- sr-users - kamailio.org

Releasing v5.8.1
by Daniel-Constantin Mierla 03 Apr '24

03 Apr '24

Hello, with many countries having public holidays around Catholic Easter, I am considering to release Kamailio v5.8.1 (out of branch 5.8) on Wednesday, April 3, 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 World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

start_recording' or 'record-call on' is not invoking
by pawan srivastava 03 Apr '24

03 Apr '24

Hi I am testing rtpengine recording with kamailio in lab. I have 1 endpoint registered to Kamailio. My kamailio and rtpengine are running on same box. I have another kamailio2 instance with 2nd endpoint to make a call to 1st endpoint. I have observed that Response dump for 'offer'/'answer' but further start recording didn't invoke. root@ip-kamailio:/etc/kamailio# rtpengine-ctl list sessions all callid: J0WtgYIKd0Kwj1iL6BiFOg.. | deletionmark: no | created: 1712120753 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: uH_ozKj18Nafshsm_sTElQ.. | deletionmark: no | created: 1712120730 | proxy:127.0.0.1:52642 | redis_keyspace:0 | foreign:no callid: VYIw791dJska8PiIFspy8w.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:54804 | redis_keyspace:0 | foreign:no callid: nP89gWwjhsg_TSZrqu16nQ.. | deletionmark: no | created: 1712120716 | proxy:127.0.0.1:56207 | redis_keyspace:0 | foreign:no root@ip-kamailio:/etc/kamailio# I am new to kamailio and missing configuration. Suggest me the correct configuration to invoke recording. attached config and logs Regards, Pawan

1 0

How to relay SIP messages to specific registered user in Kamailio?
by o.atef＠fiberme.com 02 Apr '24

02 Apr '24

Hello guys, I'm recently new player in Kamailio SIP server and want to use it as SIP proxy server to Asterisk PBX server. first of all, I've successfully registered my Asterisk to Kamailio as user "test" to get its "location" because my Asterisk is behind NAT. Simply now I want to relay SIP messages to this user's location to receive them all in Asterisk. After I searched, I found this function rewritehost() but it needs to be a host or IP address, but in my case I want some way or function to relay to user "test" and then it well get the user location from Kamailio datebase automatically? I hope it's clear XD I tried to find a way to do MSQL query in kamalio.cfg to get the user contact from "location" table and then relay to the result but it did not work. I guess that I did it in wrong way but finally the Kamailio tutorials are so rare on google. Thanks in advance, Omar

5 10

Can we discuss the internals of Kamailio's routing logic and how to customize it for specific needs?
by Inextrix Technologies Pvt Ltd 28 Mar '24

28 Mar '24

This question digs into the heart of Kamailio's functionality. It asks about: Routing Logic: How Kamailio decides where to send SIP messages (calls, registrations, etc.) based on various factors like caller ID, destination number, or custom rules. Customization: How you can modify this built-in logic to fit your specific needs. This could involve using scripting languages, manipulating SIP messages, or defining custom routing rules. The discussion would be for developers who want to unlock Kamailio's full potential and tailor it for their unique VoIP deployments. Visit us for More Info : https://inextrix.com/services/kamailio-development/

2 1

Callee details not visible in dialog
by egemen ulus 28 Mar '24

28 Mar '24

Hi, I am wondering why callee info is not visible in dialog details: Here is the case: client-1 > kamailio > two asterisks > kamailio > client-2 registration is on asterisk. dialog module is enabled and I want to see the asterisk IP that is determined by dispatcher module. I have tried to put dlg_manage function even after dispatch function, but did not help. I have also tried to update to header before dlg_manage function, it didn't help either. Any way to have asterisk IP in dialog details. It shows only this: { h_entry: 483 h_id: 8644 ref: 1 call-id: v489LdfOFD5mnlHupQHOvw.. from_uri: sip:1515@12.12.12.12;transport=UDP to_uri: sip:2222@12.12.12.12 state: 5 start_ts: 0 init_ts: xxxx end_ts: xxxx duration: xxxx timeout: 0 lifetime: 3600 dflags: 512 sflags: 0 iflags: 0 caller: { tag: d0283007 contact: sip:1515@1.1.1.1:64966;transport=UDP cseq: 1 route_set: socket: udp:172.22.10.10:5060 } callee: { tag: <null string> contact: <null string> cseq: <null string> route_set: <null string> socket: <null string> } profiles: { } variables: { } } Regards Get Outlook for Android<https://aka.ms/AAb9ysg>

