would like to add/remove/update entries in uacreg database. The easiest way I found is to add data using mysql and then call kamcmd uac.reg_reload. I end up reloading too often: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs I would like to optimize it and I tried to use sudo kamcmd uac.reg_add Unfortunately, the data is not persistent and seems to be only in memory. Does it means my only choice is to follow this procedure: Add a user: 1/ sudo kamcmd uac.reg_add 2/ add the equivalent mysql entry Remove a user 1/ sudo kamcmd uac.reg_remove 2/ remove the equivalent mysql entry Let me know if an alternative exists! Regards Aymeric -- Antisip - http://www.antisip.com

2 2

Kamailio as a Presence server only
by djcsalvador＠gmail.com 15 Feb '24

15 Feb '24

I all, I'm looking into Kamailio as a solution to replace our current Presence Server in our IMS Network, but feeling a bit overwhelmed by the feature size and complexity of Kamailio. I have a few questions (some might be dumb), if you guys could help me I would appreciate. 1. As I only need Kamailio as a Presence Server (with XCAP and SIP Interface), is there a way to install only the necessary components and/or modules? Our security team would probably raise some questions if we need to install unused code on the network. For what I could investigate, the modules i would need are: * Presence * Presence_XML * Xcap_server * xhttp * RLS * Pua * database * sl * tm Am I missing something? 2- As refered on point 1, our server will need a xcap and sip interface. Some presence updates will come through XCAP, others from SIP Publish. My question is, whenever a presence update is made through XCAP interface, will the kamailio server trigger the necessary SIP Notify to notify all Subscribed users ? I ask this because this was a problem with our current supplier. 3 - our client is a bit "peeky" with performance. I found here (https://www.kamailio.org/wikidocs/kemi/performance-tests/5.2.x/#results) some performance tests, but for what i can see the test primary goal was to test the response time. Is there any resource on "requests per second"? Thanks in advance for all the help David Salvador

2 4

Kamailio - handover between LTE and Wifi
by Christian Hahn 15 Feb '24

15 Feb '24

Hi all, While researching for Open Source IMS Services I’v found the presentation “What’s up with IMS & VoLTE?” at KamailioWorld 2018. There is a quote that raises a question: “Things, you shouldn’t expect (at least initially): ● Seamless handover between LTE and Wifi, as this requires some integration with the PGW of the LTE” My question is: Has the VoLTE <-> VoWiFi Handover been implemented in the meantime? BR Christian

3 3

Trying to better understand pre-made IMS configs
by Enzo Damato 10 Feb '24

10 Feb '24

Hello. I am a new Kamailio user, and I have been following the instructions for setting up a new IMS system. So far, I have a system that is working with the default example configs, but I am am trying to figure out the best way to expand on these default configurations. Currently, from my reading, the pcscf and icscf have minimal configuration/customization beyond the basic setting required to get them working, and most of the configuration related to routing, features, and other typical telecom customization should be done in the scscf. Is this mostly correct? Additionally, is there a guide to the features that are required in order to get the IMS systems working above and beyond the minimal config to get Kamailio itself working, so that I know what it's safe to modify and what should be left alone? Thank you, Enzo Damato

1 0

High CPU with DMQ on ARM64 (graviton) - Tuning values for DMQ worker_usleep
by Ben Kaufman 09 Feb '24

09 Feb '24

I encountered a behavior with DMQ using excessive CPU when running kamailio 5.8 on AWS Graviton processors (running in Docker on Amazon Linux 2023), even on an otherwise idle system. Finding the issue below pointed me towards using polling. I set the value for worker_usleep to 1000 as a test, and that seems to have resolved the CPU issue, but I have no idea of what a "good" value is for this setting, and 1000 was simply the example present in the documentation. My specific use case for DMQ is synchronizing USRLOC data for ~7,500 users, registering with expiration times of 60 - 300 seconds with 7 - 9 nodes on the DMQ bus. Any general recommendations on tuning values for this setting? https://github.com/kamailio/kamailio/issues/822 Regards, Kaufman

2 3

PRACK RFC3262
by Social Boh 09 Feb '24

09 Feb '24

Does Kamailio support the SIP PRACK method? If yes, How? Regards -- --- I'm SoCIaL, MayBe

3 3

Release_tcpconnection when using RTJSON routing with TLS
by kjartan.solli＠intility.no 08 Feb '24

08 Feb '24

Hello, I wish to use RTJSON for routing inbound calls from Kamailio to external endpoint(s), based on JSON received from my API. My rtjson setup is based on guide below, with my route(RELAY) sending Invite via t_relay(); https://wazo-platform.org/blog/kamailio-routing-with-rtjson-and-http-async-… I am including rtjson_init_routes("$var(rtjson)");, rtjson_push_routes(); & rtjson_update_branch(); in my routes. Also have defined: listen=tls:10.129.1.168:5061 advertise ${PUBLIC_IPV4}:5061 Routing from rtjson is working fine for TCP / UDP connections, however - once I try to forward an INVITE via rtjson using TLS, then the TLS connection is simply closed with Kamailio reporting "tcp_read_data(): EOF on connection". Please see logs and setup below. - tcp_reuse_port = yes, tcp_children = 8, tcp_accept_no_cl= yes & tcp_connection_lifetime=3605 - Also no success with tcp_reuse_port = no - I have HOMER setup and see here that SIP headers from the RTJSON is being used (Outbound SIP Invites is being duplicated to HOMER before connection is released) - I see no drop in traffic from my firewall As you can see from logs below, Kamailio is reusing the socket (0x7fc3ebc65060) from the exsisting connection (not new socket 0x7fc3ef70aab0). I am sending SIP Options toward the external endpoint (which is working fine), and my theory is that Kamailio is reusing the exisitng SIP OPTIONS tcp connection for the SIP INVITE and somehow close the connection when RTJSON is used (since routing without RTJSON is working fine). Any idea how to solve this? Thank you! (x.x.x.x:5061 is external endpoint) 14(43) DEBUG: rtjson [rtjson_routing.c:364]: rtjson_init_serial(): rewrite dst-uri to: [sip:x.x.x.x:5061;transport=tls] 14(43) DEBUG: rtjson [rtjson_routing.c:388]: rtjson_init_serial(): trying to set send socket to: [tls:10.129.1.168:5061] 14(43) DEBUG: <core> [core/socket_info.c:726]: grep_sock_info(): checking if host==us: 12==12 && [10.129.1.168] == [10.129.1.168] 14(43) DEBUG: <core> [core/socket_info.c:730]: grep_sock_info(): checking if port 5061 (advertise 5061) matches port 5061 ......... 14(43) DEBUG: tm [../../core/forward.h:276]: msg_send_buffer(): sending to: x.x.x.x:5061, force_socket=4, send_sock=0x7fc3ef70aab0 14(43) DEBUG: <core> [core/tcp_main.c:1741]: _tcpconn_find(): found connection by peer address (id: 9) 14(43) DEBUG: <core> [core/tcp_main.c:2627]: tcpconn_send_put(): tcp connection found (0x7fc3ebc65060), acquiring fd 14(43) DEBUG: <core> [core/tcp_main.c:2637]: tcpconn_send_put(): c=0x7fc3ebc65060, n=16 23(52) DEBUG: <core> [core/tcp_main.c:3982]: handle_ser_child(): read response= 7fc3ebc65060, 2, fd -1 from 14 (43) 14(43) DEBUG: <core> [core/tcp_main.c:2665]: tcpconn_send_put(): after receive_fd: c= 0x7fc3ebc65060 n=8 fd=20 14(43) DEBUG: <core> [core/tcp_main.c:2842]: tcpconn_do_send(): sending... 14(43) DEBUG: <core> [core/tcp_main.c:2878]: tcpconn_do_send(): after real write: c= 0x7fc3ebc65060 n=1372 fd=20 14(43) DEBUG: <core> [core/tcp_main.c:2879]: tcpconn_do_send(): buf= ......... 23(52) DEBUG: <core> [core/tcp_main.c:4671]: handle_tcpconn_ev(): sending to child, events 2001 23(52) DEBUG: <core> [core/tcp_main.c:4299]: send2child(): checking per-socket generic workers (48/19..-1828836960/21964) [tls:10.129.1.168:5061] 23(52) DEBUG: <core> [core/tcp_main.c:4326]: send2child(): WARNING: no free tcp receiver, connection passed to the least busy one (idx:0 busy:2) 23(52) DEBUG: <core> [core/tcp_main.c:4330]: send2child(): selected tcp worker idx:0 proc:19 pid:48 for activity on [tls:10.129.1.168:5061], 0x7fc3ebc65060 19(48) DEBUG: <core> [core/tcp_read.c:1782]: handle_io(): received n=8 con=0x7fc3ebc65060, fd=14 19(48) DEBUG: <core> [core/tcp_read.c:280]: tcp_read_data(): EOF on connection 0x7fc3ebc65060 (state: 0, flags: 118) - FD 14, bytes 0, rd-flags 10000 ([x.x.x.x]:5061 -> [x.x.x.x]:5061)19(48) DEBUG: <core> [core/tcp_read.c:1544]: tcp_read_req(): EOF 19(48) DEBUG: <core> [core/tcp_read.c:1702]: release_tcpconn(): releasing con 0x7fc3ebc65060, state -1, fd=14, id=9 ([x.x.x.x]:5061 -> [x.x.x.x]:5061) 19(48) DEBUG: <core> [core/tcp_read.c:1705]: release_tcpconn(): extra_data 0x7fc3ebc305b0 23(52) DEBUG: <core> [core/tcp_main.c:3744]: handle_tcp_child(): reader response= 7fc3ebc65060, -1 from 0 23(52) DEBUG: <core> [core/tcp_main.c:3668]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) 23(52) DEBUG: <core> [core/tcp_main.c:3676]: tcp_emit_closed_event(): no callback registering for handling TCP closed event 23(52) DEBUG: tls [tls_server.c:732]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fc3ebc305b0

1 0

412 issue
by satyaprakash ch 08 Feb '24

08 Feb '24

Hi, Does the below link issue is improved in updated version of Kamailio, Or kindly anyone help on this 412 issue? Link - https://github.com/kamailio/kamailio/issues/2048 Thanks.

1 0

Get the amount of open TCP connection from routing logic
by Sebastian Damm 06 Feb '24

06 Feb '24

Hello, My plan is to reject new inbound requests if there is already a certain number of open connections on one Kamailio server. I could answer with a 503 and directly close the connection forcing the client to reconnect and the loadbalancer to send traffic to a different node on the next connect. It's just a last-resort measure to prevent Kamailio from running into the configured TCP connection limit (workarounding a poorly-implemented external load balancer). Since I didn't find anything in the docs: Is there a pseudovariable containing the current number of open tcp/tls connections on this Kamailio? I can (and already do) query them via kamcmd or rpc, but can I access the variable from inside the routing script as well? Regards, Sebastian

3 6

Development for v5.8.x is frozen
by Daniel-Constantin Mierla 02 Feb '24

02 Feb '24

Hello, the formal notification that the development for the next major version 5.8.0 is now frozen. The focus has to be on testing the master branch. Also, the master branch should not get commits with new features till the branch 5.8 is created, expected to happen in 2-4 weeks, a matter of how testing goes on. Meanwhile, the commits with new features in the C code can be pushed to personal branches, new pull requests can still be done, but they will be merged after branching 5.8. Can still be done commits with documentation improvements, enhancements to related tools (e.g., kamctl, kamcmd), merging exiting pull requests at this moment, exporting missing KEMI functions and completing the functionality of the new modules added for 5.8. Once the branch 5.8 is created, new features can be pushed again to master branch as usual. From that moment, the v5.8.0 should be out very soon, time used for further testing but also preparing the release of packages. If someone is not sure if a commit brings a new feature, just make a pull request and it can be discussed there on github portal or via sr-dev mailing list. A summary of what is new in upcoming 5.8 is going to be built at: * https://www.kamailio.org/wikidocs/features/new-in-5.8.x/ Upgrade guidelines will be collected at: * https://www.kamailio.org/wikidocs/install/upgrade/5.7.x-to-5.8.0/ Everyone is more than welcome to contribute to the above wiki pages, especially to the upgrade guidelines, to help everyone else during the migration process from v5.7.x to 5.8.x. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

MS Teams Kamailio SBC tcp timeouts
by maps＠ua.pt 01 Feb '24

01 Feb '24

Hello everyone. I have setup Kamailio as an SBC for Microsoft Teams following the link provided in the documentation https://skalatan.de/en/blog/kamailio-sbc-teams and everything was working perfectly until recently. The problem I am having is that starting a few weeks back, Kamailio starting showing the errors below. These errors appear without the need to do anything, meaning without making any sort of call, even with Kamailio sitting idle. Since the problem started I looked around and found out that the issue seems to come from the fact that sometimes the SIP Options request being sent to microsoft is not being answered. I have configured everything following the guide and I did not make any changes to Kamailio since then that might have caused this issue. I am using kamailio with docker, version 5.7.2-bullseye. Thank you very much. 31(37) ERROR: <core> [core/tcp_main.c:4675]: tcpconn_main_timeout(): connect 52.114.76.76:5061 failed (timeout) 9(15) ERROR: <core> [core/tcp_main.c:1321]: tcp_do_connect(): connect 52.114.76.76:5061 failed Cannot assign requested address 9(15) ERROR: <core> [core/tcp_main.c:1324]: tcp_do_connect(): connect 52.114.76.76:5061: (99) Cannot assign requested address 9(15) ERROR: <core> [core/tcp_main.c:1456]: tcpconn_finish_connect(): 52.114.76.76:5061: tcp_do_connect for 0x7f623d0e91a8 failed 9(15) ERROR: <core> [core/tcp_main.c:2112]: tcp_send(): 52.114.76.76:5061: tcpconn_finish_connect(0x7f623d0e91a8) failed 9(15) ERROR: tm [../../core/forward.h:292]: msg_send_buffer(): tcp_send failed 9(15) ERROR: tm [uac.c:688]: send_prepared_request_impl(): Attempt to send to precreated request failed

1 0

Freezing development for 5.8.x series
by Daniel-Constantin Mierla 01 Feb '24

01 Feb '24

Hello, I propose to aim freezing the development for 5.8.x series at the end of the 1st of February 2024 (Thursday). New features that one wants to get in this release series have to be pushed to git repository or pull requests made for them. Afterwards usually follows a 4-6 weeks of testing till the first release 5.8.0. Unfreezing will happen earlier, after the first weeks of testing when the 5.8 branch will be created. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

print_dset(): no r-uri or branches
by Ali Taher 01 Feb '24

01 Feb '24

Hello all, I'm using Kamailio as SIP redirect where I'm appending the contact header then send reply back using sl_send_reply("300","Multiple Choices"). But I'm getting "print_dset(): no r-uri or branches" as warning in the log and no packet is sent back from Kamailio to the original sender. I'm using same Kamailio configuration in another setup and it's working correctly without facing this error. I'm not able to detect the issue. Can you please assist? Regards,

3 3

What happens when using exit or return in branch route in parallel branching?
by Benoit Panizzon 31 Jan '24

31 Jan '24

Hi I was wondering what exactly happens with parallel branching in this situation: branch_route[BR_T] { if (condition) { # Stuff below not required exit or return } do other stuff before relaying } Will this break, stop the processing of all branches with higher index as the one being processed? From what I observe, I fear this is the case. Could I safely use return(1) (vs return which probably is return(0) thus eq exit)? Or would the only safe option be to wrap the 'other stuff' in an else statement? 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 ______________________________________________________

2 5

install location when compiling
by David Villasmil 30 Jan '24

30 Jan '24

Hello guys, i compiled kamailio and i want it installed on debian's deb location, i.e.: /etc/kamailio /usr/lib/x86_64-linux-gnu/kamailio/modules etc.. I know if i do make prefix=/ all make prefix=/ install the config will end up on /etc/kamailio, but then the modules will be on /lib64... i don't see a way of doing this, is there? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Same number of max concurrent requests on backends
by Unai Rodriguez 30 Jan '24

30 Jan '24

Dear List, We're load balancing MRCP requests using Kamailio dispatcher module's call load distribution algorithm. We want to achieve the same maximum number of concurrent sessions on each MRCP backend but because the duration of each request varies, we’re unable to achieve this (at the moment we send the same number of requests to each backend but they all have different maximum concurrent requests). Is there a way/module that will send requests to the backend with the least number of ongoing sessions? My guess is, this module needs to connect to the backends to know their status. Any pointers appreciated. Thank you With best wishes, Unai Rodriguez

3 6

Kamailio 5.6 (and 5.7) core dumping.
by Mattis Lind 29 Jan '24

29 Jan '24

Hello Kamailio list! We have a scenario that makes use of the UAC-module to send SIP MESSAGE and then in some cases the Kamailio process core dumps after some time after processing messages. I have been able to gather a core dump which shows this backtrace appended below. We are using Kamailio 5.6 retrieved from the kamailio repository: http://deb.kamailio.org/kamailio56. We are running Kamailio in a Docker container which runs on "5.10.0-25-cloud-amd64 #1 SMP Debian 5.10.191-1 (2023-08-16) x86_64 GNU/Linux" We have previously tried to use Kamailio 5.7 but it gave the same type of crashes. We have been using the uac module a lot but it is just in this scenario we get a core dump. We have some kind of relation to a specific scenario but it can take from several seconds from the last SIP message of this scenario up to 50 minutes until the crash occurs. To me this sounds like some kind of cleanup that is not handled properly. The back trace indicates that free of shared memory could be the issue, but I don't know the code unfortunately. The last things we see in the log file is: 2024-01-23T13:18:08.828+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4789]: INFO: <script>: Incoming SIP TCP request conid 21 call-id un0rihsRLJLvP-grn6LO-A 2024-01-23T13:18:08.835+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4789]: INFO: <script>: Incoming SIP TCP request conid 21 call-id WQjhldpRJbxjZRYe7fWgbw 2024-01-23T13:18:08.860+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4795]: CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 49 2024-01-23T13:18:09.486+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: ALERT: <core> [main.c:783]: handle_sigs(): child process 4777 exited by a signal 11 2024-01-23T13:18:09.486+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: ALERT: <core> [main.c:787]: handle_sigs(): core was generated 2024-01-23T13:18:09.516+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2024-01-23T13:18:09.570+01:00 Started /root/sipconfig/startkamailio.sh 2024-01-23T13:18:09.570+01:00 info: :-) Starting Kamailio Just before all the crashes we see the "CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 49" log line. Best regards, Mattis Lind # gdb /usr/sbin/kamailio /core *GNU gdb (Debian 10.1-1.7) 10.1.90.20210103-git* Copyright (C) 2021 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: <https://www.gnu.org/software/gdb/bugs/>. Find the GDB manual and other documentation resources online at: <http://www.gnu.org/software/gdb/documentation/>. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from /usr/sbin/kamailio... (No debugging symbols found in /usr/sbin/kamailio) warning: Can't open file /dev/zero (deleted) during file-backed mapping note processing [New LWP 4777] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -DD -M 18 -m 192 -A serverId=17173 -A sendTraceLocal="sip:10'. Program terminated with signal SIGSEGV, Segmentation fault. #0 __strlen_evex () at ../sysdeps/x86_64/multiarch/strlen-evex.S:77 77 ../sysdeps/x86_64/multiarch/strlen-evex.S: No such file or directory. (gdb) bt full #0 __strlen_evex () at ../sysdeps/x86_64/multiarch/strlen-evex.S:77 No locals. #1 0x00007f5425e34b78 in __vfprintf_internal (s=s@entry=0x55a06e969a60, format=format@entry=0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n", ap=ap@entry=0x7ffd6bebfb50, mode_flags=mode_flags@entry=0) at vfprintf-internal.c:1647 len = <optimized out> step0_jumps = {0, 1717, 1621, 3413, 3317, 3997, 2677, 2837, 3613, 1773, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 4101} space = <optimized out> is_short = <optimized out> use_outdigits = 0 outc = <optimized out> step1_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 1773, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 0} group = 0 prec = -1 step2_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 0} string = 0x756d6f7266222c22 <error: Cannot access memory at address 0x756d6f7266222c22> left = 0 is_long_double = <optimized out> width = 0 signed_number = <optimized out> step3a_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4213, 0, 0, 0, 4389, 2789, 4197, 3917, 3221, 0, 0, 0, 0, 1925, 0, 0, 0, 0, 0, 0} alt = <optimized out> showsign = 0 is_long = 0 is_char = <optimized out> pad = <optimized out> step3b_jumps = {0 <repeats 11 times>, 3517, 0, 0, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 0, 0, 0} step4_jumps = {0 <repeats 14 times>, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 0, 0, 0} args_value = <optimized out> is_negative = <optimized out> number = {longlong = <optimized out>, word = <optimized out>} base = <optimized out> the_arg = {pa_wchar = 4777 L'\x12a9', pa_int = 4777, pa_long_int = 4777, pa_long_long_int = 4777, pa_u_int = 4777, pa_u_long_int = 4777, pa_u_long_long_int = 4777, pa_double = 2.3601515901836347e-320, pa_long_double = 1.74131181638025811763e-4947, pa_float128 = 3.09319115455554459548860449034534676e-4962, pa_string = 0x12a9 <error: Cannot access memory at address 0x12a9>, pa_wstring = 0x12a9 <error: Cannot access memory at address 0x12a9>, pa_pointer = 0x12a9, pa_user = 0x12a9} spec = 115 's' _buffer = {__routine = 0x4, __arg = 0xd, __canceltype = 1855363680, __prev = 0xe0} _avail = <optimized out> thousands_sep = 0x0 grouping = 0xffffffffffffffff <error: Cannot access memory at address 0xffffffffffffffff> done = 238 f = 0x55a06e5580a7 "s:%lu. Exec from %s:%u.\n" lead_str_end = 0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n" end_of_spec = <optimized out> work_buffer = "h\r\000\000\000\000\000\000\000\000\000\000\060\000\000\000\000\000\000\000\375\177\000\000 \372\353k\375\177\000\000\237MSn\n\000\000\000\000\000\000\000\240U", '\000' <repeats 18 times>, "P\225Un\240U\000\000\000\000\000\000[\214A\323\f\000\000\000\000\000\000\000\377\377\377\377\377\377\377\377\206\002", '\000' <repeats 14 times>, "\004\000\000\000\000\000\000\000 \367\353k\375\177\000\000\301\225Un\240U\000\000\323\262\vn\240U\000\000\000\000\000\000\000\000\000\000s\374On\240U\000\000W\225Un\240U\000\000\350>\212\027T\177\000\000 \323\365$T\177\000\000\060\a\354k\375\177\000\00--Type <RET> for more, q to quit, c to continue without paging-- 0\220\371\227n\240U\000\000"... workend = 0x7ffd6bebf9f8 "" ap_save = {{gp_offset = 16, fp_offset = 48, overflow_arg_area = 0x7ffd6bebfc30, reg_save_area = 0x7ffd6bebfb70}} nspecs_done = 10 save_errno = 4 readonly_format = 0 do_longlong_number = <optimized out> __result = <optimized out> #2 0x00007f5425ec079f in __vsyslog_internal (pri=<optimized out>, fmt=0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n", ap=0x7ffd6bebfb50, mode_flags=0) at ../misc/syslog.c:233 now_tm = {tm_sec = 8, tm_min = 18, tm_hour = 12, tm_mday = 23, tm_mon = 0, tm_year = 124, tm_wday = 2, tm_yday = 22, tm_isdst = 0, tm_gmtoff = 0, tm_zone = 0x55a06e94c5e0 "UTC"} now = 1706012288 fd = <optimized out> f = 0x55a06e969a60 buf = 0x0 bufsize = 0 msgoff = 21 saved_errno = <optimized out> failbuf = "`\232\226n\240U\000\000\000\204\201\247[\214A\323`\374\353k\375\177\000\000\300p\371%T" clarg = {buf = <optimized out>, oldaction = <optimized out>} #3 0x00007f5425ec0c46 in __syslog (pri=<optimized out>, fmt=<optimized out>) at ../misc/syslog.c:117 ap = {{gp_offset = 48, fp_offset = 48, overflow_arg_area = 0x7ffd6bebfc70, reg_save_area = 0x7ffd6bebfb70}} #4 0x000055a06e3b7839 in ?? () No symbol table info available. #5 0x000055a06e3bc039 in qm_free () No symbol table info available. #6 0x000055a06e3c7c28 in qm_shm_free () No symbol table info available. #7 0x00007f542325fb8e in uac_send_tm_callback () from /usr/lib/x86_64-linux-gnu/kamailio/modules/uac.so No symbol table info available. #8 0x00007f5424a2f002 in run_trans_callbacks_internal () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #9 0x00007f5424a2f179 in run_trans_callbacks () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #10 0x00007f54249d5e8c in free_cell_helper () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #11 0x00007f5424aa8f82 in wait_handler () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #12 0x000055a06e37a263 in ?? () No symbol table info available. #13 0x000055a06e37a79d in ?? () No symbol table info available. #14 0x000055a06e37acc6 in timer_main () No symbol table info available. #15 0x000055a06e0a5f62 in main_loop () No symbol table info available. --Type <RET> for more, q to quit, c to continue without paging-- #16 0x000055a06e0b120c in main () No symbol table info available.

