- sr-users - kamailio.org

Kamailio v5.6.6 Released
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, Kamailio SIP Server v5.6.6 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.5. 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 v5.6.6. Note that 5.6 is now the third last stable branch, v5.6.6 being the last planned released in 5.5.x series. The latest two stable branch are 5.7 and 5.8, with v5.8.2 being released a while ago. For more details about version 5.6.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-6-6-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 v5.6.6
by Daniel-Constantin Mierla 03 Jul '24

03 Jul '24

Hello, I am considering to release Kamailio v5.6.6 (out of branch 5.6) later this week (likely on Wednesday, July 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. This is going to be the last release out of branch 5.6 to mark the end of official/packaging maintenance. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Extracting and Handling Via Headers in failure_route
by sadik.oualla.mohamed＠gmail.com 02 Jul '24

02 Jul '24

Hello, I am currently working on a scenario where I need to use Kamailio in the failure_route to process SIP responses, specifically handling negative SIP replies (e.g., 486 Busy Here) that Kamailio may receive. Here is the topology: ##B2BUA (acts as a UAC with the Kamailio, because it sends the INVITE) -> Kamailio -> Callee## In my setup, Kamailio needs to extract all Via headers from the received SIP response in the failure_route, focusing particularly on the bottom Via header (from the B2BUA' UAC part). However, I am encountering difficulties in achieving this. Here is what I have tried so far: 1. To extract the Via headers, I used the following line of code in the failure_route: ```$var(viaHeaderValues) = $T_rpl($hdr(Via));``` However, this only returns the topmost Via header. 2. I attempted to get the first and last Via headers values with: ```$var(viaHeaderValueFirst) = $T_rpl($hdr(Via)[0]); $var(viaHeaderValueLast) = $T_rpl($hdr(Via)[1]);``` Unfortunately, this also only returns the topmost Via header, which contains the address of Kamailio. My goal is to make sure that when Kamailio sends the SIP reply to the UAC with send_reply method on the failure_route, it uses the extracted *Via header* of the incoming SIP response (only the bottom one). Here is an example scenario: - Received SIP Response in kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.4;branch=z9hG4bK9157.090080a91105c0ec6279bb56882d1dc8.0 Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` - Desired SIP Response from kamailio: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;received=200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` The current issue is that Kamailio modifies the Via header when sending the reply with send_reply method, resulting in: ``` SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 200.200.200.3;rport=5060;branch=z9hG4bKF1aaZ7ea7yc2m;received=200.200.200.3 From: <sip:caller@200.200.200.3>;tag=222222 To: <sip:callee@200.200.200.4:5060>;tag=4545454 Call-ID: 65695DSQ(a)200.200.200.3 CSeq: 1 INVITE Content-Length: 0 ``` Due to this change, the UAC does not correctly react to or understand the SIP response "SIP Reply follows the RFC standards, but my current UAC waits for a Via that looks like Via of the previous SIP messages in the structure of parameters.". It is crucial for the Via header to maintain the same important order such as the branch, received, and rport values as the previous exchanged SIP messages. Could you please provide guidance on how to force Kamailio to send the SIP reply using send_reply method with the extracted bottom Via header from the incoming SIP response? Any advice or examples on handling this situation would be greatly appreciated. Thank you for your time and assistance.

1 0

Kamailio v5.7.6 Released
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, Kamailio SIP Server v5.7.6 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.5. 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 v5.7.6. For more details about version 5.7.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/07/kamailio-v5-7-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note that 5.7.x is currently the previous stable release series, the latest is 5.8.x and v5.8.2 was released on June 12, 2024. 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

1 0

Releasing v5.7.6
by Daniel-Constantin Mierla 02 Jul '24

02 Jul '24

Hello, I am considering to release Kamailio v5.7.6 (out of branch 5.7) on Tuesday, July 2, 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

1 1

SRV in advertise?
by David Villasmil 01 Jul '24

01 Jul '24

Hello guys, I'm trying to use an SRV as an advertised address but kamailio fails to start. Is this not possible? Thanks, David Villasmil email: david.villasmil.work(a)gmail.com

