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37510 discussions

db_mysql startup warnings
by Juha Heinanen 17 Jun '24

17 Jun '24

When I start Kamailio master version that has been build on Debian 12 with default-libmysqlclient-dev depending on libmariadb-dev-compat, I get several warnings like this: 2024-06-17T14:48:27.702827+03:00 siika /usr/bin/sip-proxy[297246]: WARNING: db_mysql [km_my_con.c:187]: db_mysql_new_connection(): opt_ssl_ca option not supported by mysql version (value (null)) - ignoring How can I prevent them? -- Juha

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How to Check Source Port in Application Layer with Kamailio
by sadik.oualla.mohamed＠gmail.com 16 Jun '24

16 Jun '24

Hello kamailio lovers, I am currently working on a Kamailio configuration where I need to check the source port of incoming SIP messages. Initially, I used the $sp pseudo-variable to get the source port, which worked perfectly while testing with SIPp. Here is the snippets code that I use " if ($sp != 5080 && $sp != 8080 && $sp != 5060){ sl_send_reply(403, "forbidden"); xlog("this $rm received, is forbidden\n"); exit; } " However, I encountered an issue when I switched to using sipsak for testing. It seems that sipsak uses a randomly assigned port for the actual network connection, even though it sets the source port in the SIP headers (e.g., Via, To and Contact headers) to a specified port using the -l option. Here is the sipsak command that I use "sipsak -vv -s sip:jack@192.168.1.2:5060-l 5080" My goal is to have Kamailio check the source port as specified in the application layer, rather than the transport layer port (i.e., the ephemeral port assigned by the OS randomly like 54691, 33017...). Could anyone guide me on how to properly configure Kamailio to extract and check the port from the application layer or SIP headers, or am I doing this in the wrong way? Any advice or examples would be greatly appreciated. "I am not a native english speak, so I use chatGPT to make my questions clear" Thank you!

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topos module issue on hangup
by Arun K R 13 Jun '24

13 Jun '24

Hello, I am using topos module to hide Record-Route and Route header as per the requirement from carrier. But the topos module hides, contact, via and routing headers also. Due to this reason when I am getting bye from the carrier I am not able to route the bye back to the agent. Bye from Agent to carrier is working fine. Bye from carrier to my internal network interface is not routing to client through public interface. I assume this may be because Via / Contact header has been hidden and changed by the topos module with the invite please help me for this Thanks Arun

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Kamailio v5.8.2 Released
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, Kamailio SIP Server v5.8.2 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.8.x. Deployments running previous v5.8.x versions are strongly recommended to be upgraded to v5.8.2. For more details about version 5.8.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/06/kamailio-v5-8-2-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

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Releasing v5.8.2
by Daniel-Constantin Mierla 12 Jun '24

12 Jun '24

Hello, I am considering to release Kamailio v5.8.2 (out of branch 5.8) on Wednesday, June 12, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

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New features in v.5.8
by Leonid Fainshtein 10 Jun '24

10 Jun '24

Hi, In the "Kamailio v5.8.0 Release Notes" I see: "SIP overload control support for server (RFC7339)". Where can I find some more details about the feature? Also, the similar question about the new "influxdbc" module. What are the possible use cases for this module? Thank you, Leonid Fainshtein Xorcom Ltd

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handling 300 redirect requests
by Rob Green 09 Jun '24