2 3

How to achieve conversion from 180 to 183sdp
by mm e 29 Jan '24

29 Jan '24

When receiving to-tag and returning 180 ring without SDP body, how can it be converted into 183 with SDP and returned to from-tag. I want to realize that when I receive 180 no sdp, I will play an audio clip for the from-tag.： INVITE (SDP) ──────────────────────────> │ 100 Trying <────────────────────────── │ │ INVITE (SDP) │ ──────────────────────────> 100 Trying │ <────────────────────────── 180 Ringing │ <────────────────────────── 183 Session Progress (SDP) │ <────────────────────────── │

2 1

usrloc database duplicate keys issue
by Benoît Panizzon 29 Jan '24

29 Jan '24

Hi I have come across an issue with the usrloc database on kamailio 5.6 and 5.7 Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: <core> [db_query.c:244]: db_do_insert_cmd(): error while submitting query Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: usrloc [ucontact.c:686]: db_insert_ucontact(): inserting contact in db failed somedudetp0216 (uloc-65b237f8-28983d-1) Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: usrloc [urecord.c:419]: wb_timer(): inserting contact into database failed (aor: somedudetp0216) settings: # ----- usrloc params ----- modparam("usrloc", "db_url", DBLOCAL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "ka_mode", 0) modparam("usrloc", "ka_filter", 1) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "ka_from", "sip:ping@sip.imp.ch") modparam("usrloc", "server_id_filter", 1) # Use DMQ to sync userloc loadmodule "dmq_usrloc.so" # Replicate usrloc via dmq # Enable userloc sync modparam("dmq_usrloc", "enable", 1) # Do a full sync on start modparam("dmq_usrloc", "sync", 1) I assume, when kamailio is being restartet there are some expired entries still present in the database which kamailio tries to insert on a new registration. This leads to this duplicate key error. Is there a way to tell kamailio to use 'replace into' instread of 'insert into'? Solution for now, is to manually delete expired entries from the database. -- 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 6

Heads-up: testing needed for major OpenSSL refactoring
by Richard Chan 28 Jan '24

28 Jan '24

Hello Kamailio community, After the January refactoring of OpenSSL integration, configurations that initialise libssl in rank 0 (thread #1) are likely to trigger errors more aggressively. This type of configuration has all along been broken due to OpenSSL use of thread-local variables. The first "victim" is https://github.com/kamailio/kamailio/issues/3727 — the configuration has dispatcher + db_unixodbc (+ some driver) — this configuration initialises libssl in rank 0(thread #1). I have a couple of PRs in the queue for db_unixodbc, db_mysql (PostgreSQL may need similar attention). master and 5.7 branches are affected Regards Richard

1 0

LCR logic
by Social Boh 27 Jan '24

27 Jan '24

Hello, LCR module logic is : When the function /load_gws()/ is called, matching gateways (that are not currently designated as defunct) are ordered for forwarding purposes as follows: 1. according to longest Request-URI user part match 2. according to tuple's priority 3. according to tuple's randomized weight or, if priority_ordering parameter is set to value 1, as follows: 1. according to tuple's priority 2. according to tuple's randomized weight and Smaller priority value means higher priority (highest priority value being 0 and lowest being 255). so I have 3 lcr_rule_target use the same lcr_rule entry but 3 different Gateways and priority. So the order of rules would be longest Request-URI (same) and the priority but the result is: added matched_gws[0]=[3, 5, 200, 2193562] added matched_gws[1]=[1, 5, 2, 6888519] added matched_gws[2]=[2, 5, 50, 2159388] same prefix but first rule target with priority 200 and then 2 and then 50. I think the result would be priority 2,50 and 100. ¿Or not? --- I'm SoCIaL, MayBe

2 2

DMQ dialog sync question
by Stefan-Cristian Mititelu 26 Jan '24

26 Jan '24

Hi, I'm trying a DMQ dialog sync up setup with 2 x Kamailios. I have this scenario: 1. Kamailio1 receives INVITE and sends KDQM to Kamailio2 to sync up dialog => OK 2. Kamailio2 receives 200OK 3. Kamailio2 does not send anything back to Kamailio1 to update dialog state => not OK, i expected a KDMQ to be send back to Kamailio1 to update dialog state I was wondering if DMQ module supports the above? I am using 5.6 version of kamailio, and I am curios if this might be implemented in later branches? Thank you, Stefan

2 3

How to correctly drop a branch?
by Benoît Panizzon 26 Jan '24

26 Jan '24

Hi https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#branch_name_-… "Assigning $null to uri attribute will drop the branch, for the rest of attributes will just set the value to null." Where do I need to assing $null to the brach URI to get it dropped? And what happens to the index, is there a gap, or do the subsequent branches get re-indexed so they don't match with the ulc indexes anymore? Actual example, I have multiple contacts I would like to add to branches. Two of the contacts are from the location database + an additional contact. # Basic Branch: $ru = $(ulc(aor=>addr)[0]); $du = $(ulc(aor=>received)[0]); $fs = $(ulc(aor=>socket)[0]); # Additional Branch: append_branch($(ulc(aor=>addr)[1])); $(branch(dst_uri)[-1]) = $(ulc(aor=>received)[1]); $(branch(send_socket)[-1]) = $(ulc(aor=>socket)[1]); and so on, actually a loop over all ulc contacts. # Lasst Additional Branch, point to other registrar, not to a usrloc location. append_branch($ru); $(branch(dst_uri)[-1]) = "sip:other-registrar:5060;transport=udp"; $fsn = "LocalUDPv4"; $(branch(send_socket)[-1]) = $fs; t_on_branch("BR_T"); branch_route[BR_T] { $var(socket) = $(ulc(aor=>socket)[$T_branch_idx]); if ($var(socket) == 0) { xlog("L_INFO", "$cfg(route): DROPPING BRANCH: $T_branch_idx No local socket on this registrar\n"); $(branch(uri)[$T_branch_idx]) = $null; } } I don't get this right! Call is still sent to that contact. -- 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 ______________________________________________________

1 2

Change in LCR Debug
by Social Boh 26 Jan '24

26 Jan '24

Hello, using DEBUG with LCR module to see how the module apply the rules, I see he use Gateway Index in the list. The Gateway Index could be different to Gateway ID and maybe can confuse the correct interpretation about the use of the rules. Is it possible change the code: matched_gws[gw_index].gw_index = t->gw_index; matched_gws[gw_index].rule_id = rule->rule_id; matched_gws[gw_index].prefix_len = pl->prefix_len; matched_gws[gw_index].priority = t->priority; matched_gws[gw_index].weight = t->weight * (kam_rand() >> 8); matched_gws[gw_index].duplicate = 0; LM_DBG("added matched_gws[%d]=[%u %u, %u, %u, %u]\n", gw_index, t->gw_index, pl->prefix_len, t->priority, matched_gws[gw_index].weight); to use gateway ID and not gateway Index? Thank you Regards -- --- I'm SoCIaL, MayBe

1 0

How to get media IP
by mm e 26 Jan '24

26 Jan '24

When receiving a call I get the following message： INVITE sip:123456789@192.168.1.118:29363 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport Max-Forwards: 70 Contact: <sip:96341@192.168.1.118:65178> To: "123456789"<sip:123456789@192.168.1.119:29363 > From: "96341"<sip:96341@192.168.1.119:29363 >;tag=af3c7c31 Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011d stamp 40820 Content-Length: 339 v=0 o=- 3 2 IN IP4 192.168.1.118 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.118 t=0 0 m=audio 32170 RTP/AVP 0 8 18 101 a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv How to print the media IP in SDP separately in the log? I didn't find the Pseudo-Variables..can you help me

3 2

412 Conditional request failed
by satyaprakash ch 26 Jan '24

26 Jan '24

Hi There were two PUBLISH requests were sent by the Kamailio Proxy server to the Kamailio Presence server. Processing of the first PUBLISH request resulted in the presentity table being updated - the "etag" changed from "a.1705053846.16611.21.1" to "a.1705053846.16613.23.2" Processing of the second PUBLISH failed because it also attempted to update the etag in the presentity table but at this point in time there was no match to the original etag value in the database Logs in kamailio.log: ------ ps_db_update_presentity(): No E-Tag match a.1705053846.16611.21.1 DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7f81e5ca5fc0: SIP/2.0 412 Conditio... shmem=0x7f81de6da068: SIP/2.0 412 Conditional request failed. Will any one suggest how to resolve the issue of 412 conditional request failed.

2 2

Is the Contact header required for the dialog module?
by Unai Rodriguez 26 Jan '24

26 Jan '24

Dear List, I’m trying to set up the dialog module and I keep getting the error "bad sip message or missing Contact hdr”: --- Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing Contact hdr Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the dialog --- This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look like this: ---- INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0 Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ Max-Forwards: 70 From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua To: <sip:some.mrcp.server.example.com:8060> Call-ID: 051df0ff-3541-123d-f090-02f373ff014c CSeq: 78491159 INVITE User-Agent: google_stt Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 329 v=0 o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10 s=- c=IN IP4 10.10.10.10 t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:new a=resource:speechrecog a=cmid:1 m=audio 12034 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 L16/8000 a=sendonly a=mid:1 ---- According to the RFC, the Contact header is not mandatory (https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog module to do without the Contact header? Any pointers appreciated. Thank you so much With best wishes, Unai Rodriguez

3 5

How to assign xavp array to htable?
by Benoit Panizzon 26 Jan '24

26 Jan '24

Hi List Via sql_xquery I get a stacked xavp aka array. I would like to store that stacked xavp into a hash table for later usage in other transactions. Is this possible? Via xarp_parameters explode or implode does not work as that would stringy the same key multiple times. 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

How to set a branch specific variable?
by Benoit Panizzon 25 Jan '24

25 Jan '24

Hi Gang I would like to set a variable specific to the branch. Is there a way? append_branch($var(aor1)); $(branch(dst_uri)[-1]) = $var($dst1) $(branch(want_plus)[-1]) = 1; append_branch($var(aor2)); $(branch(dst_uri)[-1]) = $var($dst2) $(branch(want_plus)[-1]) = 0; t_on_branch("BR_T"); branch_route[BR_T] { if ($branch(want_plus)) { $rU = "+" + $rU; } } Or would there be a way to query the index of the added branch so I could stack an avp with that index to access in the branch route? 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 ______________________________________________________

1 0

Re: STIR/SHAKEN
by Anthony Wittig 25 Jan '24

25 Jan '24

Hey everyone, I've set it up so that when the invite comes into Kamailio I suspend the transaction (save the transaction info in sht) and use dispatcher to send a request to our NSS server. On the reply from NSS I throw the identity header into a sht and continue the original transaction. Does that seem like the right way to do it? The next things I need to look at are: * if first NSS server doesn't respond quickly, try the second (change some timeouts and use dispatcher to try the backup) * if both NSS servers aren't working, I still want to resume the original call and let it through. Maybe I can keep track of the failures in a sht and if I have two failures, resume the original invite without the identity header. Sound like an ok plan? -- Anthony Wittig Mango Voice Developer > > ---------- Forwarded message ---------- > From: Anthony Wittig <awittig(a)mangovoice.com> > To: sr-users(a)lists.kamailio.org, david.villasmil.work(a)gmail.com > Cc: > Bcc: > Date: Tue, 23 Jan 2024 20:39:09 -0700 > Subject: [SR-Users] STIR/SHAKEN > Thanks! > > This is what I've got so far: > > 1. Kamailio invite -> NSS > 2. NSS 302 -> Kamailio > 3. Kamailio ack -> NSS > > I can see the identity header on the 302 response in my onreply_route. How > can I replay the same invite from step #1 with the identity header I get > back in step #2? Should I be doing something with a transaction on step #1, > maybe start a branch that goes out to NSS and when it comes back resume the > transaction (these are all words I've heard smart people say, but I don't > fully understand them...). > > Any pointers would be greatly appreciated, thanks! > -- > Anthony Wittig > Mango Voice Developer > > ---------- Forwarded message ---------- > From: David Villasmil <david.villasmil.work(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Tue, 23 Jan 2024 14:56:52 +0100 > Subject: [SR-Users] Re: STIR/SHAKEN > Ditto, i do first approach. > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> These are all valid approaches, depending on preference. Catching 302s >> and extracting exactly the desired info might be easier with Kamailio, so I >> suppose, ceteris paribus, I'd recommend the first one. >> >> Dispatcher is a great approach! >> >> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < >> sr-users(a)lists.kamailio.org> wrote: >> > >> > Hello, >> > >> > We're attempting to add identities to our invites. I believe we want to >> send our invites to the NSS, and it'll reply with a 302'd invite with an >> identity header. We currently use FreeSWITCH for dialplans and media and >> Kamailio as our SBC. >> > >> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS >> (302) -> Kamailio -> PSTN >> > >> > or would it make more sense to do: FreeSWITCH -> NSS (302) -> >> FreeSWITCH -> Kamailio -> PSTN >> > >> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right >> approach? We currently use it for our two Kamailios and our FreeSWITCHs. >> We're going to have two NSS servers, one for failover. >> > >> > Any thoughts would be greatly appreciated, thanks! >> > >> > -- >> > Anthony Wittig >> > Mango Voice Developer >> > __________________________________________________________ >> > Kamailio - Users Mailing List - Non Commercial Discussions >> > To unsubscribe send an email to sr-users-leave(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: >> >> -- >> Alex Balashov >> Principal Consultant >> Evariste Systems LLC >> Web: https://evaristesys.com >> Tel: +1-706-510-6800 >> >

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STIR/SHAKEN
by Anthony Wittig 24 Jan '24

24 Jan '24

Thanks! This is what I've got so far: 1. Kamailio invite -> NSS 2. NSS 302 -> Kamailio 3. Kamailio ack -> NSS I can see the identity header on the 302 response in my onreply_route. How can I replay the same invite from step #1 with the identity header I get back in step #2? Should I be doing something with a transaction on step #1, maybe start a branch that goes out to NSS and when it comes back resume the transaction (these are all words I've heard smart people say, but I don't fully understand them...). Any pointers would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer ---------- Forwarded message ---------- From: David Villasmil <david.villasmil.work(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: Bcc: Date: Tue, 23 Jan 2024 14:56:52 +0100 Subject: [SR-Users] Re: STIR/SHAKEN Ditto, i do first approach. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < sr-users(a)lists.kamailio.org> wrote: > These are all valid approaches, depending on preference. Catching 302s and > extracting exactly the desired info might be easier with Kamailio, so I > suppose, ceteris paribus, I'd recommend the first one. > > Dispatcher is a great approach! > > > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < > sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > > > We're attempting to add identities to our invites. I believe we want to > send our invites to the NSS, and it'll reply with a 302'd invite with an > identity header. We currently use FreeSWITCH for dialplans and media and > Kamailio as our SBC. > > > > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS > (302) -> Kamailio -> PSTN > > > > or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH > -> Kamailio -> PSTN > > > > Assuming we wanted Kamailio -> NSS, is using dispatcher the right > approach? We currently use it for our two Kamailios and our FreeSWITCHs. > We're going to have two NSS servers, one for failover. > > > > Any thoughts would be greatly appreciated, thanks! > > > > -- > > Anthony Wittig > > Mango Voice Developer > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(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: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(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: >

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STIR/SHAKEN
by Anthony Wittig 23 Jan '24

23 Jan '24

Hello, We're attempting to add identities to our invites. I believe we want to send our invites to the NSS, and it'll reply with a 302'd invite with an identity header. We currently use FreeSWITCH for dialplans and media and Kamailio as our SBC. Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS (302) -> Kamailio -> PSTN or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH -> Kamailio -> PSTN Assuming we wanted Kamailio -> NSS, is using dispatcher the right approach? We currently use it for our two Kamailios and our FreeSWITCHs. We're going to have two NSS servers, one for failover. Any thoughts would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer

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How to check for branches with a pending reply in MANAGE_REPLY to correctly handle rtpengine?
by Benoit Panizzon 23 Jan '24

23 Jan '24

Hi gang location database contains multiple contacts for an AOR The invite therefore is parallel branched to those multiple contacts. Each reply from each branch is handled by MANAGE_REPLY. rtpengine_manage() is called in MANAGE_REPLY to stop rtpengine processing a failed call. Unfortunately, if any of the contacts replies with an error rtpengine_manage() will stop RTP processing. Is there a way I could check if there are branches with a pending reply from within MANAGE_REPLY so I could call rtpengine_manage() only when the last reply (no pending replies) are present? Hmm, realizing, if I process the last pending error reply and a prior one got a positive reply, this would still kill media. => How do I solve this? 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 ______________________________________________________

1 0

Info: Kamailio World 2024 - Call for presentations
by Daniel-Constantin Mierla 22 Jan '24

22 Jan '24

Hello, a short note to inform that the Call For Presentations is now open for Kamailio World 2024. Everyone is welcome to submit proposal for presentations to share the knowledge about Kamailio or Real Time Communication services, security, high availability, scalability, etc. Submission form and more details are available at: - https://www.kamailioworld.com/k2024/call-for-speakers/ Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

2 2

IMS registration failure
by srinidhikrs＠yahoo.com 21 Jan '24

21 Jan '24

I am trying to get imsdroid register to pcscf . But failing with 504 server timeout . Please help to figure out the reason for register failure . <is_registered_fallback2ip> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <is_registered_fallback2ip> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <pcscf_unregister> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg.lex:1643]: sr_push_yy_state(): including file: route/register.cfg (/etc/kamailio_pcscf/route/register.cfg) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <pcscf_is_registered> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'core' (num=56, size=224) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'tcp' (num=27, size=108) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 0 (old) + 1 (new) = 1 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'tm' (num=36, size=168) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'maxfwd' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'siputils' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 1 (old) + 1 (new) = 2 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'xlog' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'dispatcher' (num=3, size=24) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'sctp' (num=19, size=76) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 2 (old) + 1 (new) = 3 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'dbg' (num=3, size=12) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'registrar' (num=11, size=56) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'rtpengine' (num=5, size=20) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 3 (old) + 1 (new) = 4 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 4 (old) + 1 (new) = 5 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 5 (old) + 1 (new) = 6 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:99]: init_pt(): registering new processes: 6 (old) + 96 (new) = 102 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/route.c:885]: fix_actions(): fixing pcscf_is_registered() [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/route.c:885]: fix_actions(): fixing pcscf_unregister() [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=9 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=7 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=9 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=10 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=11 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=12 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=15 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=22 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=17 a=25 n=is_present_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=18 a=25 n=pcscf_save_pending [0;39;49m[0;39;49m 4(310124) DEBUG: ims_registrar_pcscf [save.c:378]: save_pending(): registering for UL callback [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=25 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=32 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=35 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=33 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=81 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=83 a=25 n=remove_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=84 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=85 a=25 n=remove_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=86 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=94 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=89 a=25 n=is_present_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=93 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=107 a=25 n=t_on_failure [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=109 a=25 n=t_on_reply [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=111 a=5 n=route [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=316 a=16 n=if [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=311 a=25 n=t_check_status [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=312 a=26 n=send_reply [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=313 a=26 n=update_stat [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=314 a=2 n=exit

1 2

failed to open database connection
by mtohidi67＠gmail.com 20 Jan '24

20 Jan '24

Hello. I have installed Kamailio version devel on Ubuntu 22.04 and use the mysql database on the same machine. when i run systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2024-01-20 11:35:09 +0330; 21min ago Docs: man:systemd-sysv-generator(8) CPU: 48ms /usr/local/sbin/kamailio[173481]: ERROR: usrloc [dlist.c:846]: register_udomain(): failed to open database connection /usr/local/sbin/kamailio[173481]: ERROR: registrar [registrar.c:767]: domain_fixup(): failed to register domain /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:717 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:720 /usr/local/sbin/kamailio[173481]: INFO: [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized kamailio[173463]: * already running kamailio[173463]: ...done. systemd[1]: Started LSB: Start the Kamailio SIP proxy server. In /usr/local/etc/kamailio/kamctlrc, I uncomment these lines: SIP_DOMAIN=ims.mnc55.mcc999.farazNET.org DBENGINE=MYSQL DBHOST=localhost in /usr/local/etc/kamailio/kamailio.cfg i apply this changes: auto_aliases=no alias="ims.mnc55.mcc999.farazNET.org" listen=udp:192.168.75.185:5060 listen=tcp:192.168.75.185:5060 how can i solved it?