4 7

SecSIPID Assistance
by Blake Ivey 01 Jul '24

01 Jul '24

Hi everyone. Wanting to see if someone could point me in the right direction. Still very knew to Kamailio but I am beginning to understand it better. I'm making an outbound proxy and have everything working well besides stir/shaken. I'm looking at the module page and have went back and forth with chatGPT and can't seem to figure this part out. I keep getting errors on the modparam lines. Obviously this is a self signed cert because I'm just testing. I am able to reach and download the cert from the Web server. Thank you for any assistance. # SECSIPID for Stir/Shaken modparam("secsipid", "private_key", "/etc/kamailio/secsipid/private.key") modparam("secsipid", "certificate", "/etc/kamailio/secsipid/cert.crt") modparam("secsipid", "authority_cert", "/etc/kamailio/secsipid/ca.crt") modparam("secsipid", "expire", 600) modparam("secsipid", "timeout", 2) route[STIRSHAKEN] { if (is_method("INVITE")) { if (!secsipid_add_identity("$fU", "$rU", "A", "", " http://myIPaddress.com/stir_shaken_cert.crt <http://myipaddress.com/stir_shaken_cert.crt>", "/etc/kamailio/secsipid/private.key")) { xlog("L_ERR", "Failed to sign call with ID: $ci - From: $fU\n"); send_reply("500", "Internal Server Error"); exit; } else { xlog("L_INFO", "Successfully signed call with ID: $ci - From: $fU\n"); } } # Relay the call after signing route(RELAY); }

3 8

removing Via Header with received=
by mosbah abdelkader 30 Jun '24

30 Jun '24

Hello All, I am trying to build a simple proxy that forwards SIP requests to a remote portaone server. But, kamailio is adding a Via header with received=@IP to replies got from portaone. How to disable this behavior as Asterisk (client in my setup) is not accepting this header. Thank you.

1 0

misctest module not found in kamailio debian packages
by Patrick Karton 29 Jun '24

29 Jun '24

Hello, how do i get misctest module in debian packages ? it seems not included in a package. is compilation the only way to get it ? im using kamailio 5.7.5 on debian (bullseye) 11.8 version: kamailio 5.7.5 (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, 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: unknown compiled with gcc 10.2.1 Thanks.

1 0

Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org
by Xenofon Karamanos 28 Jun '24

28 Jun '24

Any news on regarding the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thanks, Xenofon ________________________________ From: Xenofon Karamanos via sr-dev <sr-dev(a)lists.kamailio.org> Sent: Wednesday, June 19, 2024 11:51 To: sr-dev(a)lists.kamailio.org Cc: Xenofon Karamanos <xk(a)gilawa.com> Subject: [sr-dev] Ubuntu 24.04 Packages for Kamailio 5.8 on deb.kamailio.org Hey everybody, While attempting to use the official Ubuntu package for Kamailio version 5.7.4 on Ubunut 24.04, we encountered an issue where TLS functionality does not seem to work as expected but 5.8 branch does. Given this, I am inquiring about the availability and status of Kamailio packages for Ubuntu 24.04 on the official Kamailio repository at deb.kamailio.org. Are there any plans to support this version of Ubuntu with updated packages? If so, is there an estimated timeline for when these packages might become available? Thank you in advance for your assistance. Best regards, Xenofon

1 0

replies using the wrong protocol
by sarah.martin＠nextiva.com 27 Jun '24

27 Jun '24

I am extremely new at this, but trying to set up TLS with a carrier. TLS connection is good, Invite goes out, we get the 100 and the 200, but subsequent messages (ACK and BYE) are being sent with UDP and I cannot figure out how to get them to maintain the TLS transport. Any suggestions? I think this is the section I'm looking for. # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if (has_body("application/sdp")) { xdbg("rtpengine_manage loop-protect MANAGE_REPLY"); #rtpengine_manage("loop-protect"); } }

6 21

Kamailio cannot forward request - invalid sendtoparameters
by serkan celik 27 Jun '24

