June 2025 - sr-users - kamailio.org

Why won't this work
by Fernando Lopes 05 Jun '25

05 Jun '25

Kamailio isn’t working on my Azure VM, even though it’s set up to use port 5060. The network rules on Azure have both inbound and outbound traffic open for all protocols. I also tried using advertised_address, but that didn’t help either. It looks like no traffic is even hitting Kamailio — none of the routing blocks are triggered. Kamailio.cfg alias=<PUBLIC-IP> listen=udp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 listen=tcp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 Sip Message: 2025/06/04 15:50:37.870679 64.16.250.10:5060 -> 10.0.0.4:5060 INVITE sip:+351200600020@<PUBLIC-IP>:5060 SIP/2.0 Record-Route: <sip:64.16.250.10;r2=on;lr;ftag=DpHvv09240egQ> Record-Route: <sip:10.255.0.2;r2=on;lr;ftag=DpHvv09240egQ> Via: SIP/2.0/UDP 64.16.250.10;branch=z9hG4bKd59a.fb67bf395ea20488a7a5ebf76b9b5685.0 v:SIP/2.0/UDP 10.239.196.24:6000;received=10.239.196.24;rport=6000;branch=z9hG4bKy9ZvBZcgretvr Max-Forwards:48 f:"+351908020050"<sip:+351908020050@sip.telnyx.com>;tag=DpHvv09240egQ t:<sip:+351200600020@<PUBLIC-IP>:5060> i:122d71bb-4f3d-473f-8853-9137023f1f14 CSeq:99972326 INVITE m:<sip:mod_sofia@10.239.196.24:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:202 P-Asserted-Identity:"+351908020050"<sip:+351908020050@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1749019739 1749019740 IN IP4 50.114.147.4 s=Telnyx c=IN IP4 50.114.147.4 t=0 0 m=audio 32498 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:32499 IN IP4 50.114.147.4 a=ptime:20

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Kamailio Deployment in AWS
by gcampos＠momenta.network 04 Jun '25

04 Jun '25

Hi community, I'm not sure if anyone has ever tried that before, but I'm trying to run Kamailio in AWS with RTPengine. I successfully installed both Kamailio and RTPengine on my EC2 instance and I'm able to make my devices register to it. However, when I make a call from one of my devices, the call is muted. As I analyze my kamailio.cfg file, I see that the "route[NATMANAGE]" is likely the cause of the mute call. I think that none of the conditions below are met, therefore, the flags are not set. Because of that, the last condition is met, thus triggering the "return" and exiting from the route[NATMANAGE] prematurely without even reaching the section where I define the RTPengine modifications. Not only that, but if I remove/comment out the "return", then the RTPengine is triggered, but there call doesn't connect. I wonder why this is happening and if there's any known workaround. # RTP relaying management and signaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO", "NATMANAGE path was followed\n"); #!ifdef WITH_NAT xlog("L_INFO", "WITH_NAT Triggered\n"); if (is_request()) { xlog("L_INFO", "is_request path was followed\n"); if (has_totag()) { xlog("L_INFO", "has_totag path was followed\n"); if (check_route_param("nat=yes")) { xlog("L_INFO", "nat=yes param detected, setting FLB_NATB\n"); setbflag(FLB_NATB); } } if (nat_uac_test("3")) { xlog("L_INFO", "nat_uac_test(3) matched, setting FLT_NATS + FLB_NATB\n"); setflag(FLT_NATS); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xlog("L_INFO", "No NAT flags set, skipping NAT handling\n"); return; } #!ifdef WITH_RTPENGINE xlog("L_INFO", "WITH_RTPENGINE Path followed\n"); if (has_body("application/sdp")) { if (is_method("INVITE")) { xlog("L_INFO", "IS_METHOD_INVITE Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } else if (is_reply()) { xlog("L_INFO", "IS_Reply Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; }

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Will kamailio handle sip messages if Request contains public Ip
by Fernando Lopes 04 Jun '25

04 Jun '25

I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. The SIP message received looks like this: INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0 Will Kamailio handle this SIP message, or will it ignore it? Right now, Kamailio is not doing anything with these messages. If it’s supposed to handle them, can someone help me fix this issue?