1 1

rr, loose_route and corex alias_subdomains
by Jon Bonilla (Manwe) 19 Jan '24

19 Jan '24

Hi I'm having an issue with the double rr for mst direct routing. I'm using subdomains like 23234.mydomain.com with a wildcard certificate and I set corex modparam alias_subdomains to mydomain.com But loose_route doesn't seem to detect the rr as myself I get the message: rr [loose.c:804]: rr_do_force_send_socket(): no socket found to match second RR (sip:20141.mydomain.com:5063;transport=tls;ftag=as7107d0fe;lr;r2=on;vsf=AAAAAAAAAAADCQMaABhMS1NYRh0cRllbR0RRQG91cC5lcw--;nat=yes;pet=mst) Regarding my setup is more or less like this: kamailio.local.cfg: #!substdef "!MSTDOMAIN!mydomain.com!g" kamailio.cfg: modparam("corex", "alias_subdomains", "MSTDOMAIN") But in the cli: kamctl rpc corex.list_aliases { "jsonrpc": "2.0", "result": [ ], "id": 1319303 } I don't remember to have this issue when I was developing the mst integration. Either I missed it or can it be a regression? kamailio version is 5.6.4 at the moment. Any ideas? cheers Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

Just need some basic help
by Jere Cassidy 18 Jan '24

18 Jan '24

Hello! Thanks in advance for any help you can provide. I have a simple setup here and am struggling a bit with getting a final working solution. This is just a single direction flow from a SIP provider inbound to servers running HMPelements with CTI integration (sip endpoint). Essentially a different DN / dial number would go to a different backend server. There is no registration involved. The kamailio server therefore needs to sit between the SIP provider and the backend servers, it will be a NAT situation where private RFC1918 addresses will be used on the backend after the server receives it with public IP information. So that's it. What is the simplest way to accomplish this? If it would help - here's what ive done so far: For the Kamailio server, I have an AWS EC2 instance with a public IP bound/assigned to the main private IP. During troubleshooting, I've added another private IP to the NIC so that I could throw up another listener. I was thinking I'd need to accept the SIP on the first listener with the "advertise" directive. The second listener would not have an advertise directive so perhaps could be used to talk to the backends. I defined the WITH NAT and rtp proxy settings (pointing to the first listener below). The RTP Proxy daemon also knows about the y.y.y.y public address below. In the kamailio.cfg, I have a listener setup : listen=udp:172.24.40.28:5060 advertise y.y.y.y:5060 listen=udp:172.24.40.26:5060 then threw in a route(CMS) in the request route and created the following route block: route[CMS] { #check URI matches phone xxxxxxxxxx if (uri=~"^sip:\+1xxxxxxxxxx@") { rewritehostport("10.12.18.193:5060"); forward(uri:host, uri:port); exit; }; } I get the SIP delivered to the backend host! but the INVITE headers are stil pointing to public IPs and the SDP info is not adjusted to be a private IP as well - as follows: z.z.z.z = sip provider , y.y.y.y = public ip (aws ec2 ip). xxxxxxxxx = calling phone, ######## = called phone (matches routeCMS above) INVITE sip:+1##########@10.12.18.193:5060 SIP/2.0 Record-Route: <sip:y.y.y.y;lr> Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK28b7.56b42fdeec9f10cd13030b3711b16f4d.0 Via: SIP/2.0/UDP z.z.z.z:11000;received=z.z.z.z;rport=11000;branch=z9hG4bKQ2yQKgv7NtKQF Max-Forwards: 48 From: "" <sip:+1xxxxxxxxxx@z.z.z.z;isup-oli=0>;tag=tUF15gaNg6N6j To: <sip:+1##########@y.y.y.y> Call-ID: 9ee7e073-c7fc-46ac-a839-f3c77eaed903 CSeq: 78244364 INVITE Contact: <sip:mod_sofia@z.z.z.z:11000> User-Agent: 2600Hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 278 X-FS-Core-UUID: 4e0b2928-d913-42c4-9809-f77a36d281be X-FS-Support: update_display,send_info P-Asserted-Identity: "" <sip:+1xxxxxxxxx@z.z.z.z> v=0 o=FreeSWITCH 1705572686 1705572687 IN IP4 z.z.z.z Thanks much!

2 1

Re: add a SIP custom header on a SIP response generated locally
by frédéric Gaisnon 18 Jan '24

18 Jan '24

Hi, Yes I've tried using append_to_reply() in several routes (tm:branch-failure, sl:local-response, tm:local-response, onreply_route, failure_route) but it was not applied. It works fine if a response is received or if it is generated by the configuration script but not if this response is locally generated by a kamailio module. Regards, Frédéric Gaisnon > Custom headers to the reply can be added using append_to_reply() function from textops module. Have you tried that? > > Regards, > > Fred Posner > p: +1 (352) 664-3733 > > On Jan 18, 2024, at 2:07 AM, frédéric Gaisnon via sr-users < sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) > > I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. > > Do you have any suggestions to accomplish this ? > > Regards, > > Frédéric Gaisnon

2 1

trigger different branch route on branch added with append_branch
by Benoit Panizzon 18 Jan '24

18 Jan '24

Hi List Imagine the following situation: Two Kamailio registrar proxies handling the same domain. Location Information synced via DMQ. So if a device registers we don't know on which of the two registrar proxies. Core proxy is dispatching calls to either one of those registrars. Core also performs AOR database lookup and adds an X- header telling which AOR the call is destined to. Registrar performs lookup of that AOR in the location database and routes the call to destination. Problem: The call could be routed to the the registrar which does not hold the active socket to that client. Solution: reg_fetch_contacts, cycle the contacts and append_branch if the contact has a local socket. Next Problem: I use a failure_route to sent the call to voicemail, if not registered, busy etc. This would trigger, if we have no local socket for the registration which is on the other registrar. So I have to find a way to send the call to the other registrar. I guess I can append_branch a branch pointing to the URI of the other registrar. This would then succeed in reaching the device. Also using the failure_route would work this way if we get a remote error from behind the other registrar or no registration at all. Next Problem: I use a branch trigger to remove undesired header I do not want to send to the customer. Now imagine that AOR has two contacts, one on registered on each registrar proxy. So I end up with two destinations added with append_branch to the destination set. One points to the device of the locally registered customer device, the other one points to the other registrar where to find the other device. I use a branch route trigger to remove the X- Headers towards the customer. But I need to X- Header containing the destination AOR to still be sent to the other registrar. Is there a way to trigger a branch route for only some of the destinations added with append_branch? Or to trigger different branch routes for different destinations in one set added with append_branch? Or am I doing this completely wrong and there is an easier way? ;-) 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 ______________________________________________________

2 2

How to change uri for db_mysql ?
by iglov 18 Jan '24

18 Jan '24

Sup guyz! On debian 12 i have kamailio 5.6.3 and when I try to use db_mysql module I always get an error: kamailio[2842268]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local server through socket '/run/mysqld/mysqld.sock' I have no idea where it finds this socket cuz I don’t have it and don’t want to. Where I can change it?

4 7

add a SIP custom header on a SIP response generated locally
by frédéric Gaisnon 18 Jan '24

18 Jan '24

Hello, I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. Do you have any suggestions to accomplish this ? Regards, Frédéric Gaisnon

3 2

Kamailio v5.7.4 Released
by Daniel-Constantin Mierla 18 Jan '24

18 Jan '24

Hello, Kamailio SIP Server v5.7.4 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.4. For more details about version 5.7.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/01/kamailio-v5-7-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Releasing v5.7.4
by Daniel-Constantin Mierla 18 Jan '24

18 Jan '24

Hello, I am considering to release Kamailio v5.7.4 (out of branch 5.7) on Thursday, Jan 18, 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 Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

IMS Register failure
by srinidhi 2wewew 18 Jan '24

18 Jan '24

I am trying to get imsdroid register to pcscf . But failing with 504 server timeout . PCSCF IP : 192.168.1.28 imsdroid IP : 192.168.1.212 Please help to figure out the reaso for register failure . I have attached pcap file and pcscf log file

1 0

tls.cfg and configuration
by Aymeric Moizard 17 Jan '24

17 Jan '24

Hi all, I'm using a central file to define my kamailio cfg configuration. First question: In my main kamailio.cfg, I use: include_file "kamailio-config.cfg" which contains: #!substdef "!MY_DOMAIN!kamailio.example.com!g" I would like to use the same in tls.cfg which is defined this way: modparam("tls", "config", "/etc/kamailio/tls.cfg") but I get: sr_cfg_parse(): tls.cfg:17:1: Unsupported option 'include_file' Would you advise some way to achieve my goal: ie, I would like only one file to edit my configuration options which would have an effect on both kamailio.cfg and tls.cfg Second question: Can someone confirm that #!substdef will also not have any effect when used in tls.cfg Thanks Aymeric -- Antisip - http://www.antisip.com

2 1

2024 FOSDEM VoIP Dinner
by Torrey Searle 15 Jan '24

15 Jan '24

Hello all, It is that time of year to start getting the headcount for the FOSDEM voip dinner on the Saturday of FOSDEM. Usually Daniel would get the headcount for me and I just make the booking. However Daniel can't make it this year so I'll also be taking the headcount If you want to come put your name up one the doodle I've set up for this https://doodle.com/meeting/participate/id/boAjDENe Hope to see you there! Torrey

1 0

Not checking SRV records
by Jon Bonilla (Manwe) 13 Jan '24

13 Jan '24

Hi This may be a stupid question but after a couple of hours and some headache I can't see where the problem is. I'm trying to use SRV records to relay to different servers as I've done many many times before. But in this case I'm getting a 478 from kamailio and I see that it's not trying to resolve SRV. From the log: [core/dns_cache.c:527]: _dns_hash_find(): (fs.mydomain.dev(16), 1), h=632 [core/resolve.c:784]: get_record(): lookup(fs.mydomain.dev, 1) failed [core/dns_cache.c:761]: dns_cache_mk_bad_entry(): (fs.mydomain.dev, 1, 60, 1) [core/dns_cache.c:701]: dns_cache_add(): adding fs.mydomain.dev(16) 1 (flags=1) at 632 [core/dns_cache.c:2807]: dns_a_resolve(): (fs.mydomain.dev, 0) returning -7 [core/dns_cache.c:3176]: dns_srv_sip_resolve(): (fs.mydomain.dev, 0, 0), ip, ret=-7 uri2dst2(): failed to resolve "fs.mydomain.dev" :unresolvable A or AAAA request (-7) When doing tcpdump port 53 I see A record query but no SRV SRV records of course exist although they are not being checked. dns params in kamailio: dns_try_ipv6=no dns_retr_time=1 dns_retr_no=1 dns_use_search_list=no use_dns_cache=on use_dns_failover=on dns_srv_lb=on dns_try_naptr=no dns_sctp_pref=1 dns_tcp_pref=1 dns_tls_pref=1 dns_udp_pref=1 Same exact config in another server is working. There's something I'm missing and can't see. Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 1

Standard on RTP ports
by ZZ Wave 12 Jan '24

12 Jan '24

Hello. I have some rant with network security dept. Is there some standard, RFC, what port range should be used for RTP traffic, how RTP ports are selected during SIP call negotiation? Thanks.

11 25

Internal DNS resolver
by Igor Potjevlesch 12 Jan '24

12 Jan '24

Hello! I have a quick question about the internal DNS resolver and cache embedded in Kamailio. Is the ability to add an entry with dns.add_a bypass the "real" DNS lookup? So, let's say that sip.example.com is publicly resolve alternatively with 1.2.3.4 & 5.6.7.8.9, if I do a dns.add_a to add my own sip.example.com with, alternatively, 127.0.0.1 and 127.0.0.2, will it work? Thanks for your highlight! Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

3 7

xavp_params_explode broken comma in value?
by Benoît Panizzon 12 Jan '24

12 Jan '24

Hey! I'm hunting down an issue with xavp_params_explode, maybe somebody has already come adross it and could confirm it exists and maybe know a solution? xavp_params_explode("a=foo;c=hello,world;e=baar", "x"); $xavp(x=>a) is 'foo' but x=>c and x=>e are null Am I right to assume the issue being caused by the comma? Is there a way to allow a comma? -- 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 4

registrar questions
by mm e 12 Jan '24

12 Jan '24

In addition to the RPC command "kamctl add username password", is there a module with a method to operate the "subscriber" database table?

2 4

Clarification on usrloc timer_interval param and contact-expires event route.
by James Lipski 12 Jan '24

12 Jan '24

Hello, Just need a bit of clarification on the usrloc parameter "timer_interval". Let's say this is configured with a value of 60 seconds; every 60 seconds, will this delete all expired contacts in the location table between runs -- as in between runs it will queue up a list of expired contacts, remove them all after the 60 seconds has elapsed, then execute the event route "event_route[usrloc:contact-expired]"? Or does the event route execute as soon as the contact expires, independent from "timer_interval" timer? Thank you.

2 1

Setting kamailio to use another DB
by SAMUEL MOYA TINOCO 11 Jan '24

11 Jan '24

Good evening everyone, I'm trying to configure my Kamailio to use another DB in case the one it commonly uses fails. Does Kamailio has any built in function to do this? Do I have to do it with an ha-proxy? Thank you in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DA431B.CC9B03A0] Soluciones inteligentes para la autonomía personal

4 4

How to know registration status
by mm e 10 Jan '24

10 Jan '24

After using the uac module to initiate registration to the remote end, is there a method to call back to notify the registration result? I want to get the registration result, such as 200 or 403

4 3

Re: force TLS RSA cipher suite (SOLVED)
by marek 10 Jan '24

10 Jan '24

good catch. thanks for the record ECC [root@sbc live]# openssl x509 -text -noout -in /etc/letsencrypt/live/<my domain>/fullchain.pem |grep -i "Public Key Algorithm" Public Key Algorithm: id-ecPublicKey certbot certonly --key-type rsa --cert-name <my domain> --manual --preferred-challenges dns RSA [root@sbc live]# openssl x509 -text -noout -in /etc/letsencrypt/live/<my domain>/fullchain.pem |grep -i "Public Key Algorithm" Public Key Algorithm: rsaEncryption but there was another problem modparam("tls", "cipher_list", "RSA") from https://kamailio.org/docs/modules/5.7.x/modules/tls.html#tls.p.cipher_list does nothing 0(2294) INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' valid option is in tls.cfg cipher_list = RSA 0(2328) INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='RSA' ServerHello Version 3.3 session_id[0]= cipherSuite TLS_RSA_WITH_AES_256_GCM_SHA384 Marek Cervenka Dne 2024-01-10 v 13:06 Lukas Tribus napsal(a): > > > On Wednesday 10 January 2024, marek via sr-users > <sr-users(a)lists.kamailio.org> wrote: > > hi, > > i'm trying force cipher list through options like > > modparam("tls", "cipher_list", "TLS_RSA_WITH_AES_256_CBC_SHA256") > > modparam("tls", "cipher_list", "RSA") > > > You are trying a RSA cipher. > > > ... > > > but > > ssldump -i enp2s0 port 5061 shows every time > > ServerHello > Version 3.3 > > > In TLS 1.2 > > > > session_id[0]= > cipherSuite TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384 > > > But the negotiated cipher suggest that you have an ECC certificate. > > In TLSv1.2 and older, ciphers available are dependent on the > certificate type (RSA vs ECC). > > In TLSv1.3 its all different again and the certificate doesn't matter > for cipher negotiation. > > Get a RSA (2048bit) certificate instead of a ECC one from Let's > Encrypt, then you should be able to use those ciphers. > > > > Lukas

1 0

Timeout on forwarding REGISTRATION
by dries＠degendt.com 10 Jan '24

10 Jan '24

Hi all, My Kamailio server has been throwing a lot of errors during the manipulation and forwarding of registrations to my Asterisk server (192.168.96.97) using uac_req_send(); and t_relay() [core/tcp_main.c:1402]: tcp_do_connect(): connect 192.168.96.97:5060: (99) Cannot assign requested address [core/tcp_main.c:1399]: tcp_do_connect(): connect 192.168.96.97:5060 failed Cannot assign requested address When looking deeper into the code it appears that the errors are thrown when "tcp_async" is enabled. I've enabled this option a few weeks ago to make sure clients can REGISTER quickly. The parameter "tcp_reuse_port" is also set to yes. I'm suspecting that on busy moments, the destination can't be reached because no TCP connections are available. Is there any of the following parameters that I can adjust to prevent this issue? tcp_children = 25 disable_tcp = no tcp_max_connections = 63536 tcp_connection_lifetime = 3605 tcp_accept_aliases = no tcp_async = yes tcp_connect_timeout = 10 tcp_conn_wq_max = 65536 tcp_crlf_ping = yes tcp_delayed_ack = yes tcp_fd_cache = yes tcp_keepalive = yes tcp_keepcnt = 3 tcp_keepidle = 30 tcp_keepintvl = 10 tcp_linger2 = 30 tcp_rd_buf_size = 80000 tcp_send_timeout = 10 tcp_wq_blk_size = 2100 tcp_wq_max = 10485760 tcp_reuse_port=yes /* tcp_connection_match=1 */ open_files_limit=63536 A quick tcpdump doesn't show outgoing packet during the error events. Best regards, Dries

1 0

force TLS RSA cipher suite
by marek 10 Jan '24

10 Jan '24

hi, i'm trying force cipher list through options like modparam("tls", "cipher_list", "TLS_RSA_WITH_AES_256_CBC_SHA256") modparam("tls", "cipher_list", "RSA") ... but ssldump -i enp2s0 port 5061 shows every time ServerHello Version 3.3 session_id[0]= cipherSuite TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384 i tried lower crypto policy to LEGACY. nothing changed https://access.redhat.com/documentation/en-us/red_hat_enterprise_linux/9/ht… certificates are from Lets Encrypt if that does matter OS Rocky9(RHEL9), kamailio 5.7.3 from official repo any ideas/tips? thanks Marek

