- sr-users - kamailio.org

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.

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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.

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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/>

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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

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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 ______________________________________________________

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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

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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 ______________________________________________________

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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.

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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

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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 ;)

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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

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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

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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

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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

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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

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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

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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

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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

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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

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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.

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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

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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

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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

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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

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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,

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