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Re: Will kamailio handle sip messages if Request contains public Ip
by airsay＠duck.com 04 Jun '25

04 Jun '25

It should. You need to add a listen and alias in the kamailio.cfg file listen=udp:1x5.2x6.1x8.1x3:5060 alias=1x5.2x6.1x8.1x3 BR, airsay > On 4 Jun 2025, at 10:46, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. > The SIP message received looks like this: > INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0

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adding codecs to rtpengine
by Jeff Brower 03 Jun '25

03 Jun '25

Hi all, Is there a documented procedure for adding codecs to rtpengine ? As I understand it so far, we would adapt rtpengine encoder and decoder interfaces to the voplib API (documented at https://github.com/signalogic/SigSRF_SDK/blob/master/codecs_readme.md) allowing use of voplib shared object libs. I've been searching (e.g. adding codecs to rtpengine site:kamailio.org) but have not found a step-by-step procedure yet. Thanks for any advice / guidance. -Jeff

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event_route[xhttp:request] is not executed
by Володимир Іванець 03 Jun '25

03 Jun '25

Hello! I'm doing some testing with http requests but got stuck at the beginning. event_route[xhttp:request] does not seem to be executed. I'm getting the following in the log with the most basic configuration. Am I missing something? Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1238]: tcpconn_new(): on port 46328, type 2, socket 23 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1630]: tcpconn_add(): hashes: 1391:1425:2615, 2 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:367]: io_watch_add(): processing io_watch_add(0xb22d80, 23, 2, 0x7f7a63877f30) - fd_no=17 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xb22d80, 23, -1, 0x0) fd_no=18 called Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4671]: handle_tcpconn_ev(): sending to child, events 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4335]: send2child(): selected tcp worker idx:1 proc:10 pid:9158 for activity on [tcp:127.0.0.1:8089], 0x7f7a63877f30 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1782]: handle_io(): received n=8 con=0x7f7a63877f30, fd=6 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:627]: tcp_read_headers(): no clen, p=A Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) ERROR: <core> [core/tcp_read.c:1555]: tcp_read_req(): bad request, state=7, error=4 buf: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: parsed: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1562]: tcp_read_req(): received from: port 46328 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): received from: ip127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1704]: release_tcpconn(): releasing con 0x7f7a63877f30, state -2, fd=6, id=2 ([127.0.0.1]:46328 -> [127.0.0.1]:8089) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1705]: release_tcpconn(): extra_data (nil) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3745]: handle_tcp_child(): reader response= 7f7a63877f30, -2 from 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3668]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3676]: tcp_emit_closed_event(): no callback registering for handling TCP closed event #!KAMAILIO children = 4 tcp_children = 4 debug=3 log_stderror=yes memdbg=5 memlog=5 fork=yes #!substdef "!LOCAL_UDP!udp:127.0.0.1:5060!g" #!substdef "!LOCAL_HTTP!tcp:127.0.0.1:8089!g" listen=LOCAL_UDP listen=LOCAL_HTTP loadmodule "xlog.so" loadmodule "pv.so" loadmodule "ctl.so" loadmodule "sl.so" loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } Thank you very much!

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Kamailio Dispatcher for Docker Asterisk
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello everyone, I have Kamailio running on port 5060, and I'm running Asterisk inside a Docker container. I'm trying to connect to Asterisk via port 5061. Could someone please advise what I should add to my dispatcher list? Currently, I have the dispatcher.list entry: 1 sip:10.0.0.5:5061;transport=tcp 0 Here’s the network info for my Kamailio host (eth0 interface): inet 10.0.0.4/24 brd 10.0.0.255 scope global eth0 And here are the relevant port mappings from Docker: 5038/tcp -> 0.0.0.0:5038 5060/tcp -> 0.0.0.0:5061 5060/udp -> 0.0.0.0:5061 ... My main question: Is the dispatcher entry sip:10.0.0.5:5061;transport=tcp correct if Asterisk is listening on port 5061 in the container? Do I need to adjust anything on the Kamailio side or change the dispatcher entry format? Thanks in advance for any help!