2 1

Re: uac_req_send + evroute + crash
by Péter Dr. Barabás 10 Jan '24

10 Jan '24

Hello, Any idea about the crash below? I found a ticket regarding to this, I made some comments and backtraces there also: https://github.com/kamailio/kamailio/issues/3635 Now this crash is very frequently, 3-4 times every day. There is a timer mechnism, where kamailio refreshes registrations towards asterisk automically to ensure continuous availability for mobile users. The crash exists 3-5 secs after this refresh process, but not always. Sometimes this process runs fine only CRITICAL logs appears (mentioned below) after it, but sometimes kamailio crashes during writing these CRITICAL logs. Peter From: Dr. Barabás Péter via sr-users <sr-users(a)lists.kamailio.org> Date: Wednesday, 2023. November 15. 17:03 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com> Subject: [SR-Users] Re: uac_req_send + evroute + crash Hello Daniel, I’m using kamailio 5.7.2 currently and this misterious crash occurs again. Now core files are generated also, here is the trace: (gdb) bt #0 __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:50 #1 0x00007fae9a2d0859 in __GI_abort () at abort.c:79 #2 0x00005614ad79afd8 in qm_debug_check_frag (qm=0x7fae78e77000, f=0x7fae79805028, file=0x7fae77c8c134 "uac: uac_send.c", line=860, efile=0x5614ad936f99 "core/mem/q_malloc.c", eline=511) at core/mem/q_malloc.c:129 #3 0x00005614ad79f7c1 in qm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:511 #4 0x00005614ad7ab410 in qm_shm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:1350 #5 0x00007fae77c3686b in uac_send_tm_callback (t=0x7fae797dcf40, type=131072, ps=0x7fff8bd990d0) at uac_send.c:860 #6 0x00007fae783abc27 in run_trans_callbacks_internal (cb_lst=0x7fae797dcfe8, type=131072, trans=0x7fae797dcf40, params=0x7fff8bd990d0) at t_hooks.c:241 #7 0x00007fae783abd1e in run_trans_callbacks (type=131072, trans=0x7fae797dcf40, req=0x0, rpl=0x0, code=0) at t_hooks.c:261 #8 0x00007fae782bd8cb in free_cell_helper (dead_cell=0x7fae797dcf40, silent=0, fname=0x7fae783e9f62 "timer.c", fline=653) at h_table.c:165 #9 0x00007fae7837649c in wait_handler (ti=724701450, wait_tl=0x7fae797dcff8, data=0x7fae797dcf40) at timer.c:653 #10 0x00005614ad50be68 in timer_list_expire (t=724701450, h=0x7fae78ef5d18, slow_l=0x7fae78ef95e0, slow_mark=33641) at core/timer.c:857 #11 0x00005614ad50c3a7 in timer_handler () at core/timer.c:922 #12 0x00005614ad50c8d5 in timer_main () at core/timer.c:961 #13 0x00005614ad45fbf3 in main_loop () at main.c:1833 #14 0x00005614ad46ae93 in main (argc=14, argv=0x7fff8bd99b48) at main.c:3086 I think it occurs only if $uac_req(evroute) is set to 1 and I define the event_route[uac:reply] route. My uac_reply route is easy: if ($uac_req(evcode)==200) { xlog("L_INFO", "Registration for user $fU has been refreshed successfully."); $sht(vtp=>asterisk_restarted) = $null; route(SAVE_REG); } else { xlog("L_INFO", "Registration failed: $uac_req(evcode)"); } the SAVE_REG route is the following: route[SAVE_REG] { xlog("L_INFO", "Saving pbx registration...:$fU"); $var(maxExpiry) = MAX_PBX_REG_EXPIRY; route(GET_CLIENT_ID); sql_pvquery("ca", "select count(*) from pbxusers where user = '$fU'", "$var(pbxusersCnt)"); if ($var(pbxusersCnt)>0) { if ($var(clientId) == 0) { sql_query("ca", "update pbxusers set expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } else { sql_query("ca", "update pbxusers set client_id = '$var(clientId)', expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } } else { sql_query("ca", "insert into pbxusers (user, expiry, client_id, online) values ('$fU', TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), '$var(clientId)', 0)"); } } And GET_CLIENT_ID route is: route[GET_CLIENT_ID] { $var(clientId) = 0; if (is_present_hf("X-AT-ClientId")) { $var(clientId) = $hdr(X-AT-ClientId); } } Any idea, is it a bug or I do something wrong in config files? Peter From: Dr. Barabás Péter via sr-users <sr-users(a)lists.kamailio.org> Date: Sunday, 2023. October 8. 18:49 To: miconda(a)gmail.com <miconda(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com> Subject: [SR-Users] Re: uac_req_send + evroute + crash Hello Daniel, I do not see any “failed to send request with authentication”. The next CRITICAL log appears after calling uac_req_send(): CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring Now we use kamailio 5.7.2 and no crash has come, but the critical logs above exist. I saw a ticket in github: https://github.com/kamailio/kamailio/issues/3522, may the crash be similar or the same to that? I will make a try next week with downgrading to 5.6.2 and try to reproduce the crash. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Friday, 2023. October 6. 13:55 To: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, do you also get error log messages that include "failed to send request with authentication"? Which CRITICAL log appears when you call uac_req_send()? You pasted a couple of them in the initial email? Cheers, Daniel On 05.10.23 17:32, Dr. Barabás Péter wrote: Hello, I used kamailio version 5.6.2. I refreshed to 5.7.2 today. As I remember it was in kamailio long time ago therefore I could skip evroute route. But this CRITICAL log appears always when I call uac_req_send(). I call it with settings: $uac_req(auser) = $var(username); $uac_req(apasswd) = $var(password); Where username and password are retrived from web service before call. In event_route I got first 401 after 200. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Date: Thursday, 2023. October 5. 17:25 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com><mailto:dr.peter.barabas@gmail.com> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, On 05.10.23 16:34, Dr. Barabás Péter via sr-users wrote: Hi All, I use kamailio In front of Asterisk and kamailio needs to refresh registrations periodically towards Asterisk to ensure the availability of users from Asterisk side. I use uac module and call uac_req_send() for sending REGISTER requests. I set $uac_req(evroute)=1 The event_route[uac:reply] is called fine, but in kamailio logs I see the next lines: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring After some time, kamailio has crashed. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) – ignoring CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 34 ALERT: <core> [main.c:774]: handle_sigs(): child process 1407950 exited by a signal 6 what version of Kamailio are you using? Is Asterisk challenging for authentication? Does it happen every time or seldom? Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com

2 1

Crash of Kamailio when using module http_async_client - function http_async_query, if module "tm.so" is not loaded (Kamailio 5.7.3)
by Chaigneau, Nicolas 10 Jan '24

10 Jan '24

Hello, When using http_async_query function, provided by module http_async_client: https://www.kamailio.org/docs/modules/devel/modules/http_async_client.html#… If module "tm.so" is not loaded, then immediately upon calling function http_async_query, Kamailio crashes: 9(3807541) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 8 0(3807527) ALERT: <core> [main.c:791]: handle_sigs(): child process 3807533 exited by a signal 11 0(3807527) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 0(3807527) INFO: <core> [main.c:818]: handle_sigs(): terminating due to SIGCHLD Of course, I now know that I must load the module "tm.so". But I think it would be better to catch the issue when Kamailio starts, rather than crashing later on when trying to call http_async_query. What do you think ? 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.

3 3

How to use uac.reg_add command?
by mm e 06 Jan '24

06 Jan '24

./kamcmd uac.reg_add 789798 789798 mydomain.com 789798 myteldomain.com . 789798 123456 . sip:sip6.telco.nl 360 0 10 mydomain.com:5060 udp:mydomain.com:5060 error: 400 - Invalid Parameters Can you provide me with a case, I don't know how to enter the correct format.

2 1

DB invalid connection parameter
by Sergiu Pojoga 04 Jan '24

04 Jan '24

Hi, Facing the following error after bumping Kam from 5.3 => 5.4 => 5.5 => 5.6, following instructions on each step about DB structure and module changes. These errors show up since. Any clue what they mean? A couple other modules are using the same DB MySQL connector. The data in the DB also seems to get updated properly. Jan 03 15:08:30 sip kamailio[20881]: ERROR: <core> [db.c:489]: db_use_table(): invalid connection parameter *Jan 03 15:08:30 sip kamailio[20881]: ERROR: usrloc [ucontact.c:1170]: db_update_ucontact_ruid(): sql use_table failedJan 03 15:08:30 sip kamailio[20881]: ERROR: usrloc [urecord.c:438]: wb_timer(): updating contact in db failed (aor: user(a)domain.com <user(a)domain.com>)* version: kamailio 5.6.5 (x86_64/linux) 48d48f compiled on 02:32:48 Jan 2 2024 with gcc 4.9.2 DB structure: location | CREATE TABLE `location` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `ruid` varchar(64) NOT NULL DEFAULT '', `username` varchar(64) NOT NULL DEFAULT '', `domain` varchar(64) DEFAULT NULL, `contact` varchar(512) NOT NULL DEFAULT '', `received` varchar(128) DEFAULT NULL, `path` varchar(512) DEFAULT NULL, `expires` datetime NOT NULL DEFAULT '2030-05-28 21:32:15', `q` float(10,2) NOT NULL DEFAULT 1.00, `callid` varchar(255) NOT NULL DEFAULT 'Default-Call-ID', `cseq` int(11) NOT NULL DEFAULT 1, `last_modified` datetime NOT NULL DEFAULT '2000-01-01 00:00:01', `flags` int(11) NOT NULL DEFAULT 0, `cflags` int(11) NOT NULL DEFAULT 0, `user_agent` varchar(255) NOT NULL DEFAULT '', `socket` varchar(64) DEFAULT NULL, `methods` int(11) DEFAULT NULL, `instance` varchar(255) DEFAULT NULL, `reg_id` int(11) NOT NULL DEFAULT 0, `server_id` int(11) NOT NULL DEFAULT 0, `connection_id` int(11) NOT NULL DEFAULT 0, `keepalive` int(11) NOT NULL DEFAULT 0, `partition` int(11) NOT NULL DEFAULT 0, PRIMARY KEY (`id`), UNIQUE KEY `ruid_idx` (`ruid`), KEY `account_contact_idx` (`username`,`domain`,`contact`), KEY `expires_idx` (`expires`), KEY `connection_idx` (`server_id`,`connection_id`), *KEY `tcpcon_idx` (`connection_id`)* ) ENGINE=InnoDB AUTO_INCREMENT=287989 DEFAULT CHARSET=latin1 > select * from version where table_name='location'; table_name: location table_version: 9 Thanks.

2 2

webRTC Test Client?
by Andy Newlands 03 Jan '24

03 Jan '24

Hi, I have a Freeswitch installation, fronted by a Kamailio proxy I would like to configure Kamailio to bridge SIP between webRTC clients and Freeswitch. The first issue I hit is how to set up a test webRTC client to make calls into Kamailio/FS. Can anyone point me to a simple webRTC client I can use to make test webRTC/SIP calls into my setup? Thanks BTW: I'm not JS experienced (I work with Python, C#, C/C++). Kind regards, Andy

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Kamailio - Initiating invite on REFER
by Muhammad Danish Moosa 02 Jan '24

02 Jan '24

Hi champs, In this flow like endpoint -> B2BUA 1-> kamailio -> B2BUA 2, there is a requirement that REFER should be handled by Kamailio and it should manage the transfer , initiating a new invite based on REFER. Any pointer will be helpful -- Danish

3 2

Connect to a remote ENUM(E.164 Number Mapping) server
by elhar.mohamed＠gmail.com 25 Dec '23

25 Dec '23

Hello, I'm using the Kamailio version 5.7.x, I didn't find in the module "enum.so" any option that allow to connect to a remote ENUM server. Could you please help me ! Thanks in advance

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Merry Christmas and Happy Holidays 2023-2024!
by Daniel-Constantin Mierla 25 Dec '23

25 Dec '23

Kamailio project continued to evolve steadily, a new major release was out during this year, next one planned in a few months, relying on an amazing community of developers and users. We hope it kept helping 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 2023 another fantastic year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services Kamailio World Conference - April 18-19, 2024, Berlin

2 1

kemi python3s - module 'KSR' has no attribute 'nathelper'
by marek 22 Dec '23

22 Dec '23

hi, trying kamailio 5.7.3 with KEMI pythons3s sample from https://github.com/kamailio/kamailio/blob/master/misc/examples/kemi/kamaili… trying simple register with sipsak sipsak -U --sip-uri "sip:cervajs@kamailio.example.com" -u cervajs -a top_secret 4(28805) ERROR: PY8 {REGISTER}: app_python3s [apy3s_kemi.c:141]: apy3s_exec_func(): error exception occurred 4(28805) ERROR: PY8 {REGISTER}: app_python3s [apy3s_exception.c:167]: apy3s_handle_exception(): apy3s_exec_func: ksr_request_route((null)): Unhandled exception in the Python code: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 31, in ksr_request_route if ksr_route_natdetect()==-255 : File "/etc/kamailio/kamailio.py", line 238, in ksr_route_natdetect if KSR.nathelper.nat_uac_test(19)>0 : AttributeError: module 'KSR' has no attribute 'nathelper' i see nat_uac_test in doc https://kamailio.org/docs/tutorials/5.7.x/kamailio-kemi-framework/modules/#… any tips/ideas? happy christmas Marek

1 1

Errors with TLS
by SAMUEL MOYA TINOCO 21 Dec '23

21 Dec '23

Hi everyone, All my environment was working properly. But a few days ago I've started to have problems with my websocket connectios. Everytime I try to register my websocket Kamailio log this Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1330]: tls_h_read_f(): protocol level error Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_util.h:50]: tls_err_ret(): TLS accept:error:0A000416:SSL routines::sslv3 alert certificate unknown (sni: unknown) Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1334]: tls_h_read_f(): src addr: IP_webclient:61423 Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1337]: tls_h_read_f(): dst addr: IP_Kamailio:8443 Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: <core> [core/tcp_read.c:1476]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fbc423053b0 r: 0x7fbc423054d8 (-1) My tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes private_key = /etc/kamailio/cert/clave_SSL.key certificate = /etc/kamailio/cert/certificado.pem Kamailio version: root@prod-kamailio:/etc/kamailio# kamailio -v version: kamailio 5.6.3 (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 12.2.0 I don't know why it stopped working. And I'm not sure what to do or what to check. Has anyone faced this problem? Any help would be appreciated Thanks in advance Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DA33E6.D2D833D0] Soluciones inteligentes para la autonomía personal

2 2

How to hang up the call in xhttp routing
by mm e 21 Dec '23

21 Dec '23

event_route[xhttp:request] { if ($hu =~ "hangupCall"){ $var(body) = $rb; xhttp_reply("200", "OK", "",""); jansson_get_field($var(body), "callid", "$var(callid)"); jansson_get_field($var(body), "cseq", "$var(cseq)"); if (t_cancel_callid("$var(callid)", "$var(cseq)", "22")) { xlog("L_NOTICE", "t_cancel_callid============"); } exit; } xhttp_reply("404", "Not Found", "",""); exit; } According to this route, the answered call cannot be hung up。How should I hang up the answered call in xhttp routing, similar to freeswitch's uuid_kill。

1 0

Kamailio - Ms Teams direct Routing - dispatcher issues | FLAGS: IP
by faisal.jamil.khan＠gmail.com 19 Dec '23

19 Dec '23

Hi Team, I am trying to setup Kamailio with MS Teams Direct Routing and after resolving TLS issues now I am getting this at my dispatcher flags. Possible suggestions required. kamcmd dispatcher.list | egrep "URI|FLAGS" URI: sip:sip.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP URI: sip:sip2.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP URI: sip:sip3.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP Dec 14 20:23:27 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKd9f6.c2e7d2f4000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-c7b52d5e CSeq: 10 OPTIONS Call-ID: 23b2dc436850e949-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux)) Dec 14 20:23:28 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip2.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKccf6.53fea6f0000000000000000000000000.0 To: <sip:sip2.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-7a126c3a CSeq: 10 OPTIONS Call-ID: 23b2dc436850e94a-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux)) Dec 14 20:23:28 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip3.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKdcf6.5a4f89e6000000000000000000000000.0 To: <sip:sip3.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-992aa610 CSeq: 10 OPTIONS Call-ID: 23b2dc436850e94b-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux))

3 33

Info: Next Kamailio World Conference - April 18-19, 2024, Berlin, Germany
by Daniel-Constantin Mierla 18 Dec '23

18 Dec '23

Hello, a short announcement to inform that the dates for next Kamailio World Conference have been set to April 18-19, 2024, the event to take place again in Berlin, Germany. It will be a different venue than the past editions, that one not being available. A pre-conference day is considered on April 17, 2024, but it is not yet known in what form and where. The conference is scheduled a bit earlier than the usual in the year because Germany hosts the Euro 2024 football championship during June-July 2024, making May very busy and harder to find adequate conferencing space in Berlin. The website and more details will be published soon. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

Regretfully we were unable to process the URI
by Gavin Baumanis 18 Dec '23

18 Dec '23

Hi Everyone, Very new to SIP And to Kamailio Configuration / administration. I have a working AWS instance with Kamailio installed on it - and it works fine. We are creating a whole new environment and I have cloned the existing AWS EC2 instances. I have edited all the configs to ensure that they are using the correct DNS names and / or IP addresses. Otherwise the "configs" are identical. While attempting to call a SIP number, I get the following message, returned to the soft phone I am using (microSIP) *Regretfully we were unable to process the URI (479/SL)* I can also see the same error message in the console of the Kamailio server. I am hoping that this might mean something to someone enough - to suggest some troubleshooting steps, please. Thanks;

3 3

Encoded Call-ID - Topoh module
by Vanderlei Torres Batistela 15 Dec '23

15 Dec '23

Hello all, How can I get the encoded Call-Id generated by TOPOH module to store in the CDR ? Regards, Vanderlei

2 1

{uri.duri} not working
by hanxiaofang＠x-y-t.cn 13 Dec '23

13 Dec '23

Hi Kamailio experts, OS: Debian11 Kamailio ver: 5.7.3 {uri.duri} doesn't seem to work The following script can be reproduced: ``` debug=2 log_stderror=yes # mpath="/usr/local/lib/kamailio/modules/" loadmodule "tm.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "ctl.so" loadmodule "htable.so" modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) request_route { xinfo("*** $sel(msg)\n"); sl_send_reply("404", "Not here"); exit; } event_route[htable:mod-init] { $var(ouri) = "sip:alice@server.com:5060;nat=yes;transport=tcp;line=xyz"; $var(duri) = $(var(ouri){uri.duri}); # => "sip:server.com:5060;transport=tcp" xinfo("ouri = $var(ouri)\n"); xinfo("duri = $var(duri)\n"); if ($var(duri) == "sip:server.com:5060;transport=tcp" ) { xinfo("uri.duri work well\n"); } else { xerr("something is wrong\n"); } } ```

2 4

Proxy Inactivity and Kamailio Log Disruption Issue.
by satyaprakash ch 12 Dec '23

12 Dec '23

HI, Scenario: We possess 2 proxies and 2 registrar servers equipped with dmq functionality. Unexpectedly, one of the Proxy IP addresses was marked as inactive (FLAG: IP) in the dispatcher list, causing the proxy to cease responding. Simultaneously, Kamailio logs stopped recording any activity. Upon restarting the Kamailio service, the problem was resolved, and the proxy resumed normal operation, with the Flag changing back to AP (ACTIVE). The sudden occurrence of this issue is not clear, and due to the cessation of logs during the incident, we lack information for the investigation. Any suggestions on how to address this issue would be appreciated. And is this scenario happened to any one in your experience?

2 3

usrloc ka_mode - Flooding OPTIONS at same time?
by Benoit Panizzon 12 Dec '23

12 Dec '23

Hi We have reached 20000 registered CPE and start facing issues. Am I observing correctly that when ka_mode 4 is enabled, OPTIONS are send simultaneously to all registered CPE and not spread over the interval? They could be the cause very high pps peaks on network equipment? If so, is there a way to spread the ka options over time so they don't cause such high peaks? Or even better, is there a way to enable them for only some customers via a variable? 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 ______________________________________________________

4 5

is_method_f(): cannot parse cseq header in xhttp:request event route?
by Benoît Panizzon 12 Dec '23

12 Dec '23

Hi all I use event_route[xhttp:request] on Kamailio 5.5 Every time a xhttp request is processed I get this log entry: ERROR: textops [textops.c:3541]: is_method_f(): cannot parse cseq header Almost identical config on Kamailio 5.6 does not show this error. Is that something that got fixed in 5.6? I see a change in the documentation, While 5.5 does not state $rm can be accessed, 5.6 states $rm contains the HTTP method. Or do I somehow leak the http request to somewhere I call is_method() on it after terminating the event route? How should the event_route[xhttp:request] properly be terminated? With 'return' or with 'exit'? -- 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 ______________________________________________________

1 0

Upgrade Minor versions
by Gavin Baumanis 11 Dec '23

11 Dec '23

Hi there, I have taken over a project that uses Kamailio. I am aware of what we use it for - I am just not sure on the how... apart from what I have read in the docs. We're currently running 5.4.2 and I'd like to upgrade to (at least) the latest 5.4 offering - as that seems as if it would be the easiest and safest option. So I am hoping I might get some direction, please - on the way forward for an upgrade. The documentation seems OK if you're upgrading minor versions... but I couldn;t find anything about upgrading patch versions. Thanks!