1 0

[Presence] Updating route information for SIP NOTIFY when re-SUBSCRIBE
by Denys Pozniak 26 Mar '24

26 Mar '24

Hello! In my architecture, SIP SUBSCRIBE messages can reach the Kamailio Presence server in several ways. And I noticed that re-SUBSCRIBEs messages do not update the record_route field in the active_watchers table (MySQL), so subsequent SIP NOTIFY messages do not inherit it and have the routes set of the initial SIP SUBSCRIBE message. Is there any way to change this behavior? modparam("presence", "db_url", DBURL) modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "expires_offset", 0) modparam("presence", "max_expires", 1800) modparam("presence", "db_update_period", 30) modparam("presence", "clean_period", 180) modparam("presence", "send_fast_notify", 1) modparam("presence", "pres_htable_size", 32) modparam("presence", "subs_htable_size", 32) modparam("presence", "publ_cache", 0) modparam("presence", "notifier_processes", 0) # kamailio -v version: kamailio 5.1.2 (x86_64/linux) -- BR, Denys Pozniak

3 4

RFC: move app_lua_sr to kamailio-archive
by Victor Seva 26 Mar '24

26 Mar '24

Hi, I would like to move app_lua_sr to kamailio-archive but I would like to hear if anyone out there has a solid reason not to do it. Since we have app_lua as KEMI for a long time I think is time to retire the old remains of app_lua Cheers -- ----------------------------------------------------------------- | ,''`. Victor Seva | | : :' : linuxmaniac(a)torreviejawireless.org | | `. `' PGP: 8F19 CADC D42A 42D4 5563 730C 51A0 9B18 CF5A 5068 | | `- Debian Developer | -----------------------------------------------------------------

1 0

Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

4 9

RTP Engine with Kamailio
by Vishal Pai 22 Mar '24

22 Mar '24

Hello Everyone. I am looking for documentation to have multiple RTP Engine servers connected with kamailio using db url and balance the load accordingly. Thanks

4 3

Kamailio to aid in seamless data center migration?
by Brad 21 Mar '24

21 Mar '24

I need to move a few dozen FreePBXen with some commercial modules running in individual VMs to a new data center. I’m trying to work out a plan to move the PBXes to the new data center in a way that will be transparent to the endpoints, or at the very least with the absolute minimum of downtime. Some of the installations are rather old, and there’s a handful of peculiarities on each, so the typical FreePBX backup/restore process hasn’t gone smoothly, at least in our tests. My current train of thought is to put a clone the VMs in the new data center and use Kamailio to route the SIP traffic to the new servers/IPs. I’ve never used it before, so I may be barking up the wrong tree, but it /appears/ to do what I’m suggesting. If so, I’m thinking I can install Kamailio on each VM, point it to the local asterisk/FreePBX initially, and clone the VM. Then, after the new instance is up and running, point Kamailio on the original VM to the cloned VM's asterisk, after which I can make appropriate DNS changes. Another option would be to stand up a single Kamailio server and redirect SIP traffic destined for individual asterisk servers to it at the router. Endpoints are mostly Yealink, and I’m not sure if they’ll feel the need to restart when the registration/SIP server’s IP changes, but a quick bounce when not in use isn’t the end of the world. I’d very much like to avoid having to send an update to each phone manually, but I can script a SIP NOTIFY if required. Anything stupid, wrong, ignorant, or just smelly about this tactic? Or, for that matter, any other suggestions?

2 1

RFC5626 Keep-Alive with CRLF
by Ihor Olkhovskyi 21 Mar '24

21 Mar '24

Hello, A bit generic question, does Kamailio supports CRLF keepalive per https://www.rfc-editor.org/rfc/rfc5626#section-4.4.1 ? It's more for tracking TCP connection states, as seems tcp_keepalive_enable is not super reliable especially in mobile networks. -- Best regards, Ihor (Igor)

3 5

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

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users