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Small Question
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello Everyone, I have a cloud vm with public ip: 135.2xx.1xx.1xx:5060 and locally 10.0.0.4 When I receive a SIP message the invite looks like this, it gets stuck here. SIP Invite: 2025/06/03 16:24:58.249284 192.76.120.10:5060 -> 10.0.0.4:5060 INVITE sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060 SIP/2.0 Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=765jrrKarBe3F> Record-Route: <sip:10.255.0.1;r2=on;lr;ftag=765jrrKarBe3F> Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bKbb37.28332dbc5901e729308403cd5e049713.0 v:SIP/2.0/UDP 10.239.44.224:6000;received=10.239.44.224;rport=6000;branch=z9hG4bKD4UtjN52Qrm9D Max-Forwards:55 f:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.com>;tag=765jrrKarBe3F t:<sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060> i:c64314db-75b9-4029-8147-be3f3d77da2a CSeq:99930157 INVITE m:<sip:mod_sofia@10.239.44.224:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:205 P-Asserted-Identity:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1748940128 1748940129 IN IP4 50.114.150.25 s=Telnyx c=IN IP4 50.114.150.25 t=0 0 m=audio 27770 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:27771 IN IP4 50.114.150.25 a=ptime:20 Kamailio Code: #!define DS_LIST "/etc/kamailio/dispatcher.list" #!define LISTEN_UDP_PUBLIC udp:135.2xx.1xx.1xx:5060 #!define LISTEN_UDP_PRIVATE udp:10.0.0.4:5060 #!define LISTEN_TCP_PUBLIC tcp:135.2xx.1xx.1xx:5060 #!define LISTEN_TCP_PRIVATE tcp:10.0.0.4:5060 #!define EXTERNAL_IP "10.0.0.4" #!define KUBERNETES_NETWORK "172.19." #!define FLAG_TELNYX_CALL 12 #set module path mpath = "/usr/lib/x86_64-linux-gnu/kamailio/modules" ####### Global Parameters ######### #mhomed=1 # doesn't work with virtual IPs #!ifdef WITH_DEBUG debug=4 #!else debug=2 #!endif log_stderror=no memdbg=5 memlog=5 #log_facility=LOG_LOCAL6 log_facility=LOG_LOCAL0 fork=yes children=4 auto_aliases=no #listen=LISTEN_UDP_PUBLIC listen=LISTEN_UDP_PRIVATE listen=LISTEN_TCP_PRIVATE ####### FLAGS ###### #!define FLAG_FROM_ASTERISK 10 #!define FLAG_FROM_PEER 11 sip_warning=no

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Kamailio as Proxy only
by disa_369＠msn.com 03 Jun '25

03 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from Avaya PBX -------> Kamailio ----> Video System. We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. I got questions. 1. Do we need install maria_db for that? 2. Do we need create subscriber\peer on Kamailio? 3. Do we need add modules for that? And main question what to add in kamailio.cfg is? Thx!!!!!!!!!

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Kamailio Regex
by Timothy Delin 03 Jun '25

03 Jun '25

Hi everyone, I need your advice. I am extracting a SIP Body message from a User Agent Client using the regex as such "$var(calling_party)= $(var(body){re.subst,/.*Calling-party:\s*sip:(\+[0-9]+)@.*/\1/});" And I tried to log it as "xlog("L_INFO", "Extracted number 1: $var(calling_party) <><><>\n");" However, the output in my logs is only showing as below, where I cannot see any value after my variables. "1(6) INFO: {1 10 SUBSCRIBE 5017ae21204c72a3} <script>: Extracted number 1: +6512345678" I suspect this is the reason why when I tried to use this variable in a redis_cmd, it returns as null as the variable may have impacted the command input to Redis. I tried to change the $var(calling_party)="+6512345678" and it is working perfectly fine Best regards, Timothy

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