2 2

Updated: Online devel meeting on matrix - 15:00 UTC, Dec 5, 2023
by Daniel-Constantin Mierla 05 Dec '23

05 Dec '23

Hello, we should consider an online devel meeting sometime soon to summarize what was done at (and still needs to be done after) devel meeting in Dusseldorf and plan a bit the targets for next major release (5.8 or 6.0)? If considered useful, I propose Dec 5 at 15:00UTC (16:00 Berlin/Paris/Madrid/Rome), but we can also look for other dates as well. Topics to be discussed can be added at: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Pull requests can be made by users without git access. Cheers, Daniel PS: Previous announcement indicated the start time 14:30 UTC, this message changes it to 15:00 UTC - 30min later due to a constraint in availability of one of the Kamailio developers. -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 3

stats.get_statistucs dialog: counter over/underrun?
by Benoit Panizzon 05 Dec '23

05 Dec '23

Hi List Kamailio 5.5 # kamcmd stats.get_statistics dialog: dialog:active_dialogs = 17 dialog:early_dialogs = 18446744073709551615 dialog:expired_dialogs = 0 dialog:failed_dialogs = 31198 dialog:processed_dialogs = 107519 Where could this early_dialog value come from? 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 ______________________________________________________

1 0

Setting up Kamailio as SIP Proxy
by Saul Ibarra 04 Dec '23

04 Dec '23

Hi, I am needing some help setting up kamailio as a sip proxy between a provider(trunk) and a test pbx The trunk requires username&password and i would like to add that information to the request within the Kamailio proxy, so that the PBX can authenticate directly by IP (without username&password) through the Kamailio proxy. As a programmer with a limited background in SIP (currently learning), I am keen on understanding the intricacies of this setup at the SIP level. Could you please direct me to resources or documentation that covers scenarios similar to mine? Any help or guidance you could provide would be greatly appreciated. Thanks

4 8

tsilo dependency on usrloc
by Alex Balashov 03 Dec '23

03 Dec '23

Hi, I wanted to revisit the topic of tsilo dependence on the location service. All three ts_append*() functions have the following quality: - ts_append(): "performing a contact lookup on the table specified by the domain parameter." - ts_append_by_contact(): "the contact lookup is performed" - ts_append_to(): "performing a contacts lookup on the table specified by the domain parameter" Why is this extensive coupling to usrloc necessary? This makes it impossible to use `tsilo` in case of providing a push-notification add-on that front-ends an upstream registrar, requiring a kind of local shadow registrar or mid-registrar. What would make more sense is a generic mechanism that allows one to "drip" new contacts into an existing transaction, whether suspended or active. Kamailio has a mechanism to add more branches to an existing transaction, but the scope of that mechanism is only from *inside* the vantage point of the transaction in question. The key parlour trick of `tsilo` is that it permits dripping new branches into a *different* transaction. ts_append_to() almost does the trick, providing a target index and label, but it just insists on doing a registrar lookup to source the contacts. What is really wanted and needed for the downstream PN gateway use-case is a means of extracting contacts from incoming registrations (or other sources, potentially) without storing them in any fashion locally, without using or even loading usrloc, and just throwing them over the fence into a different transaction. Is this somehow possible by means other than tsilo? Am I overlooking something? Cheers, -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 3

Unable to use db_alias_lookup to call multiple extensions
by Merul Patel, PhD 01 Dec '23

01 Dec '23

Hi, Kamailio newby here. Having some trouble with using alias_db to forward an incoming call to two extensions. Thanks in advance for any help. I've tried using the Ubuntu Jammy ARM64 packages, as well as compiling the source for version 5.7.3. In both cases, I've added three extensions, and calls directly between them work fine with the vanilla kamailio.cfg. However, I then wanted to add an alias so that when called it would dial two extensions, but only the first extension is ever called. My kamailio.cfg only differs in a few places: $ diff kamailio.cfg kamailio.cfg.original 3,6d2 < #!define WITH_AUTH 1 < #!define WITH_DEBUG 1 < #!define WITH_ALIASDB 1 < # 268,270d263 < loadmodule "db_sqlite.so" < #!define DBURL "sqlite:///usr/local/kamailio-devel/etc/kamailio/kamailio.sqlite" < 367c360 < modparam("tm", "failure_reply_mode", 0) --- > modparam("tm", "failure_reply_mode", 3) 450d442 < modparam("alias_db", "append_branches", 1) I can see from the log that when dialling the desired alias (5000) an alias_db query is performed: Dec 01 20:00:23 merrivale /usr/local/kamailio-devel/sbin/kamailio[3264]: DEBUG: {1 2 INVITE 651794328853009-230332414159388(a)192.168.17.145} alias_db [alookup.c:169]: alias_db_query(): new URI [0] is [sip:6002@merrivale.localdomain] Dec 01 20:00:23 merrivale /usr/local/kamailio-devel/sbin/kamailio[3264]: DEBUG: {1 2 INVITE 651794328853009-230332414159388(a)192.168.17.145} alias_db [alookup.c:169]: alias_db_query(): new URI [1] is [sip:6001@merrivale.localdomain] But only the first result is called. I've changed the order of the aliases in the sqlite db and just the first result is called. What am I missing? Merul

1 0

How to use: kamcmd ul.db_users?
by Benoit Panizzon 01 Dec '23

01 Dec '23

Hi List I want to find a way to check how many users are regsitered. https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.r.db… But no matter how I try, I get a 500 error... # kamcmd ul.db_users location And yes, the table is called 'location'. I did try to find out how kamctl ul show --brief pulls that information, but did not succeed. Also that command takes way too long how that we have several thousand registered users. Or would I need to query this directly from the database? 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 ______________________________________________________

2 2

Info: RTC Devroom at FOSDEM 2024
by Daniel-Constantin Mierla 01 Dec '23

01 Dec '23

Hello, another edition of FOSDEM is approaching and it will include again a Real Time Communications devroom. If anyone wants to share their interesting applications or experiences in the field, consider to submit a presentation proposal. The CfP is available at: - https://lists.fosdem.org/pipermail/fosdem/2023q4/003518.html It does not have to be about Kamailio, all RTC topics are welcome, just think to be something useful for developers, as the event is targeting them. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

No access to variables within timer route
by dries＠degendt.com 30 Nov '23

30 Nov '23

Hi, I have a timer which is called each 5 seconds: modparam("timer", "declare_timer", "REFRESH_CALLSTATE_TIMER=REFRESH_CALLSTATE_TIMER,5000,slow"); I'm enabling setting the variable "testvar" and enabling the timer when an INVITE is received: if (is_method("INVITE|ACK|UPDATE|CANCEL|BYE")) { $var(testvar) = "blablabla"; timer_enable("REFRESH_CALLSTATE_TIMER", "1"); ... route[REFRESH_CALLSTATE_TIMER] { xlog("L_INFO", "Testvariable: $var(test)\n"); } However, the variable is always 0: INFO: <script>: Testvariable: 0 I've also tried printing pseudovariables like $ci, $du but nothinh seems accessible. Is there a workaround for this? Best regards, Dries

3 6

kamailio t_on_failure tcp timeout
by Julien Chavanton 29 Nov '23

29 Nov '23

Hi, I would like to catch TCP timeouts when I use a socket that was not closed. In this example I am sending SIP MESSAGE ... t_on_reply("ON_REPLY_MESSAGE"); t_on_failure("FAILURE_MESSAGE"); if (!t_relay()) sl_reply_error();

1 2

Evapi Async - suspend transaction for a limited time
by Ilie Soltanici 29 Nov '23

29 Nov '23

Hello, I'm trying to use the evapi_async_relay function from the evapi module, however,I don't want the transaction to be suspended for more than 1 second. For instance, if there's no response from the external application within this time frame, I want the script to continue and remove that transaction from memory. One potential approach I am considering is to add suspended transactions to a hash table with a timestamp value. Then, using rtimer module, I could periodically parse this table every second and process the results, but I'm not sure how efficient is that. Maybe you know other alternative, more efficient solutions (from a performance point of view) that could achieve the same goal? Any insights or recommendations would be greatly appreciated. Thank you.

3 6

5.6.4 debian htable dbmode=1 not writing at shutdown
by Bernd Krueger-Knauber 29 Nov '23

29 Nov '23

Hi, I have still the problem that my htables are not written into a sqlite db when I shutdown kamailio. Manual writing via: kamcmd htable.store ... is working. As I can see in the logs with highest loglevel, the function destroy() is never called. ht_db_url.len is > 0 ht_db_init_con() should work, else nothing would work with the db. But I can not see the debug messages from ht_db_open_con() Any ideas? Best regards, Bernd

5 5

Incompatibility of the DMQ htable protocol with older Kamailio versions
by Denys Pozniak 29 Nov '23

29 Nov '23

Hello! I noticed that there are differences in the DMQ htable replication protocol for the old version of Kamailio 5.2 and the new 5.7. For example, the new version uses action 2 when updating values and action 4 to remove. The old Kamailio version uses the values 1 and 3 respectively. Well, as a result, the htable data is not shared between such nodes, plus there are errors in the syslog of the old version: ERROR: htable [ht_dmq.c:328]: ht_dmq_handle_msg(): failed to replay action It would probably be worth introducing some kind of DMQ htable protocol versioning to maintain compatibility? And is there any option to make them work together (I wouldn’t really want to change the action number in sip/sdp/json on the fly)? Kamialio 5.2 {"action":1,"htname":"shared","cname":"user","type":2,"strval":"uri","mode":1} {"action":3,"htname":"shared","cname":"user","mode":0} Kamailio 5.7 {"action":2,"htname":"shared","cname":"user","type":2,"strval":"uri","mode":1} {"action":4,"htname":"shared","cname":"user","mode":0} -- BR, Denys Pozniak

2 2

tps_storage_fill_contact(): no r-uri user
by Denys Pozniak 28 Nov '23

28 Nov '23

Hello! My topology is like this: PBX -> Kamailio Topos -> User When calling the user, PBX does not add the user part in the Contact URI, something like: Contact: <sip:10.10.10.10:5080;transport=udp> When the customer sends a reinvite, the topos turns off for this transaction and the sip partially breaks down with error: topos [tps_storage.c:319]: tps_storage_fill_contact(): no r-uri user topos [tps_storage.c:615]: tps_storage_record(): failed to store As far as I understand, the RFC allows the absence of the user part in the URI and this occurs quite often. I can provide the pcap file and debug log in a private message if this helps identify the essence of the problem. # kamailio -v version: kamailio 5.7.1 (x86_64/linux) 1cf389-dirty -- BR, Denys Pozniak

3 5

Kamailio ver 5.7.3 - source installation warnings
by Muhammad Danish Moosa 28 Nov '23

28 Nov '23

Hi, I am planning to upgrade Kamailio to version 5.7.3 to manage some memory leaks issues. I am installing from source and came across these warnings, just wanted to be sure if they are safe to be ignored. cr_func.c: In function ‘set_next_domain_on_rule’: cr_func.c:160:2: warning: missing braces around initializer [-Wmissing-braces] pv_value_t val = {0}; ^ cr_func.c:160:2: warning: (near initialization for ‘val.rs’) [-Wmissing-braces] cr_func.c: In function ‘actually_rewrite’: cr_func.c:348:2: warning: missing braces around initializer [-Wmissing-braces] pv_value_t val = {0}; ^ cr_func.c:348:2: warning: (near initialization for ‘val.rs’) [-Wmissing-braces] cr_func.c: In function ‘ki_cr_load_user_carrier_helper’: cr_func.c:782:2: warning: missing braces around initializer [-Wmissing-braces] pv_value_t val = {0}; ^ -- LD (gcc) [M p_usrloc.so] p_usrloc.so CC (gcc) [M evrexec.so] evrexec_mod.o evrexec_mod.c:64:1: warning: missing braces around initializer [-Wmissing-braces] static evrexec_info_t _evrexec_info = {0}; ^ evrexec_mod.c:64:1: warning: (near initialization for ‘_evrexec_info.data’) [-Wmissing-braces] LD (gcc) [M evrexec.so] evrexec.so CC (gcc) [M avp.so] avp.o LD (gcc) [M avp.so] avp.so CC (gcc) [M acc_diameter.so] diam_message.o CC (gcc) [M acc_diameter.so] acc_diameter_mod.o CC (gcc) [M acc_diameter.so] diam_avp.o CC (gcc) [M acc_diameter.so] diam_tcp.o LD (gcc) [M acc_diameter.so] acc_diameter.so CC (gcc) [M dmq.so] bind_dmq.o CC (gcc) [M dmq.so] message.o CC (gcc) [M dmq.so] dmq_funcs.o dmq_funcs.c: In function ‘ki_dmq_send_message’: dmq_funcs.c:304:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:304:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] dmq_funcs.c: In function ‘ki_dmq_bcast_message’: dmq_funcs.c:374:3: warning: missing braces around initializer [-Wmissing-braces] dmq_peer_t new_peer = {0}; ^ dmq_funcs.c:374:3: warning: (near initialization for ‘new_peer.peer_id’) [-Wmissing-braces] CC (gcc) [M dmq.so] notification_peer.o CC (gcc) [M dmq.so] dmq.o dmq.c:61:1: warning: missing braces around initializer [-Wmissing-braces] sip_uri_t dmq_server_uri = {0}; ^ dmq.c:61:1: warning: (near initialization for ‘dmq_server_uri.user’) [-Wmissing-braces] dmq.c:67:1: warning: missing braces around initializer [-Wmissing-braces] static sip_uri_t dmq_notification_uri = {0}; ^ dmq.c:67:1: warning: (near initialization for ‘dmq_notification_uri.user’) [-Wmissing-braces] Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 2

Info: Issue and PR automatic management
by Daniel-Constantin Mierla 27 Nov '23

27 Nov '23

Hello, during the Kamailio Development Meeting that took place in Dusseldorf earlier this month, one topic was related to administrative tasks related to project development and management, how to simplify/automatize such tasks. To reduce the work load on volunteering contributors, GitHub Actions were already used for various tasks related to project development and management (e.g., automatic builds on commits and pull requests to detect compile errors or code formatting mistakes). In Dusseldorf another task was configured to be managed with GitHub Actions, respectively the check of open issues and pull requests to evaluate the interest of submitters, developers and community users to pursue them. If there is no activity on an issue (potential bug or feature request) or a pull request, after 6 weeks it is marked with the label `stale`. After two more weeks of no activity, the issue or the pull request is marked with the label `expired` and closed. Note that any comment postpones the expire timeline, being considered that there is interest in pursuing the issue or the pull request. Requests for features were already treated in this way: if nobody commits to implement it, it can be closed after one month, but it needed manual work and many were still kept open. Potential bug reports that become very old are hard to tackle if the source code changes or new major releases are out, they might even not be valid anymore. Anyhow, this automatic operations can be reverted if there is still interest in pursuing the specific issues or pull requests. A registered developer can remove labels and reopen a closed issue or pull request. The non-registered-developer contributors have to make a comment that includes the token `/notstale` to remove the label `stale` or includes the token `/notexpired` to reopen a closed item. This new kind of automatic task management might add a little inconvenience because one has to restate the interest from time to time for those items that could not be addressed. However, considering that Kamailio is an open source collaborative project, in order to be fair for those that volunteer to spend time and resources for development of Kamailio, also the users/submitters have to stay engaged, not just report and forget about. The process to automatize tasks related to Kamailio development and administration is work in progress. Everything can be adjusted based on feedback (e.g., time lines), feel free to suggest improvements or new solutions to make things easier for everyone within the project ecosystem. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

How to remove record-route and via from a forwarded call?
by Benoit Panizzon 27 Nov '23

27 Nov '23

Hi List Two Kamailio Nodes situation. Node A: Routing Instance. Node B: Registrar Instance. An invite is sent from Node A to B. Customer registered on B is 'busy' as example. B initiates Call Forwarding by adding a Diversion Header and sending the Invite back to A with a new R-URI towards the CFB destination. The Invite sent from B to A still has Node B in it's Record-Route. So all subsequent SIP messages pass via B, which is not needed and cause issue with rtpengine running on B. I attempted to remove_record_route(); when sending the call back to A, but this causes: parse_headers(): bad header field [;lr;ftag=3910078620-883101371> It looks like only the beginning of the RR header is removed leaving back a competely broken line. What is the propper was to remove a node from RR and Via routing? 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 4

WEBRTC-to-SIP & Vice Versa
by Vishal Pai 26 Nov '23

26 Nov '23

Hello All Good Day Here I am trying to achieve something like this WSS to UDP Ext registartion which works well using https://github.com/havfo/WEBRTC-to-SIP/tree/master/etc Note here i am forwarding the registartion to asterisk pbx server and define WSS of kamailio Now for calls - Webrtc - SIP calls works - SIP - Webrtc calls not working Here I am getting the request of the call on kamailio but kamailio send 404 not found to Asterisk PBX Webrtc - Webrtc calls not working Can anyone guide me on what I am doing wrong? Thanks Vishal P

3 3

refer-to to new INVITE
by Social Boh 26 Nov '23

26 Nov '23

Hello, I'd like to know if there is a way, from Kamailio side, take the refer-to header Kamailio receive in a REFER request and create a new INVITE whit this data; then bridge the call between the originating user and the new destination user. Thank you Regards -- --- I'm SoCIaL, MayBe

3 4

Re: RADIUS - Can't use extra attributes
by Duarte Rocha 24 Nov '23

24 Nov '23

Hello Daniel, Thanks for the reply. The extra dictionary is already included otherwise an error is thrown and Kamailio doesn't start. However, i realized my mistake now and you were right. I was checking for the extra attributes in a tcpdump and on radius server and i didn't had the dictionary configured on both of them. The extra attributes are in the packet but weren't being listed. One more question. Kamailio doesn't have a way to do a totally custom RADIUS access-request packet right? (Sorry if this opens another thread. I usually just hit reply on the mailing list and let Outlook create the e-mail, but it usually creates another thread and i'm not sure why). Thanks again. Cheers, Duarte

1 0

Re: RADIUS - Can't use extra attributes
by Duarte Rocha 24 Nov '23

24 Nov '23

Hello Daniel, Thanks for the reply. The extra dictionary is already included otherwise an error is thrown and Kamailio doesn't start. However, i realized my mistake now and you were right. I was checking for the extra attributes in a tcpdump and on radius server and i didn't had the dictionary configured on both of them. The extra attributes are in the packet but weren't being listed. One more question. Kamailio doesn't have a way to do a totally custom RADIUS access-request packet right? (Sorry if this opens another thread. I usually just hit reply on the mailing list and let Outlook create the e-mail, but it usually creates another thread and i'm not sure why). Thanks again. Cheers, Duarte

1 0

Max-Forwards header missing in OPTIONS keep-alive messages
by Christian Schnell 24 Nov '23

24 Nov '23

Hello, in RFC 3261 (sec. 8.1.1.6) it states about the Max-Forwards header field: > A UAC MUST insert a Max-Forwards header field into each request it originates with a value that SHOULD be 70. > > Source: https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1.6 When Kamailio sends its OPTIONS keep-alive requests (from the nathelper module I believe), Max-Forwards is missing: > OPTIONS sip:alice@example.com:61266 SIP/2.0 > Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK5124450 > From: sip:pinger@sip.example.com;tag=uloc-6550aa14-7456-2461-6390f173-5fdb4667 > To: sip:alice@example.com:61266 > Call-ID: aba25282-2ca72be1-9d1fd53(a)aa.bb.cc.dd > CSeq: 1 OPTIONS > Content-Length: 0 The user agent library I'm using (libre from baresip) rejects these requests with a "400 Bad Request" response, I looked at the sources and it denies these requests because of the missing Max-Forwards header. Of course this suffices for the keep-alive logic because all that's needed is any traffic in both directions to keep both (potential) NATs open. However you might agree that it does look a bit confusing at first, and the RFC is clear about that. So it's not a bug, but is this behavior intended? Thanks, also thanks for all the great work that has been put into Kamailio. Christian

3 3

error sending message to heplify server.
by masoud96＠mail.ru 24 Nov '23

24 Nov '23

Hello, I have an error when sending a message to the heplify server: Nov 23 15:09:59 sip1-life3 kamailio[1045006]: CRITICAL: {2 1395 INVITE ad25bc92-16a8-4be6-a29f-a4e4ee3a2c0c} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045006]: ERROR: {2 1395 INVITE ad25bc92-16a8-4be6-a29f-a4e4ee3a2c0c} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044995]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549fc5f0, len: 445, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044995]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044995]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f2154948b18, len: 1511, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044994]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044995]: ERROR: {2 28117 BYE 65297c88-0834-44ce-a4da-cfd7012c2b85} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f2154979898, len: 485, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044995]: CRITICAL: {2 28117 BYE 65297c88-0834-44ce-a4da-cfd7012c2b85} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044995]: ERROR: {2 28117 BYE 65297c88-0834-44ce-a4da-cfd7012c2b85} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549d9c50, len: 532, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044994]: CRITICAL: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549d9c80, len: 560, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044994]: CRITICAL: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549d9c40, len: 529, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044994]: CRITICAL: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044994]: ERROR: {1 1 INVITE yjaewdcbicjidajxj0td800z8ba3ta09(a)10.18.5.64} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044996]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f215494b6a8, len: 823, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044996]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044996]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1045015]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549fc5a0, len: 507, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1045015]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045015]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1044996]: ERROR: {2 16492 INVITE sOPq2dosBQcFpqitxVMlyLxFiuWNLZyk} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549fc158, len: 1037, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1044996]: CRITICAL: {2 16492 INVITE sOPq2dosBQcFpqitxVMlyLxFiuWNLZyk} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1044996]: ERROR: {2 16492 INVITE sOPq2dosBQcFpqitxVMlyLxFiuWNLZyk} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f2154978c18, len: 506, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1045009]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1045195]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f2154948e00, len: 567, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1045195]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045195]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: {2 6972 INVITE 8f4b30bb-01fa-462d-ab66-8b30c0f7b78e} <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f2154948fe8, len: 488, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1045009]: CRITICAL: {2 6972 INVITE 8f4b30bb-01fa-462d-ab66-8b30c0f7b78e} siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: {2 6972 INVITE 8f4b30bb-01fa-462d-ab66-8b30c0f7b78e} siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: <core> [core/udp_server.c:697]: udp_send(): sendto(sock, buf: 0x7f21549fc158, len: 542, 0, dst: (10.241.60.115:9060), tolen: 16) - err: Bad file descriptor (9) Nov 23 15:09:59 sip1-life3 kamailio[1045009]: CRITICAL: siptrace [../../core/forward.h:231]: msg_send_buffer(): udp_send failed Nov 23 15:09:59 sip1-life3 kamailio[1045009]: ERROR: siptrace [siptrace_hep.c:229]: trace_send_hep3_duplicate(): cannot send hep duplicate message version: kamailio 5.7.2 (x86_64/linux) 95629c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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: 95629c compiled on 14:30:56 Nov 20 2023 with gcc 10.2.1