27 Jun '24

Hi all, I face a problem when trying to forward an UPDATE request to the IMS which communicates with UDP transport protocol on one side UDP and SCTP on other side. When I try to send UPDATE towards the UDP side of IMs I see following error on kamailio side and request cannot be forwarded Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: ERROR: <core> [core/udp_server.c:597]: udp_send(): sendto(sock, buf: 0x7fc180deb5f0, len: 1509, 0, dst: (172.18.0.20:5060), tolen: 16) - err: Invalid argument (22) Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: CRITICAL: <core> [core/udp_server.c:603]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net The UPDATE message which is tried to be forwarded is as follow: Here there are double rr one is UDP and one is sctp (Route: < sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>) ) That would be great if you can help why this UPDATE cannot be forwarded. By the way double rr is on on kamailio config Jun 25 16:19:18 prashant-0lk-sbc-1 /usr/sbin/kamailio[1223341]: DEBUG: <core> [core/forward.c:577]: forward_request(): Sending:#012UPDATE sip:alice@10.10.141.188;ob;sbc-id=xpllrk47odqh4pb2ofpib6vsym SIP/2.0#015#012Via: SIP/2.0/UDP 127.0.0.1:50601;branch=z9hG4bKddab.3554948ce3cfe321729448a2b082c2f3.0#015#012Via: SIP/2.0/UDP 127.0.0.1:7060;branch=z9hG4bKnHN8Q3ya#015#012To: < sip:alice@ims.wingcon.com>;tag=49IZjY2irfb#015#012From: < sip:bob@ims.wingcon.com>;tag=hfdqjj2pmui#015#012Call-Id: 3mzzecWt3Cbgvy#015#012CSeq: 1 UPDATE#015#012Max-Forwards: 70#015#012Contact: <sip:bob@10.10.141.188;ob;sbc-id=o7dpxe7qu2r34zwzio5pis2vqi>#015#012Content-Type: application/sdp#015#012Route: <sip:mt@172.18.0.20;lr;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, <sip:mt@172.18.0.20;lr;transport=sctp;did=6aa.aaa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.20;lr;transport=sctp;did=6aa.9aa;ftag=49IZjY2irfb;r2=on>, < sip:mo@172.18.0.120;lr>#015#012Session-Expires: 180;refresher=uas#015#012Supported: replaces, 100rel, timer, norefersub#015#012Min-SE: 90#015#012P-Charging-Vector: icid-value=e277783432fd11efba03fa163eed425f;icid-generated-at=172.18.0.120;orig-ioi=172.18.0.120;term-ioi=172.18.0.120#015#012User-Agent: PJSUA v2.14 Linux-5.4.0.172/x86_64/glibc-2.35#015#012Content-Length: 409#015#012#015#012v=0#015#012o=- 3928313948 3928313950 IN IP4 172.18.0.120#015#012s=pjmedia#015#012b=AS:84#015#012t=0 0#015#012a=X-nat:0#015#012m=audio 30106 RTP/AVP 8 96 97 121 120#015#012c=IN IP4 172.18.0.120#015#012b=TIAS:64000#015#012a=ssrc:1986855169 cname:1fca52e812ced466#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 AMR/8000#015#012a=rtpmap:97 AMR-WB/16000#015#012a=rtpmap:121 telephone-event/16000#015#012a=fmtp:121 0-16#015#012a=rtpmap:120 telephone-event/8000#015#012a=fmtp:120 0-16#015#012a=sendrecv#015#012a=rtcp:30107#015#012. Regards Serkan

1 0

Handling final response retransmissions after a reInvite
by hollis.rhonda＠gmail.com 26 Jun '24

26 Jun '24

I have a scenario where Kamailio is receiving a retransmit of a 200 OK to a late offer INVITE after it has sent the ACK, and after it has begun to handle a reInvite from the calling party. This results in the following commands to the rtpengine: Offer (reInvite), Offer (200 retransmit), Answer (ACK to retransmitted 200). At this point, I see STUN binding errors both in Chrome, where the webRTC client (called party) is running, and in the rtpengine logs. ERROR:port.cc(498)]: Received STUN BINDING error response: class=4 number=1 reason='Unauthorized' [core] STUN authentication mismatch from x.x.x.x:63396 [ice] Received ICE/STUN response code 487 for candidate pair dl2efuRG06eK4nFs:2713745946:1 from x.x.x.x::64509 to x.x.x.x [ice] ICE role change, now controlled [ice] Recalculating all ICE pair priorities [ice] Triggering check for dl2efuRG06eK4nFs:2713745946:1 [ice] ICE/STUN response with unknown transaction received (from x.x.x.x:64509 on interface x.x.x.x:16184) So, briefly, the call flow up to the point of error looks like this: INVITE w/o SDP 200 OK w/SDP, Offer to rtpengine ACK w/SDP, Answer to rtpengine reInvite w/SDP, Offer to rtpengine retransmission of 200 OK, Offer to rtpengine ACK to retransmitted 200, Answer to rtpengine I have dialog tracking enabled in my config, and it seems like Kamailio is behaving as expected when the retransmitted 200 and ACK are forwarded on, but I'm curious if other people have run into a scenario like this and how they have dealt with it.