09 Jun '24

I am trying to proxy a request from server A to server C with me being server B. i send the invite from A to B, then i need to send to server R which returns back to me a 300 redirect, which i then need to take the answer from the Contsct and then proxy the original invite from A to C. server A can not handle 3xx responses and must be proxied. I havent been able to find any examples where the 300 is intercepted and a new invite to the target is sent. i have the following code but it results in no new INVITE and the original session is hung until it is canceled. Any idea what i may be doing wrong? In the example below i am just hardcoding the uri for the new invite instead of getting from the Contact as a debug exercise. I am using Kamailio 5.7.3 route[HANDLEREDIRECT] { xlog("L_INFO", "DBG: HANDLEREDIRECT"); msg_apply_changes(); $var(test) = $hdr(Contact); xlog("L_INFO","DBG: redirect headers [$var(test)]: from=[$fu] from_user=[$fU], to=[$tu] to_user=[$tU]"); # send to gateway $var(new_uri) = "sip:" + $tU + "@" + "gateway"; if ($var(new_uri) != "") { xlog("DBG: Redirecting to: $var(new_uri)\n"); $ru = $var(new_uri); $tu = $ru; record_route(); route(RELAY); exit; # i have also tried using #t_newtran(); # and #t_relay(); } else { xlog("DBG: No contact header found in 300 response\n"); } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("DBG: new branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "DBG: manage branch"); route(NATMANAGE); return; } onreply_route[MANAGE_REPLY] { xdbg("DBG: incoming reply\n"); xlog("L_INFO", "DBG: manage reply"); if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: got 3xx redirect"); route(HANDLEREDIRECT); drop; } return; } I also tried using uac_redirect module, but it stalls the incoming invite also, and i cant figure out how to modify the uri from the contact before sending (to add params) # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "DBG: following 3xx redirect"); get_redirects("3:1"); t_relay(); exit; } } Rob Green Chief Information Officer (CIO) Tel: 1-352-218-1490 Cell: 1-407-620-0932 Email: rob.green(a)nuwave.com Web: www.nuwave.com This communication is the Property of NUWAVE Communications and may contain confidential or privileged information. Unauthorized use of this communication is strictly prohibited and may be unlawful. If you have received this communication in Error, Please immediately notify the sender by reply e-mail and destroy all copies of the communication and any attachments.

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Setting Key/Value Pairs at Dialogue Level
by Pavan Kumar 08 Jun '24

08 Jun '24

Hi everyone, I am creating a dialogue when I receive INVITE. Is there a way to set key/value pair like attributes at dialog level, so that, when I receive BYE message I can retrieve that information and do some processing? Thank you, Pavan Kumar

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Recommendation for a signaling B2BUA to work with Kamailio on Ubuntu
by Benoît Panizzon 07 Jun '24

07 Jun '24

Hi I wonder if anyone could maybe recommend what to use as signaling B2BUA to work with Kamailio to handle spiralling / looping calls and to reduce the number of Route, Via Header and message size, which cause issue with a large number of CPE. Requirements: * New CallID needs to be generated * Handling Route and Via * ~ 2000 concurrent calls. * Preserving custom SIP X- header * Routing NOTIFY for MWI etc. Handling RTP is not required! What I have been looking at more closely: = FreeSwitch = Got stuck trying to solve the dependencies to get it to run/build on ubuntu jammy. Has anyone experience in running it in a docker container? Will it perform? = Sippy = Very small, very simple, unfortunately removes customer SIP headers and does not forward NOTIFY. (It's python, so maybe this could be easily implemented/fixed). = SEMS = Does not seem to be actively developed in the last, couple of years. Looks quite bloated to me for what I need. = Asterisk = Each customer header has to be copied, wants to handle RTP and bridge by issuing Re-Invites. Not sure how to route stuff like NOTIFY. Also from experience I fear asterisk would translate sip reply codes (like 608 and 607) it does not know to some 4XX default. -- 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 ______________________________________________________

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Filtering codes in db acc
by Jon Bonilla (Manwe) 06 Jun '24

06 Jun '24

Hi all I'm trying to make kamailio not to insert a row in missed_calls table when reply code is 407. But even if I set the "failed_filter" modparam the 407 row is inserted. The acc mmodparams are: modparam("acc", "db_table_acc", "kam_acc") modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACC_MISS) modparam("acc", "db_url",DBURL) modparam("acc", "db_insert_mode", 1) modparam("acc", "time_mode", 2) modparam("acc", "time_attr", "time_hires") modparam("acc", "failed_filter", "401,407") modparam("acc", "db_extra", "whatever" db is mysql. Kamailio version is 5.7.4 When I send a call that receives a 407 and after auth a 480 y get both rows in missed_calls table. Log: DEBUG: acc [acc_mod.c:404]: mod_init(): acc db table initialized to: kam_acc DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 0 = 401 DEBUG: acc [acc_mod.c:362]: parse_failed_filter(): failed_filter 1 = 407 DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cec807480) event type 512, reply code 407 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 407 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for t(0x7f7cebf59bc0) event type 512, reply code 486 DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc state - code: 486 flags: 0x220 DEBUG: acc [acc_logic.c:449]: should_acc_reply(): failed acc is off am I configuring this wrong? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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