2 1

Hump presence mwi htable
by David VILLAUME 24 Nov '23

24 Nov '23

Hello, I'm running presence_mwi without database. I'm looking for a way to dump the presence htable content, is there any way to perform that ? Regards [https://storage.letsignit.com/6009878cd5039f000742d6a1/generated/effects_bf…] David VILLAUME Ingénieur Voix Tel : +33 156377321 david.villaume(a)sewan.fr 2 cité Paradis - 75010 PARIS www.sewan.fr<https://cloud.letsignit.com/collect/bc/649d2ea2264803323682b26f?p=qPDZoXyim…> [https://storage.letsignit.com/6009878cd5039f000742d6a1/generated/effects_78…]<https://cloud.letsignit.com/collect/bc/649d2ea2264803323682b26f?p=qPDZoXyim…> [https://storage.letsignit.com/6009878cd5039f000742d6a1/generated/effects_13…]<https://cloud.letsignit.com/collect/bc/649d2ea2264803323682b26f?p=qPDZoXyim…> [https://storage.letsignit.com/6009878cd5039f000742d6a1/generated/effects_24…]<https://cloud.letsignit.com/collect/bc/649d2ea2264803323682b26f?p=qPDZoXyim…>

2 1

RADIUS - Can't use extra attributes
by duarterocha91＠gmail.com 24 Nov '23

24 Nov '23

Hello all, I'm currently implementing a Kamailio with RADIUS. I want to send an Access-Request to my RADIUS server and in order to do that i'm using misc_radius module and radius_does_uri_exist() function without any arguments : "if (radius_does_uri_exist()) {" For my module configuration I have the following configurations : "modparam("misc_radius", "radius_config", "/etc/radcli/radiusclient.conf") modparam("misc_radius", "uri_extra", "Sip-Method=$rm")" From what i understand, with this config my access-request should contain the default attributes and Sip-Method Attribute but that is not happening. Only defaults attributes are included. I've tried to run Kamailio with "-ddd" flag for better debug and i only get this error : "1(176) DEBUG: misc_radius [functions.c:615]: radius_does_uri_user_host_exist(): failure". Is there any other configuration that i must do? Thanks for the help. Cheers, Duarte

2 1

Online devel meeting on matrix - 14:30 UTC, Dec 5, 2023
by Daniel-Constantin Mierla 24 Nov '23

24 Nov '23

Hello, we should consider an online devel meeting sometime soon to summarize what was done at (and still needs to be done after) devel meeting in Dusseldorf and plan a bit the targets for next major release (5.8 or 6.0)? If considered useful, I propose Dec 5 at 14:30UTC (15:30 Berlin/Paris/Madrid/Rome), but we can also look for other dates as well. Topics to be discussed can be added at: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Pull requests can be made by users without git access. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

sips to sip
by David Villasmil 24 Nov '23

24 Nov '23

Hello guys, I have this setup where one side is TLS and the other UDP. Normally this works fine, but we have this provider sending sips as the schema everywhere (from, to, rr, contacts), kamailio sends the same sips to the upstream usp freeswitch. My problem is when FS sends back a 200OK and kamailio forwards it back to the provider, the provider sends an ACK and kamailio can't match it with the dialog and doesn't know where to forward it. i think this is happening because FS when is sees SIPS is setting the contact port as 5081 instead of the usual 5080... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

5 8

Dialog + DMQ: dlg_ontimeout() triggered on wrong node!
by Benoît Panizzon 23 Nov '23

23 Nov '23

Hello World! Kamailio 5.5 in use. I wonder, how I could prevent this issue. modparam("dialog", "send_bye", 1) modparam("dialog", "default_timeout", 21600) in the corresponding route: $dlg_ctx(timeout_route) = "DIALOG_TIMEOUT"; dlg_manage(); Dialog starts on Node01 and it's variables, status and timer are DMQ synced to Node02 As both nodes have the same information, I guess both arm the timeout trigger and sometimes the wrong node is more trigger friendly. If dlg_ontimeout() is triggered on Node02 then: * No Bye is send, as Node02 is not handling that Dialog. * Dialog is removed from Memory on Node02 and state synced back to Node01 * Dialog is removed from Memory on Node01 too, but NOT from database [*1] * Dialog CDR is never commited on Node01 * Call is never sent to timeout_route And subsequent in dialog messages on Node01 result in: dlg_onroute(): unable to find dialog Did I misconfigure something? Is there a way to make sure the timer is NOT triggered on the node not handling the dialogue? Shall I try by setting the default_timeout and then use a timeout_avp to set that timeout slightly lower AFTER the dialog has started? Or would this be synced to the other node too? I found out, session timer changes are not synced after an initial timer was synced, so you can not extend the session timer after setting a initial timeout. So maybe this is also true for dialog timeout? [*1] this also explains why upon restarting kamailio, the dialog suddenly is back and after timing out again, a CDR is then written with a way too long duration. -- 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 ______________________________________________________

1 3

set_body() adding or replacing existing body?
by Benoit Panizzon 23 Nov '23

23 Nov '23

Hi I'm trying to change the lifetime timer on a KDMQ dialog message before handing it with dmq_handle_message(); if(has_body("application/json") && $fU == 'dialog') { if (jansson_get("lifetime", $rb, "$var(lifetime)")) { xlog("L_INFO", "$cfg(route): DIALOG JSON action: $var(action) state: $var(state) lifetime: $var(lifetime)\n"); $var(new_lifetime) = $var(lifetime) + 60; $var(newrb) = $rb; jansson_set("integer", "lifetime", $var(new_lifetime), "$var(newrb)"); xlog("L_INFO", "$cfg(route): DIALOG JSON NEW: $var(newrb)\n"); set_body("$var(newrb)","application/json"); msg_apply_changes(); } } dmq_handle_message(); exit; The JSON string looks good after that manipulation, it contains the updated lifetime. But dmq_handle_message() does not seem to get the body anymore. So I wonder what set_body() exactly does. Is it replacing the body or appending to it? Is there a way to crank up debug output in the dialog and/or dmq module? 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 ______________________________________________________

1 1

Kamailio v5.6.5 Released
by Daniel-Constantin Mierla 23 Nov '23

23 Nov '23

Hello, Kamailio SIP Server v5.6.5 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.4. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.5. Note that 5.6 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.7, with v5.7.3 being release out of it. For more details about version 5.6.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/11/kamailio-v5-6-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

1 0

Releasing v5.6.5
by Daniel-Constantin Mierla 23 Nov '23

23 Nov '23

Hello, I am considering to release Kamailio v5.6.5 (out of branch 5.6) later this week (likely on Thursday or Friday, Nov 23/24, 2023). 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 and Development Services Kamailio Advanced Training -- asipto.com

1 1

Crash on core/mem/q_malloc.c
by Igor Potjevlesch 22 Nov '23

22 Nov '23

Hello! I'm facing an issue 2/3 times a day with the following errors in the logs: Nov 22 00:09:02 /usr/local/sbin/kamailio[3720]: CRITICAL: <core> [core/mem/q_malloc.c:150]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten(c0c0c000, abcdefed)[0x7f9600b754f8:0x7f9600b75530]! Memory allocator was called from core: core/re.c:430. Fragment marked by core: db_res.c:139. Exec from core/mem/q_malloc.c:391. Nov 22 00:09:02 /usr/local/sbin/kamailio[3720]: CRITICAL: <core> [core/mem/q_malloc.c:155]: qm_debug_check_frag(): BUG: qm: prev. fragm. tail overwritten [0x7f9600b75460:0x7f9600b75498] - fragment marked by tm: t_fwd.c:230 Nov 22 00:09:02 abrt-hook-ccpp: Process 3720 (kamailio) of user 0 killed by SIGABRT - ignoring (repeated crash) Do you have any idea where it could come from? I'm running Kamailio 5.6.4. Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

2 2

Obsoleting unmaintained modules
by Daniel-Constantin Mierla 21 Nov '23

21 Nov '23

Hello, during the Kamailio Developers Meeting 2023 in Dusseldorf that took place last week, it was proposed to obsolete modules that seem to be unmaintained and no activity about them was noticed during the past years. It is quite some overhead in packaging them and trying to keep them compiling when they have external dependencies, therefore such step should spare some resources in the future. The list (see below) was built based on the options of those present at the meeting, now we want to discuss it on the larger communities of developers and users. If you are using any of these modules or you think any of them worth keeping, reply with the names of the modules that you want to be kept. The proposed action is to relocate the obsoleted modules to a new git repository "kamailio-obsolete" to still keep some visibility to them and in the eventually of future interest on any of them, it can be reintroduced in the main repository. Next is the initial list of modules proposed to be considered obsolete: - app_java - app_lua_sr - app_mono - app_python - app_sqlang - auth_identity - call_control - db2_ldap - db2_ops - db_cassandra - db_perldvdb - dnssec - domainpolicy - h350 - mediaproxy - osp - peering - print - print_lib - pua_xmpp - ratelimit - uid_auth_db - uid_avp_db - uid_domain - uid_gflags - uid_uri_db - uri_db - xmpp - xprint Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training -- asipto.com

3 5

UAC DNS SRV timeout
by Sergiu Pojoga 17 Nov '23

17 Nov '23

Hi, Setup: kamailio acting as a UAC needs to register with a server whose outbound proxy is DNS SRV based only, there's no A record on the proxy's FQDN. The provider advertises two SRV records. ;; ANSWER SECTION: _sips._tcp.sip.provider.com. 10 IN SRV 10 50 5061 sip11-sip.provider.com. _sips._tcp.sip.provider.com. 10 IN SRV 15 50 5061 sip10-sip.provider.com. Everything was working fine, until the provider changed the SRV priority, putting a lower priority on the host that is actually down at the moment. In this case, sip11-sip.provider.com is DOWN and sip10-sip.provider.com is UP. Now, what's happening is Kamailio tries to REGISTER with the lower priority host, gets no reply back (TCP connection doesn't get established). It then takes 30 seconds for Kamailio to try the host with the higher priority, registering successfully. And so it goes for every re-register: failure => 30 seconds => register OK. The same happens for INVITES to the downstream: INVITE => timeout => 30 seconds => INVITE (other host 200 OK). I've tried to play with TCP and DNS params, but so far no luck: no matter what I did - it took the same 30 seconds for Kamailio to detect failure and retry the secondary host. Any suggestions will be highly appreciated. Kamailio 5.6.4 (x86_64/linux) 239d0a tcp_connection_lifetime=3604 tcp_max_connections=60 tcp_connect_timeout=10 tcp_keepalive = yes tcp_crlf_ping=yes tcp_keepcnt=3 tcp_keepidle=30 tcp_keepintvl=30 tcp_reuse_port=1 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 use_dns_cache = on use_dns_failover = on dns_srv_loadbalancing = on dns_try_naptr = on dns_try_ipv6=no dns_try_naptr=no dns_use_search_list=no dns_retr_time=1 dns_retr_no=3 Thanks --Sergiu

1 0

auth_check : sip user / authentication user mismatch
by Jawaid Bazyar 17 Nov '23

17 Nov '23

Hi, I am trying to register an endpoint as follows: REGISTER sip:siptest.booger.net SIP/2.0 Via: SIP/2.0/UDP X.X.X.X.76:31114;branch=z9hG4bK223420485;rport From: sip:1720XXXXXXX@siptest.booger.net;tag=962873337 To: sip:1720XXXXXXX@siptest.booger.net Call-ID: 2053687751-32761-1(a)EH.BJI.BAE.HG CSeq: 2067 REGISTER Contact: sip:1720XXXXXXX@X.X.X.X:31114;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-000B82A1C58B>" Authorization: Digest username="654fe87c-67ff-42dd-8d5e-79786a99713c", realm="siptest.booger.net", nonce="ZVaJtmVWiIrsuWc5hkzM3MUUUs8MgRFR", uri=sip:siptest.booger.net, response="119a3b6426cdcfa037bf1825b69139a5", algorithm=MD5 X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2170 1.0.11.64 Supported: path Expires: 120 MAC: 00:0b:82:a1:c5:8b Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Note that the Authorization username differs from the To-From username. This is intentional, as I am testing use of randomly generated credential . This worked with this endpoint directly against Asterisk, for instance. However, registering against Kamailio now, I am getting: Nov 16 21:24:26 vs-rs01 /usr/sbin/kamailio[584855]: DEBUG: {1 2067 REGISTER 2053687751-32761-1(a)EH.BJI.BAE.HG} auth_db [authorize.c:503]: auth_check(): authentication username mismatch with from/to username Even though I have this set in the config: if (!auth_check("$fd", "subscriber", 0)) { // default 1, change to 0 to skip check 0 is supposed to mean no user checks performed. I am doing this because I want to use the complex credentials for authentication, but want the endpoint to tell me the phone number in an INVITE to/from, instead of putting in the extremely long credential. Is there a way to get Kamailio to accept different SIP User ID and Authentication ID fields? Thanks, Jawaid

2 2

pv cache limit is going to be exceeded
by Ali Taher 17 Nov '23

17 Nov '23

Hello all, I'm using the following Lua script to fetch data from Redis, which is in form of list: function split(str, delimiter) for token in string.gmatch(str, "[^" .. delimiter .. "]+") do return token end end function contains(mainString, subString) return string.find(mainString, subString, 1, true) ~= nil end function longest_match_acode() local currentSeconds = os.time() local milliseconds = os.clock() * 1000 local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) .. string.format(".%03d", milliseconds) KSR.xlog.xinfo("starting date:" ..ddate) local size = 0 local result local acode = "29192" local inputbcode = "4474173" local delimiter = ":" while size == 0 do local currentSeconds = os.time() acode = string.sub(acode, 1, string.len(acode) - 1) KSR.ndb_redis.redis_cmd("srvN", "LRANGE price:" .. acode .." 0 -1", "r") size = tonumber(KSR.pv.get("$redis(r=>size)")) end local left = 0 local right = size - 1 while left <= right do local currentSeconds = os.time() local milliseconds = os.clock() * 1000 local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) .. string.format(".%03d", milliseconds) resultRight = tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])")) resultLeft = tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])")) local bcodeLeft = split(resultLeft, delimiter) local bcodeRight = split(resultRight, delimiter) if contains(inputbcode, bcodeRight) then KSR.xlog.xinfo("Input bcode contains bcode value from Redis: " .. resultRight .. " at index: " .. right .. " with date: " .. ddate) end if contains(inputbcode, bcodeLeft) then KSR.xlog.xinfo("Input bcode contains bcode value from Redis: " .. resultLeft .. " at index: " .. left .. " with date: " .. ddate) end left = left + 1 right = right - 1 end local currentSeconds = os.time() local milliseconds = os.clock() * 1000 local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) .. string.format(".%03d", milliseconds) KSR.xlog.xinfo("ending date:" ..ddate) end function ksr_request_route() if KSR.is_INVITE() then longest_match_acode() KSR.sl.send_reply("404", "Not found"); end end I'm getting the following warning in the log : WARNING: LUA {INVITE}: <core> [core/pvapi.c:340]: pv_cache_add(): pv cache limit is going to be exceeded - pkg memory may get filled with pv declarations I think the issue is in the following lines: resultRight = tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])")) resultLeft = tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])")) the list coming from Redis has around 30k items, which I'm iterating through using value[x] which is exhausting the pv declaration as per my understanding. How to overcome this issue? Is it possible to save the list as an array variable and then iterate through that array? Regards,

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Kamailio v5.7.3 Released
by Daniel-Constantin Mierla 17 Nov '23

17 Nov '23

Hello, Kamailio SIP Server v5.7.3 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.3. For more details about version 5.7.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/11/kamailio-v5-7-3-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 Kamailio World Conference - www.kamailioworld.com

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Releasing v5.7.3
by Daniel-Constantin Mierla 17 Nov '23

17 Nov '23

Hello, I am considering to release Kamailio v5.7.3 (out of branch 5.7) soon (likely on Friday 17 or during the beginning of next week). 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 -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - www.kamailioworld.com

1 1

SIP Trunk registration In Kamailio
by tanmoykarmakar10＠gmail.com 17 Nov '23

17 Nov '23

I want to register SIP Trunk directly in Kamailio. How can I register the SIP trunk in Kamailio. Please find the below sample SIP trunk details. Network IP 10.2.2.2 Subnet Mask 255.255.255.252 Customer IP 10.3.3.3 Gateway IP 10.1.1.1 SBCIP 10.4.4.4 Username 12345678 Password 1234 Please help me with configuration to register SIP trunk in kamailio.

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dlg_req_within kamailio 5.7 CSeq values
by Jonathan Hunter 16 Nov '23

16 Nov '23

Hi All, I am playing around with the dlg_req_within to send a REINVITE to toggle some rtpengine parameters and this works fine. The first time I use the command after tracking the dialog, it sends a REINVITE with an incremented CSEQ value. However when I issue it again, it sends another REINVITE but with the same CSEQ value, so again 2 for example. How can you ensure its incremented or can you set it somehow? As would like to generate at least 2 reinvites per dialog. Many thanks Jon Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

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Re: uac_req_send + evroute + crash
by Dr. Barabás Péter 15 Nov '23

15 Nov '23

Hello Daniel, I’m using kamailio 5.7.2 currently and this misterious crash occurs again. Now core files are generated also, here is the trace: (gdb) bt #0 __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:50 #1 0x00007fae9a2d0859 in __GI_abort () at abort.c:79 #2 0x00005614ad79afd8 in qm_debug_check_frag (qm=0x7fae78e77000, f=0x7fae79805028, file=0x7fae77c8c134 "uac: uac_send.c", line=860, efile=0x5614ad936f99 "core/mem/q_malloc.c", eline=511) at core/mem/q_malloc.c:129 #3 0x00005614ad79f7c1 in qm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:511 #4 0x00005614ad7ab410 in qm_shm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:1350 #5 0x00007fae77c3686b in uac_send_tm_callback (t=0x7fae797dcf40, type=131072, ps=0x7fff8bd990d0) at uac_send.c:860 #6 0x00007fae783abc27 in run_trans_callbacks_internal (cb_lst=0x7fae797dcfe8, type=131072, trans=0x7fae797dcf40, params=0x7fff8bd990d0) at t_hooks.c:241 #7 0x00007fae783abd1e in run_trans_callbacks (type=131072, trans=0x7fae797dcf40, req=0x0, rpl=0x0, code=0) at t_hooks.c:261 #8 0x00007fae782bd8cb in free_cell_helper (dead_cell=0x7fae797dcf40, silent=0, fname=0x7fae783e9f62 "timer.c", fline=653) at h_table.c:165 #9 0x00007fae7837649c in wait_handler (ti=724701450, wait_tl=0x7fae797dcff8, data=0x7fae797dcf40) at timer.c:653 #10 0x00005614ad50be68 in timer_list_expire (t=724701450, h=0x7fae78ef5d18, slow_l=0x7fae78ef95e0, slow_mark=33641) at core/timer.c:857 #11 0x00005614ad50c3a7 in timer_handler () at core/timer.c:922 #12 0x00005614ad50c8d5 in timer_main () at core/timer.c:961 #13 0x00005614ad45fbf3 in main_loop () at main.c:1833 #14 0x00005614ad46ae93 in main (argc=14, argv=0x7fff8bd99b48) at main.c:3086 I think it occurs only if $uac_req(evroute) is set to 1 and I define the event_route[uac:reply] route. My uac_reply route is easy: if ($uac_req(evcode)==200) { xlog("L_INFO", "Registration for user $fU has been refreshed successfully."); $sht(vtp=>asterisk_restarted) = $null; route(SAVE_REG); } else { xlog("L_INFO", "Registration failed: $uac_req(evcode)"); } the SAVE_REG route is the following: route[SAVE_REG] { xlog("L_INFO", "Saving pbx registration...:$fU"); $var(maxExpiry) = MAX_PBX_REG_EXPIRY; route(GET_CLIENT_ID); sql_pvquery("ca", "select count(*) from pbxusers where user = '$fU'", "$var(pbxusersCnt)"); if ($var(pbxusersCnt)>0) { if ($var(clientId) == 0) { sql_query("ca", "update pbxusers set expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } else { sql_query("ca", "update pbxusers set client_id = '$var(clientId)', expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } } else { sql_query("ca", "insert into pbxusers (user, expiry, client_id, online) values ('$fU', TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), '$var(clientId)', 0)"); } } And GET_CLIENT_ID route is: route[GET_CLIENT_ID] { $var(clientId) = 0; if (is_present_hf("X-AT-ClientId")) { $var(clientId) = $hdr(X-AT-ClientId); } } Any idea, is it a bug or I do something wrong in config files? Peter From: Dr. Barabás Péter via sr-users <sr-users(a)lists.kamailio.org> Date: Sunday, 2023. October 8. 18:49 To: miconda(a)gmail.com <miconda(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com> Subject: [SR-Users] Re: uac_req_send + evroute + crash Hello Daniel, I do not see any “failed to send request with authentication”. The next CRITICAL log appears after calling uac_req_send(): CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring Now we use kamailio 5.7.2 and no crash has come, but the critical logs above exist. I saw a ticket in github: https://github.com/kamailio/kamailio/issues/3522, may the crash be similar or the same to that? I will make a try next week with downgrading to 5.6.2 and try to reproduce the crash. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Friday, 2023. October 6. 13:55 To: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, do you also get error log messages that include "failed to send request with authentication"? Which CRITICAL log appears when you call uac_req_send()? You pasted a couple of them in the initial email? Cheers, Daniel On 05.10.23 17:32, Dr. Barabás Péter wrote: Hello, I used kamailio version 5.6.2. I refreshed to 5.7.2 today. As I remember it was in kamailio long time ago therefore I could skip evroute route. But this CRITICAL log appears always when I call uac_req_send(). I call it with settings: $uac_req(auser) = $var(username); $uac_req(apasswd) = $var(password); Where username and password are retrived from web service before call. In event_route I got first 401 after 200. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Date: Thursday, 2023. October 5. 17:25 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com><mailto:dr.peter.barabas@gmail.com> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, On 05.10.23 16:34, Dr. Barabás Péter via sr-users wrote: Hi All, I use kamailio In front of Asterisk and kamailio needs to refresh registrations periodically towards Asterisk to ensure the availability of users from Asterisk side. I use uac module and call uac_req_send() for sending REGISTER requests. I set $uac_req(evroute)=1 The event_route[uac:reply] is called fine, but in kamailio logs I see the next lines: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring After some time, kamailio has crashed. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) – ignoring CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 34 ALERT: <core> [main.c:774]: handle_sigs(): child process 1407950 exited by a signal 6 what version of Kamailio are you using? Is Asterisk challenging for authentication? Does it happen every time or seldom? Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com

1 0

How to check for presence of a file from a route? (Or any other way to tell kamailio not to accept new calls)
by Benoit Panizzon 13 Nov '23

13 Nov '23

Hi I'm still looking for a better way to tell Kamailio that we want to enter 'maintenance' and STAY in maintenance after a restart. Maintenance is: Reject all messages without totag with 503 to prevent creating new dialogues. I could use a shared pv and the use kamcmd pv.shvSet to toggle it. But when kamailio is restarted, that pv is also reset to it's initial state. Not good, if I want to make sure the node stays in maintenance mode after a restart as for example after pushing a config change via ansible. So what comes to my mind is to check for the presence of a file. something like: route[CHECK_MAINTENANCE] { if (!has_totag() && is_method("INVITE")) { if (file_exist("/etc/kamailio/maintenance.flag")) { xlog("L_ERR", "Maintenance flag present! Rejecting INVITE\n"); t_send_reply("503", "Maintenance mode - no new calls accepted"); exit; } } } 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 ______________________________________________________

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Inconsistent behavior between reg_fetch_contacts (registrar module) and reginfo_handle_notify (pua_reginfo module)?
by Mohammad Hossein Ahmadi 13 Nov '23

13 Nov '23

Hello everyone, I have previously set up Kamailio 5.7.2 IMS configuration (including P-CSCF, S-CSCF and I-CSCF) and successfully made voice calls. Now I'm trying to set up an IP-SM-GW AS using the example here (with some tweaks): https://github.com/kamailio/kamailio/blob/master/src/modules/smsops/example… Currently I'm trying to send the SMS using a SIP software client (MicroSIP). The problem is, while the SMS is received at the AS and stored in the database, it does not send it to the callee. While troubleshooting the problem, I've found that reg_fetch_contacts("location", "$var(uri)", "caller") in route[SEND_SMS] cannot find the contact which was added by reginfo_handle_notify("location") in route[NOTIFY] in the usrloc database. By tracing the problem more, I've found that reginfo_handle_notify (when handling reg event NOTIFY) adds the complete AoR in the "username" field in the location table, e.g. as "bob(a)ims.mnc001.mcc001.3gppnetwork.org". But reg_fetch_contacts (when trying to send the SMS) looks up the table using only the user part in the "username" field, e.g. "bob". So it cannot find the callee contact and the configuration assumes the callee is still not registered; so it tries to send out a SUBSCRIBE every time it tries to send an SMS to the callee, instead of actually sending the SMS. Does anybody have any idea on what could have caused this inconsistent behavior between reginfo_handle_notify and reg_fetch_contacts?