1 0

Need help on how to implement presence_xml
by Michel Pelletier 25 Jun '24

25 Jun '24

Hello, I need to implement Event: presence, application/pidf+xml as described in RFC-3856. I understand I need to use the presence_xml module and generate PUBLISH requests to update the information and ensure that the NOTIFY get sent. I know nothing abour XCAP or wether I need it or not. Any information source out there that could help me? Regards, Michel Pelletier

1 0

TLS: client doesn't check certificate CN
by Marat Gareev 20 Jun '24

20 Jun '24

Hello, I noticed that Kamailio can route either to IP and FQDN. This means the server certificate CN is not checked by the client. How to enable something like the 'verify_peer' option? Thanks

1 0

Possibility to print values of Core Keywords in Kamailio configuration file
by sadik.oualla.mohamed＠gmail.com 20 Jun '24

20 Jun '24

Hello everyone, I have a question regarding the usage of core keywords, specifically the possibility of printing their values in log messages. While I understand that core keywords such as 'dst_port' and 'af' etc... can be accessed directly in if conditions, I am interested to know if there is a way to print the values of these core keywords for logging purposes. Thank you.

4 6

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Wed, 19 Jun 2024 11:54:05 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > What you should expect using this method is blocking while the exec runs > which could run you out of child processes while they complete. I'd also > expect to see an "unusual amount of CPU activity" which will be the result > of the forked shells being created and destroyed. > > All of that being said, yes it works. It's architecturally overly simple > and very inefficient. If your load is tiny, it might be ok. Anything > moderate like Henning and others have said, will use an inordinate amount > of cpu. I would not personally risk it. > > Thank you Brett, Henning and Alex Sadly the load won't be tiny and cannot risk it. For now I'll deploy a local nginx and fpm to run the scripts, which are in php, and will use the http_async method. Sadly we have to "http all the things" nowadays. Let's see if I'm able to write down all the logic tomorrow and do some testing. cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 0

Adding custom headers to HTTP server response xhttp
by Sasivarunan 19 Jun '24

19 Jun '24

HI Team, I use tls and xhttp module to enable websocket interface for webrtc client to connect. I wanted to add "Strict-Transport-Security" header to xhttp response to add security. But currently I don't see a way to add it anywhere. I can see only server_header as an option, But this will get added to SIP messages as well. It 'd be great if anybody has a suggestion around this. Thanks Varunan

2 2

Re: exec concurrency
by Jon Bonilla (Manwe) 19 Jun '24

19 Jun '24

El Tue, 18 Jun 2024 13:54:41 -0500 Brett Nemeroff <brett(a)voicefoxtelephony.com> escribió: > Just want to add that exec is heavy and slow. I would not recommend it. > > Is there a reason you want to do this over http-ifying your script and > using async? > > Not really. The scripts are provided. Maybe I'm against "http all the things" which seems to be what everyone does nowadays. Why would http client be lighter than executing a script? That would be a design problem, right? -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 1

exec concurrency
by Jon Bonilla (Manwe) 18 Jun '24

18 Jun '24

Hi all I've been working with async http client and async db queries in the past but now I have to execute scripts and store the return values in avps. Since I have no experience with that, I wonder how to achieve concurrency with that scenario. Is there a way to exec async or suspend until the exec has finished and resume later? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 2

Psresence dialoginfo question
by Stefan Mititelu 17 Jun '24

17 Jun '24

Hello, I am doing BLF but I want different users to be able to setup different dialog info initial states. My question is if there's an already implemented way for doing that? I've looked over docs and C code so far and this is what I've found: 1. modparam("presence_dialoginfo", "force_dummy_dialog", 1) => will force sending initial NOTIFY with dialog state "terminated" 2. Function calls: dlginfo_agg_nbody()->dlginfo_agg_nbody_empty() which will build the pre-defined xml body with that "terminated" state. If not implemented so far, maybe this can be used to set a different initial state for a user (e.g. export a cfg function or take that info from DB?) Any opinions on this? Thank you, Stefan

2 2

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

1 1

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!

1 0

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

3 5

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

1 0

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