2 6

Dispatcher list reloading itself from file?!?
by Benoît Panizzon 13 Nov '23

13 Nov '23

Hi List I wanted to take one of our cores down gracefully by not destroy running dialogues to be able to implement a config change on Kamailio 5.5 For this I did: kamctl dispatcher.remove ID SIP-URI, for every IC This worked fine, Kamailio is instructed to reply 503 on out of dialogue traffic from unknown endpoints causing the endpoints to re-route traffic to the other core, eventually making it possible to stop and restart Kamailio without harm. dialog count was steadily decreasing, until after about 10 minutes after removing the dispatcher URI's it started to increase again. kamctl dispatcher.list shows me, all dispatcher URI I removed are back! There is no ds_reload() being executed by calls. In the module description, I find no hint, that the dispatcher config is being reloaded from file in some interval or so. What could be the cause of my removed dispatcher suddenly being back? -- 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 ______________________________________________________

2 2

pseudo variable in include_file not possible
by Bernd Krueger-Knauber 13 Nov '23

13 Nov '23

Hi, since we use git also for our kamailio stuff, we need to check in also host specific stuff. (same kamailio.cfg but different addresses and ... ) For this I thought an include_file for different hosts would be nice, but I can not check in the different host specific files, because I can not use something like: |include_file "$HN(n).cfg" It is not allowed in kamailio 5.6.x Or better: the pseudo variable is not replaced. Is there a way to work arround, or is it possible to make it possible? Best regards Bernd |

3 2

Working with kamailio websocket and asterisk
by SAMUEL MOYA TINOCO 13 Nov '23

13 Nov '23

Good evening everyone, I'm trying to set a scenario with my extensions registered against kamailio via wss and then once authenticated forward the register to my asterisk. After this, I use dispatcher module to balance invites to my asterisks. I have a few questions, the first part seems to be working as I want but when asterisk receives an invite and responds with INVITE (SDP) it doesn't appear on my kamailio (but every other message appears) You can see it in this sngrep captures Kamailio: [cid:image001.png@01DA0E88.5F82F510] Asterisk [cid:image002.png@01DA0E88.5F82F510] I guess it is because something in my asterisk pjsip configuration is wrong but I can't figure out what. So I'm start to think that maybe It is because I need to configure an rtpproxy app to change SDP and make RTP traffic go through kamailio Maybe there's something else I need to do, or there's a guide to do this. But I can't find anything Any help will be very appreciated Thank you in advance Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image003.png@01DA0E88.5F82F510] Soluciones inteligentes para la autonomía personal

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Info: qm and fm memory managers aligning addresses to 16 bytes
by Daniel-Constantin Mierla 10 Nov '23

10 Nov '23

Hello, following the Kamailio Developers Meeting discussions and provided that GNU/Linux malloc() aligns allocated memory to 16 bytes on x86_64 (64b CPUs), I just updated the native Kamailio qm (quick malloc - the default one) and fm (fast malloc) memory managers to also align to 16 bytes: - https://www.gnu.org/software/libc/manual/html_node/Aligned-Memory-Blocks.ht… By default they do it to 16 also for 32b CPUs, the overhead at startup with default config was rather insignificant (like 5000 bytes for private memory set to 8MB), so I didn't want to bother that much (that's also because some 32b CPUs could require larger alignment that the size of the pointer address), but for flexibility I added the an option to set the alignment size at compile time with KSR_MEMORY_ALIGN define (e.g., -DKSR_MEMORY_ALIGN=8UL). The main benefit at this moment is that Kamailio should be able to use the OS distribution (e.g., Debian) packaged libwolfssl instead of bundling and building the library inside it to compile with custom flags, like it is done now. Also, it is safe for the future to use other libraries that leverage internally the OS memory alignment size for specific optimizations. I also increased the optimize factor to 15, which should speed up a bit dealing with larger chunks of memory (now up to 32KB, previously was up to 16KB). Considering that it is more and more common to deal with larger SIP messages (e.g., a webrtc invite can easily be like 20KB) as well as tls/encryption needs addition space that the unencrypted udp/tcp. Hopefully these updates don't have side effects, it was nothing consistent changed, a few defines and fields to match the new alignment constraints. But testing is important, try to play with the master and your configs whenever you have any chance. Unfortunately the tlsf memory manager seems to be specifically designed for aligning to 8 bytes, a rather old related issue in that project is not concluded: - https://github.com/mattconte/tlsf/issues/16 That means the tls_wolfssl module linked with the officially packaged libwolfssl has to be used with qm or fm malloc from now on. Or it has to be built like so far to use tlsf memory manager. The tls/tlsa module can still be used with tlsf memory manager. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services

1 0

Development environment for the pre-commit hooks and clang-format
by Daniel-Constantin Mierla 10 Nov '23

10 Nov '23

Hello, during the Kamailio Developers Meeting 2023 in Dusseldorf early this week, Victor Seva shared how one can setup own development environment for facilitating the automatic checks for clang-format and few other useful bits (e.g., training white spaces) using pre-commit tool. So I thought it would be useful to shared further to both developers and users communities (as the later can make patches and pull requests as well). If I forgot something, Victor should amend me, but the main steps are: - install pre-commit package (either via apt or via pip3) - install clang-format (debian/ubuntu should have it as a package) - go to the folder with git clone of Kamailio source and run: pre-commit install The above command can take some time, but it is needed only once. After that, on evey commit, the checks should be performed a summary should be presented on screen, like: $ git commit src/ check yaml...........................................(no files to check)Skipped check xml............................................(no files to check)Skipped fix end of files.........................................................Passed trim trailing whitespace.................................................Passed check for merge conflicts................................................Passed mixed line ending........................................................Passed clang-format.............................................................Passed If the check of clang-format results in "Failed", like: clang-format.............................................................Failed the pre-commit hooks will correct it, so you can just run again the commit command and the 2nd time should be good to go. If for whatsoever reasons pre-commit is installed but you need to ignore/skip the pre-commit hooks, just provide -n or --no-verify to the commit command, like: $ git commit -n src/ By using pre-commit hooks, one could easily avoid making pull requests that are failing to meet the code formatting checks on github portal as well as ensure that as developer one does no forget to run clang-format before commit. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services

1 0

Issue with Dialog Handling in Kamailio Post 2-Minute Duration
by aaron.soto＠dialapplet.com 10 Nov '23

10 Nov '23

I'm encountering an issue with dialog handling in Kamailio. Initially, everything functions correctly, and if a connection terminates within two minutes, the BYE method works as expected. However, if the call exceeds two minutes, the BYE method doesn't seem to be recognized, and there are no related entries in the logs. As someone new to working with Kamailio, I'm finding this quite perplexing. Could anyone offer guidance or troubleshooting steps to resolve this issue? Thank you for your assistance.

3 6

Sharing load between rtpengine servers
by David Cunningham 09 Nov '23

09 Nov '23

Hello, We have a Kamailio configuration with the following, however all the load is going to the server at 22.22.22.22 (hiding the real IP obviously) instead of being shared evenly between 22.22.22.22 and 33.33.33.33. Can anyone please tell me why? The rtpengine server at 11.11.11.11 is intended to only receive calls if the other two are offline. Thank you very much. #!define RTPENGINE_ADDR "udp:11.11.11.11:7724=1 udp:22.22.22.22:7724=99999999 udp:33.33.33.33:7724=99999999" modparam( "rtpengine", "rtpengine_sock", RTPENGINE_ADDR ) -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

5 10

Redis issue when getting results in Kamailio
by Ali Taher 09 Nov '23

09 Nov '23

Hello all, I have redis database where the data is inserted in JSON format: #Redis Command JSON.GET Index1J:12345678 #Redis Result [ { "SupplierID": "7", "BCode": "987654", "Rate": 0.159 }, { "SupplierID": "7", "BCode": "765432", "Rate": 0.159 }, { "SupplierID": "7", "BCode": "5432159", "Rate": 0.159 } ] When I try to get the result using Lua script in Kamailio like the following: #kamailio command if KSR.ndb_redis.redis_cmd("srvN", "JSON.GET pricelistJ:441246890", "ra") then KSR.xlog.xinfo("array size: $redis(ra=>size)\n"); KSR.xlog.xinfo("Redis Response: " .. tostring(ra) .. "\n") KSR.xlog.xinfo("first values: $redis(ra=>value[0]) , $redis(ra=>value[1])\n"); end Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}: <script>: array size: <null> Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}: <script>: Redis Response: nil Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}: <script>: first values: <null> , <null> I'm getting null results. I don't know what I'm doing wrong here. Can you please assist?

2 3

Sequential fork with different Call-Ids
by Vanderlei Torres Batistela 09 Nov '23

09 Nov '23

Hello all, I have a scenario on wich requires sequential fork (up to 30 possible destinations) with different Call-ID for each leg. I am using topoh module enabled, but it always puts the same Call-ID. Someone have a solution for this scenario? Regards

4 3

3xx is not relayed to the caller if there are 2 or more branches in the transaction
by Михаил Горюцкий 07 Nov '23

07 Nov '23

Hello, I encountered such a problem: An INVITE request comes to Kamailio. Kamailio finds a location and forwards this INVITE to a device. The device receives a push notification and registers on a new IP: port. The tsilo module creates a new branch. And this branch sends the INVITE. The device responds with 100 -> 180 -> 302. 100 and 180 are forwarded to the calling subscriber, but anything greater than 300, such as 302, is not forwarded. DEBUG: tm [t_reply.c:1363]: t_should_relay_response(): ->>>>>>>>> T_code=180, new_code=302 DEBUG: tm [t_reply.c:1448]: t_should_relay_response(): store - other branches still active DEBUG: tm [t_reply.c:1925]: relay_reply(): reply status=3 branch=1, save=1, relay=-1 icode=0 msg status=302 Kamailio saves this code and does not pass it to the caller. Is there a way to immediately pass this 302? It seems like something can be done in t_on_branch_failure, which is invoked, but I can't figure out what needs to be done there. Is there a parameter that can solve my problem? A similar question has been asked before, but the link in Daniel's response leads to a non-existent ID: https://lists.kamailio.org/pipermail/sr-users/2009-March/022447.html

2 2

defexps not working as expected
by Richard Chan 07 Nov '23

07 Nov '23

In the config file: #!define KEYVALUE "key=s:value" modparam("pv", "varset", KEYVALUE) works. But if I try to use defexp or defexps the ID KEYVALUE is not set as expected and the parser complains. ## this doesn't parse #!defexp KEYVALUE2 "key=s:value" #!defexps KEYVALUE3 "key=s:value" # both these lead to syntax error modparam("pv", "varset", KEYVALUE2) modparam("pv", "varset", KEYVALUE3) Any ideas why defexp doesn't behave the same way here? Shih-Ping

1 0

SIP Proxy : BYE next hop
by Ali Taher 03 Nov '23

03 Nov '23

Hello all, I'm using Kamailio as SIP proxy between session manager and SBC. I came across an issue in BYE request where the SBC sent in the header list of Routes where first route is Kamailio IP, and the second is SM IP, in addition to other routes (last route is the caller party) What Kamailio did is to route the BYE to the address set in the Request-URI, which is the caller party. What should be done instead is that Kamailio should remove the Route header including its own IP, and route the BYE to the next hop which is in this case the SM IP. How to accomplish that? Following is the Route header in the BYE request coming from SBC: [truncated]Route: <sip:KAMAILIO_IP;lr;ftag=xxxxxxxxxxxxxxxxxxxxxxx>,<sip:SM1@SM_IP;transport=udp;lr;av-asset-uid=xxxxxxxxxx>,<sip:127.0.0.2:15060;transport=udp;ibmsid=local.xxxxxxxxxxxx_xxxxxxxxx_xxxxxxxxx;lr;ibmdrr>,sips:127.0.0.2:15061;ibmsid=local.xxxxxxxxxx_xxxxxxxxxx_xxxxxxxx;lr;ibmdrr, sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xxxxxx, sip:CALLER_PARTY_IP:5061;transport=tls;lr Regards,

3 6

PLMN 31501 , VoLTE IMS Kamalio
by snahmed.snahmed＠gmail.com 31 Oct '23

31 Oct '23

Hi, I was able to test the PLMN 00101 for VoLTE, but unable to test it with other PLMNs. I am specifically looking to have a successful test with PLMN31501. VoLTE is not enabling once I configure it. I followed this https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ Any one has experience for PLMN 315?

1 0

Registrar / Path
by Jawaid Bazyar 31 Oct '23

31 Oct '23

Hi, I have a scenario where I have an edge proxy (named RS), which locally stores registrations from Endpoints in its USRLOC, but which then also forward's REGISTER requests on to another proxy (named CN), which stores registrations in its USRLOC. (which then does DMQ with another CN like it). Both RS and CN are storing Path information, for example: kamctl ul show displays on both RS and CN: "Path": "sip:vs-rs01.blah.foo.bar;lr;received=sip:3.4.5.6:33577", So I know the path info is in there. I want CN , when presented with an INVITE, to reply with a 302 Redirect. Right now I have this: if (!lookup("location",sip:$tU@nodomain)) { } reg_fetch_contacts("location", "$tu", "called"); xlog("location info $ulc(called=>path)"); sl_send_reply("302","Redirect"); and the xlog does display the path. Good, so I have it being sent all the way through RS and CN and into CN’s USRLOC. but this isn’t complete. I am now faced with two issues. One, I need to mangle the Path into a Contact header for the Redirect. Two, I need to delete the Contact header that sl_send_reply is putting on there, which is the Contact as RS sees it from the endpoint. This needs to be removed and replaced with a Contact of only the RS server. It feels like there ought to be a simpler way to do this than parsing the Path string and reassembling a Contact from it, but if there is it's not readily apparent. Both Google and chatgpt have failed me :) Thanks in advance, Jawaid

3 7

Kamailio 5.7 still segfaulting on TLS with Ubuntu Jammy
by Benoît Panizzon 31 Oct '23

31 Oct '23

Hi Gang I just updated our dev plattform from 5.6 to 5.7 5.5 was working fine with TLS under Ubuntu Focal. 5.6 under Jammy started to dump cores. 5.7 still shows the same behaviour, usually shortly after a restart. Oct 20 09:45:01 dev-cpereg01 kamailio[11925]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f1bb0dd4bd8), called from tls: tls_init.c: ser_free(405), first free tls: tls_init.c: ser_free(405) - ignoring Oct 20 09:45:11 dev-cpereg01 kamailio[11905]: ALERT: <core> [main.c:776]: handle_sigs(): child process 11924 exited by a signal 11 Oct 20 09:45:11 dev-cpereg01 kamailio[11905]: ALERT: <core> [main.c:779]: handle_sigs(): core was generated Could anyone point me the correct direction on how to fix this issue? Am I the only one seeing this issue? -- 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 ______________________________________________________

2 4

RPM Repos Maintenance Needed
by tyler moore 30 Oct '23

30 Oct '23

Hi All, I started a discussion on the matrix channel about some 404 RPM repos and wanted to bring it here to further discuss. We typically configure our various RPM-based package managers to allow patch updates from the repos. As an example, if trying to install 5.7.x, can be done as follows: |yum -y install yum-utils yum-config-manager --add-repo https://rpm.kamailio.org/centos/kamailio.repo yum-config-manager --disable \* yum-config-manager --enable kamailio-5.7 yum install kamailio | This will fail though because https://rpm.kamailio.org/centos/7/5.7/5.7/x86_64/ does not exist. Throughout the repos for centos/rhel/fedora I found the existence of packages is inconsistent. We can tell the intention was to allow the above behavior as the kamailio.repo comes with the above configuration. The above example will update /etc/yum.repos.d/kamailio.repo as follows: |[kamailio-5.7] name=Kamailio - 5.7 - Packages for the latest Kamailio 5.7 release baseurl=https://rpm.kamailio.org/centos/$releasever/5.7/5.7/$basearch/ enabled=1 metadata_expire=30d repo_gpgcheck=0 gpgkey=https://rpm.kamailio.org/rpm-pub.key type=rpm skip_if_unavailable=True | That is for centos though, and I found that some of the other distros have incomplete kamailio.repo files as well. In example, the rhel kamailio.repo is missing separate entries for 5.7, but it does exist in the centos version. Looks like some small cleanup is in order to make the end user experience more consistent. I believe sergey was maintaining those repos last, please chime in here if you can. If needed I can pick up some of those responsibilities as well. -- Regards, *Tyler Moore* <https://devopsec.net/> Full Stack Software Engineer <http://www.google.com/search?q=full+stack+software+engineer+job+description> Flyball Labs <http://goflyball.com/> Office: 888-907-2085, ext: 34 <tel:8889072085;ext=34> Cell: 248-909-2769 <tel:2489092769> Email: tmoore(a)goflyball.com FLYBALL TECHNICAL SOLUTIONS, LLC <http://flyballlabs.com/> -- Tyler Moore Regards, *Tyler Moore* <https://devopsec.net/> Full Stack Software Engineer <http://www.google.com/search?q=full+stack+software+engineer+job+description> Flyball Labs <http://goflyball.com/> Office: 888-907-2085, ext: 34 <tel:8889072085;ext=34> Cell: 248-909-2769 <tel:2489092769> Email: tmoore(a)goflyball.com FLYBALL TECHNICAL SOLUTIONS, LLC <http://flyballlabs.com/>

3 2

RPM Repos Maintenance Needed
by tyler moore 28 Oct '23

28 Oct '23

Hi All, I started a discussion on the matrix channel about some 404 RPM repos and wanted to bring it here to further discuss. We typically configure our various RPM-based package managers to allow patch updates from the repos. As an example, if trying to install 5.7.x, can be done as follows: yum -y install yum-utils yum-config-manager --add-repo https://rpm.kamailio.org/centos/kamailio.repo yum-config-manager --disable \* yum-config-manager --enable kamailio-5.7 yum install kamailio This will fail though because https://rpm.kamailio.org/centos/7/5.7/5.7/x86_64/ does not exist. Throughout the repos for centos/rhel/fedora I found the existence of packages is inconsistent. We can tell the intention was to allow the above behavior as the kamailio.repo comes with the above configuration. The above example will update /etc/yum.repos.d/kamailio.repo as follows: [kamailio-5.7] name=Kamailio - 5.7 - Packages for the latest Kamailio 5.7 release baseurl=https://rpm.kamailio.org/centos/$releasever/5.7/5.7/$basearch/ enabled=1 metadata_expire=30d repo_gpgcheck=0 gpgkey=https://rpm.kamailio.org/rpm-pub.key type=rpm skip_if_unavailable=True That is for centos though, and I found that some of the other distros have incomplete kamailio.repo files as well. In example, the rhel kamailio.repo is missing separate entries for 5.7, but it does exist in the centos version. Looks like some small cleanup is in order to make the end user experience more consistent. I believe sergey was maintaining those repos last, please chime in here if you can. If needed I can pick up some of those responsibilities as well. Regards, Tyler Moore Full Stack Software Engineer Flyball Labs Office: 888-907-2085, ext: 34 Cell: 248-909-2769 Email: tmoore(a)goflyball.com

2 2

Kamailio 5.4.3 Troubleshooting Traffic Blocking Issue After Docker and OpenSSL Update
by Pintu Lohar 27 Oct '23

27 Oct '23

Hi All, We have recently encountered a challenging issue in our environment that I wanted to bring to your attention. A few months ago, we performed updates to our Docker and OpenSSL installations, upgrading OpenSSL to version 1.1.1u. Since then, we have been experiencing intermittent traffic blocking (TLS and UDP both for around 10 minutes and getting recovered) on some of our instances. Prior to the OpenSSL upgrade, we had not observed any such issues. However, the situation has become more complex due to a substantial increase in traffic, with each instance now handling 4K-5K concurrent calls and a daily total of approximately 200,000 calls per instance. Our initial suspicion was related to a Docker iptable issue. To investigate further, we conducted a GDB trace using "kamctl pstrap" and discovered specific log entries indicating potential issues within the "pthread_rwlock_wrlock" function. This has prompted us to explore whether the problems are linked to the OpenSSL upgrade. We are considering upgrading Kamailio to the latest version, although this process typically requires a significant amount of time for proper planning and testing. Given the high volume of calls we are currently handling, we are also exploring the possibility of a temporary solution involving a rollback of the OpenSSL version to see if it resolves the issue. Curiously, during a recent load test using SIPP, which simulated 10,000 concurrent calls with a rate of 100 calls per second, we did not encounter the same problem. any hints or insights you may have that could help us pinpoint the root cause of the problem . We do have a complete GDB trace available, but its size necessitates sharing a truncated version for now. version: kamailio 5.4.3 (x86_64/linux) 0178a5 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_BLACKLIST, 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: 0178a5 compiled on 12:36:56 Oct 22 2023 with gcc 4.8.5 Operating System Host : LSB Version: :core-4.1-amd64:core-4.1-noarch Distributor ID: Rocky Description: Rocky Linux release 8.8 (Green Obsidian) Release: 8.8 Codename: GreenObsidian container os (Centos7) Kamctl pstraip output (truncated to last few lines) ---start 154 ----------------------------------------------------- kamailio: No such file or directory. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". 0x00007fbf6f38a0c3 in __epoll_wait_nocancel () from /lib64/libc.so.6 $1 = 82 $2 = {pid = 154, unix_sock = 182, idx = 27, status = 1, rank = 48, desc = "tcp receiver (tcp:xxx.xxx.xxx.xx:5060 <http://223.62.234.57:5060/>) child=27", '\000' <repeats 81 times>} #0 0x00007fbf6f38a0c3 in __epoll_wait_nocancel () from /lib64/libc.so.6 No symbol table info available. #1 0x0000000000684858 in io_wait_loop_epoll (h=0xb41720 <io_w>, t=2, repeat=0) at core/io_wait.h:1039 n = 84 r = 0 fm = 0x2 revents = 8201656 __FUNCTION__ = "io_wait_loop_epoll" #2 0x0000000000699e7a in tcp_receive_loop (unix_sock=186) at core/tcp_read.c:1997 __FUNCTION__ = "tcp_receive_loop" #3 0x00000000005279e7 in tcp_init_children () at core/tcp_main.c:5134 r = 27 i = -1 reader_fd_1 = 186 pid = 0 si_desc = "tcp receiver (tcp:xxx.xx.xxx.xxx:5060 <http://223.62.234.57:5060/> )\000\000\000\355\217c\000\000\000\000\000\240\213{\001\001\000\000\000\017\000\000\000\000\000\000\000\250\a2o\276\177\000\000\200\b\231o\000\000\000\000\260\006\346\261\376\177\000\000\230\210w\000\000\000\000\000=\000\000\000>\000\000\000\270\016z\350\274\177\000\000\060;8o\002\000\000\000P\224\346\352\001\000\000" si = 0x0 __FUNCTION__ = "tcp_init_children" #4 0x000000000042ae66 in main_loop () at main.c:1771 i = 16 pid = 94 si = 0x0 si_desc = "udp receiver child=15 sock=xxxx.xxx.xxx.xx:5060\000\000\000\v\235\200", '\000' <repeats 13 times>, "P\r\346\261\376\177\000\000\270%}\000\000\000\000\000T\000\000\000\000\000\000\000\060;8o\277\177\000\000\343\207\201\000\000\000\000\000\277;8o\277\177\000\000 \271A\000\000\000\000\000@\374Ko\276\177\000" nrprocs = 16 woneinit = 1 __FUNCTION__ = "main_loop" #5 0x0000000000433ad6 in main (argc=9, argv=0x7ffeb1e60e38) at main.c:2856 cfg_stream = 0x16c9010 c = -1 r = 0 tmp = 0x7ffeb1e61f0c "" tmp_len = 2496 port = 2496 proto = 1472 ahost = 0x0 aport = 0 options = 0x7d5238 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 2208520674 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fbf6f40f3a0 <intel_02_known> p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 76, st_ino = 120506675, st_nlink = 2, st_mode = 16877, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1698340416, tv_nsec = 618631009}, st_mtim = {tv_sec = 1698340416, tv_nsec = 618631009}, st_ctim = {tv_sec = 1698340416, tv_nsec = 618631009}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 376 times>... option_index = 0 long_options = {{name = 0x7d742f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d2a34 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7d7434 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7d743a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7d7440 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7d7449 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7d7453 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7d745d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7d7468 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7d7471 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7d747c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" [Inferior 1 (process 154) detached] ---end 154 ------------------------------------------------------- ---start 155 ----------------------------------------------------- kamailio: No such file or directory. [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". 0x00007fbe6e5bd39e in pthread_rwlock_wrlock () from /lib64/libpthread.so.0 $1 = 83 $2 = {pid = 155, unix_sock = -1, idx = -1, status = 1, rank = -4, desc = "tcp main process", '\000' <repeats 111 times>} #0 0x00007fbe6e5bd39e in pthread_rwlock_wrlock () from /lib64/libpthread.so.0 No symbol table info available. #1 0x00007fbe68b745c9 in CRYPTO_THREAD_write_lock () from /lib64/libcrypto.so.1.1 No symbol table info available. #2 0x00007fbe68afe07e in CRYPTO_free_ex_data () from /lib64/libcrypto.so.1.1 No symbol table info available. #3 0x00007fbe68ea52c2 in SSL_free () from /lib64/libssl.so.1.1 No symbol table info available. #4 0x00007fbe69149480 in tls_h_tcpconn_clean_f (c=0x7fbceeae00d0) at tls_server.c:655 extra = 0x7fbceec3b898 __FUNCTION__ = "tls_h_tcpconn_clean_f" #5 0x00000000004fbf2d in _tcpconn_free (c=0x7fbceeae00d0) at core/tcp_main.c:1529 __FUNCTION__ = "_tcpconn_free" #6 0x0000000000510912 in tcpconn_put_destroy (tcpconn=0x7fbceeae00d0) at core/tcp_main.c:3284 __FUNCTION__ = "tcpconn_put_destroy" #7 0x00000000005156a5 in handle_tcp_child (tcp_c=0x7fbe6f813440, fd_i=-1) at core/tcp_main.c:3723 tcpconn = 0x7fbceeae00d0 response = {140449434960080, -2} cmd = -2 bytes = 16 n = 1865551792 t = 887186228 crt_timeout = 410 con_lifetime = 57680 __FUNCTION__ = "handle_tcp_child" #8 0x00000000005208ed in handle_io (fm=0x7fbe6f9d7c78, ev=1, idx=-1) at core/tcp_main.c:4542 ret = 1 __FUNCTION__ = "handle_io" #9 0x00000000004e93ed in io_wait_loop_epoll (h=0xad5740 <io_h>, t=5, repeat=0) at core/io_wait.h:1070 n = 1 r = 0 fm = 0x7fbe6f9d7c78 revents = 1 __FUNCTION__ = "io_wait_loop_epoll" #10 0x0000000000523c99 in tcp_main_loop () at core/tcp_main.c:4827 si = 0x0 r = 28 __FUNCTION__ = "tcp_main_loop" #11 0x000000000042b1be in main_loop () at main.c:1779 i = 16 pid = 0 si = 0x0 si_desc = "udp receiver child=15 sock=xxx.xx.xxx.xxx:5060 <http://223.62.234.76:5060/>\000\000\000\v\235\200", '\000' <repeats 13 times>, "P\r\346\261\376\177\000\000\270%}\000\000\000\000\000T\000\000\000\000\000\000\000\060;8o\277\177\000\000\343\207\201\000\000\000\000\000\277;8o\277\177\000\000 \271A\000\000\000\000\000@\374Ko\276\177\000" nrprocs = 16 woneinit = 1 __FUNCTION__ = "main_loop" #12 0x0000000000433ad6 in main (argc=9, argv=0x7ffeb1e60e38) at main.c:2856 cfg_stream = 0x16c9010 c = -1 r = 0 tmp = 0x7ffeb1e61f0c "" tmp_len = 2496 port = 2496 proto = 1472 ahost = 0x0 aport = 0 options = 0x7d5238 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 2208520674 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x7fbf6f40f3a0 <intel_02_known> p = 0xf0b5ff <Address 0xf0b5ff out of bounds> st = {st_dev = 76, st_ino = 120506675, st_nlink = 2, st_mode = 16877, st_uid = 0, st_gid = 0, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1698340416, tv_nsec = 618631009}, st_mtim = {tv_sec = 1698340416, tv_nsec = 618631009}, st_ctim = {tv_sec = 1698340416, tv_nsec = 618631009}, __unused = {0, 0, 0}} tbuf = '\000' <repeats 376 times>... option_index = 0 long_options = {{name = 0x7d742f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x7d2a34 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x7d7434 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x7d743a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x7d7440 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x7d7449 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x7d7453 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x7d745d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x7d7468 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x7d7471 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x7d747c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __FUNCTION__ = "main" [Inferior 1 (process 155) detached] ---end 155 ------------------------------------------------------- Thanks & Regards Pintu Lohar

1 0

Inquiry Regarding Kamailio Behavior and Drop Requests
by satyaprakash ch 26 Oct '23

26 Oct '23

Hi, We possess a code within Kamailio, which specifically handles malformed SIP responses. Below is the code snippet: *reply_routeCopy code if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop;* Additionally, we have Kamailio drop request statistics as follows: Command: *kamctl stats | grep core:drop_requests* Output of the command: *core:drop_requests = 5* My queries are: 1. Is the aforementioned "reply_route" related to the drop requests we are experiencing? 2. If it is not related, what could be the cause of the drop requests?

3 8

Information about Kamailio operating as an SBC
by Alvaro Escobar 24 Oct '23

24 Oct '23

Hello, I am looking information about Kamailio operating as an SBC. I saw this video of Kamailio World 2021: Kamailio As An SBC For Network Segregation: https://www.youtube.com/watch?v=UW6l3R4OnsY&t=1381s But where I can find more detailed technical information ? Regards

3 3

How to filter out xhttp request in event_route[siptrace:msg]
by Benoit Panizzon 24 Oct '23

24 Oct '23

Hi Further trying to eliminate every possible cause of UDP drops... We use Homer as HEP server in conjunction with the siptrace module. So to prevent DQM traffic to be sent to Homer I added: event_route[siptrace:msg] { if(is_method("KDMQ")) { drop(); } } And had a closer look to the traffic sent to homer. I noticed, also xhttp:requests, as far as I see the replies, not the requests, get mirrored to homer. xhttp is not a sip method I could match I guess. The event_route[siptrace:msg] docs state only method can be filtered. So I can not filter requests on the separate http port I listen on for xhttp json-rpc requests. How can I prevent xhttp traffic to be mirrored to homer by the siptrace module? 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 ______________________________________________________

4 5

STIR/SHAKEN with Kamailio
by Olle E. Johansson 23 Oct '23

23 Oct '23

Hello Kamailians! As STIR/SHAKEN seems to cross the ocean and arrive on the European shores, I’m curious on how you’ve implemented it with Kamailio. I asked on our Matrix chat and got two responses that seems to not use the Kamailio STIR/SHAKEN support but rather a 3rd party service that they’ve integrated using restful APIs. Please help me and share how you’re implementing STIR/SHAKEN with Kamailio. I’m not shaken, nor stirred. yet. Regards, /O

7 9

How to configure ndb_redis sentinel
by mayamatakeshi 21 Oct '23

21 Oct '23

Hi, regarding ndb_redis, I have 3 sentinels like this: "192.168.88.155:26379" "192.168.88.156:26379" "192.168.88.157:26379" So was thinking that I would configure ndb_redis this way: modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=1;sentinel= 192.168.88.155:26379") modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=0;sentinel= 192.168.88.156:26379") modparam("ndb_redis", "server", "name=srvZ;sentinel_group=myredis;sentinel_master=0;sentinel= 192.168.88.157:26379") But the samples in the documentation show an extra sentinel key in the config lines: # sentinel (for a redis slave) modparam("ndb_redis", "server", "name=srvZ;sentinel_group=group_name;sentinel_master=0;sentinel=1.2.3.4:26379 ;sentinel=1.2.3.5:26379") # sentinel (for a redis master) modparam("ndb_redis", "server", "name=srvZ;sentinel_group=group_name;sentinel_master=1;sentinel=1.2.3.4:26379 ;sentinel=1.2.3.5:26379") So I am not sure if my configuration is correct. Can someone clarify?

1 0

lookup() and database
by Jawaid Bazyar 20 Oct '23

20 Oct '23

Hi, I have an application as follows: Multiple Kamailio Nodes Shared Postgres database Auth information will be stored in DB and accessed with Authdb. Here is the behavior I seek: I would like all the Kamailo nodes to cooperate in updating the database, each updating data as appropriate: When a registration comes in to any Node, and it authenticated, create or update a database record in location table. When a registration expires on that Node, delete database record in location table. When INVITE comes in on any Node, ignore local cache and always lookup() record from database. #1 and #2 are used so that a third SIP system can query the database and look up the Node a NAT’d subscriber is ‘connected’ to. #3 is to let NON-NAT’d subscribers be contacted from any Node. Now I have something close to this now – using following settings: modparam("usrloc", "db_mode", 1) # immediately write registrations to the database. modparam("usrloc", "db_timer_clean", 0) # do not expire DB records separately modparam("usrloc", "db_check_update", 1) This takes care of #1 and #2 above. However, #3 seems to still be relying purely on local cache and does not hit the database. Is there a way to do #3? I can set db_mode to 3 but that would seem to have other effects besides merely making lookup() read from database. Thanks in advance, Jawaid

3 8

TOPOS + Forcing the send socket
by Marrold 19 Oct '23

19 Oct '23

Hi all, I am using Kamailio (5.7.2, Debian 11, Python KEMI) in a mutli-homed environment with the topos module to hide the topology. I have noticed that when I use `set_send_socket` or `$fsn` to force the socket it's sent from, it breaks topos and the ACK is not proxied to the other leg of the call. I am calling record_route() *after* forcing the socket. If I disable TOPOS the ACK works as expected, and the signalling looks correct. Or, with TOPOS enabled, if I comment out set_send_socket TOPOS works as expected. If I disable enable_double_rr, TOPOS works as expected and the ACK is forwarded, but without the double RR subsequent in-dialog requests use the wrong socket. One other observation, when I change the send socket I also see these warnings in the logs: 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1348]: dlg_onroute(): tight matching failed for ACK with callid='!!:MByLMlFAM.NfWxFAM.cAMxyfWjyLz.yAO.y6MxF1MxVZWG4ZMy**'/55, ftag='2023101714101800015'/19, ttag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36 and direction=0 24(32) WARNING: PY3 {ACK}: dialog [dlg_handlers.c:1355]: dlg_onroute(): dialog identification elements are callid='2023101714101800015@2900-0601-0284-80'/37, caller tag='2023101714101800015'/19, callee tag='2f55349a-2c59-4e37-bf58-fd84fb69ece9'/36 Is anyone using TOPOS + forcing the socket, or could someone advise where I am going wrong? Thanks in advance Matthew

3 8

Kamailio SIP proxy role
by Ali Taher 18 Oct '23

18 Oct '23

Hello, I'm using Kamailio as SIP proxy between Session Manager and SBC So here is the scenario : Session Manager sends a call (Invite) to Kamailio, kamailio needs to relay it to SBC after doing some modification. (-----Session Manager-----) ----------- > (----Kamailio----) ----------- > (-----SBC-----) My routing logic is similar to the below: route{ if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } if (is_method("INVITE")) { sql_query("cb", "select number from pool1 order by random() limit 1", "ra"); $var(rand)=$dbr(ra=>[0,0]); uac_replace_from($var(rand),"sip:$var(rand)@192.168.1.1"); xlog("L_INFO","Random: $var(rand)"); $var(ip) = $(ct{s.select,1,(a)}{s.select,0,;}{s.replace,>,}); remove_hf("Contact"); insert_hf("Contact: $var(rand) <sip:$var(rand)@$var(ip)>\r\n"); remove_hf("P-Asserted-Identity"); remove_hf("Route"); insert_hf("Route: <sip: 192.168.1.10>\r\n","Route"); record_route(); t_relay(); $td = "192.168.1.10"; $rd = "192.168.1.10"; t_relay(); } } Where: 192.168.1.1 is Kamailio IP 192.168.1.10 is SBC IP My question here is that Kamailio handle all in dialog messages between Session Manager and SBC? For example, after forwarding the invite packet to SBC, SBC sends 183 Session Progress to Kamailio, which in its turn relay it to the session manager, knowing that this is not mentioned in my routing logic, so how Kamailio knew that it must relay it to session manager in this case? Also I need to add to the 183 Session Progress sent to session manager by Kamailio record route and Via headers , how to do it ? Regards,

1 0

Dispatcher HA:won't route headers be wrong if in-dialog request is sent to backup?
by Richard Chan 18 Oct '23

18 Oct '23

Hi sr-users, In the case of dispatcher-based HA such as the Wazo example ( https://wazo-platform.org/blog/kamailio-ha-dispatcher-and-dmq) won't the Route: headers be wrong for in-dialog requests if the request is sent to the backup proxy? front proxy -> proxy A, B, C, D,... -> network Suppose the INVITE goes through proxy A, then proxy A goes down. Based on the HA logic, an in-dialog request will go to a backup, say, proxy B. However, won't the Route headers be wrong (since they are placed by A) in the original transaction? How do you handle Route:, Record-Route: headers at the HA tier? 1. In the front proxy & HA tier don't use route headers at all ?? 2. Manually check and remove route headers (i.e proxy B treats route headers for proxy A as "myself") ?? I am curious as to what is your practice. Regards Shih-Ping

1 0

Restarts Kamailio after general protection fault
by Daian Conrad 17 Oct '23

17 Oct '23

Hi all, I have started using kamailio as a proxy register for asterisk's and works fine, but in some moment kamailio show WARNINGs, ALERTS and CRITICAL logs and restart. I don't no how debug correctly, but the crash show this log lines *Oct 16 21:34:26 proxy /usr/sbin/kamailio[2148]: CRITICAL: <core> [core/mem/q_malloc.c:123]: qm_debug_check_frag(): BUG: qm: fragm. 0x7f14a27a5ab8 (address 0x7f14a27a5af0) beginning overwritten (7f14a27a8e58)! Memory allocator was called from uac: uac_send.c:860. Fragment marked by UH��AWAVAUATSH��:139726602009368. Exec from core/mem/q_malloc.c:511.* ------------------------------------------------------------- *Oct 16 21:39:46 proxy /usr/sbin/kamailio[2556]: CRITICAL: <core> [core/mem/q_malloc.c:123]: qm_debug_check_frag(): BUG: qm: fragm. 0x7ff7ad456330 (address 0x7ff7ad456368) beginning overwritten (0)! Memory allocator was called from uac: uac_send.c:860. Fragment marked by (null):0. Exec from core/mem/q_malloc.c:511.Oct 16 21:39:46 proxy /usr/sbin/kamailio[2578]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 19Oct 16 21:39:46 proxy /usr/sbin/kamailio[2539]: ALERT: <core> [main.c:774]: handle_sigs(): child process 2556 exited by a signal 6Oct 16 21:39:46 proxy /usr/sbin/kamailio[2539]: ALERT: <core> [main.c:777]: handle_sigs(): core was not generatedOct 16 21:39:46 proxy systemd[1]: kamailio.service: Main process exited, code=exited, status=1/FAILUREOct 16 21:39:46 proxy systemd[1]: kamailio.service: Failed with result 'exit-code'.Oct 16 21:39:46 proxy systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart.Oct 16 21:39:46 proxy systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 9.Oct 16 21:39:46 proxy systemd[1]: Stopped Kamailio - the Open Source SIP Server.* any advice is apreciated -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

2 3

ERROR: run_top_route
by satyaprakash ch 17 Oct '23

17 Oct '23

Hi, We are having an error in the Kamailio logs which we need to resolve this issue, ERROR is ::* /usr/local/sbin/kamailio[10149]: ERROR: tm [t_reply.c:1081]: run_failure_handlers(): error running run_top_route for failure handler.* We are getting this error at the time of the 3xx response, Can anyone help me on this? Thank you.

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