- sr-users - kamailio.org

AEM Implementation Services,
by surekhatech.seo＠gmail.com 02 May '24

02 May '24

As an Official Adobe Bronze Solution Partner, we specialize in AEM development services, consulting, implementation, migration, and cloud services for your business.

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Question: if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio
by jennifer yang 02 May '24

02 May '24

Hello Can anyone help answer these questions: - if SIP Timer L and Timer M from RFC6026 has been defined at Kamailio (version v5.5.x)? Timer L: Wait time for accepted INVITE request retransmits Timer M: Wait time for retransmission of 2xx to INVITE or additional 2xx from other branches of a forked INVITE - if no specific Timer defined, which timer will be used instead? thanks br jenni

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Issue with serial call forking to same destination
by Mathias 30 Apr '24

30 Apr '24

Hi I've an issue with serial call forking and I hope somebody can help me... What I'm trying to reach: We are using Kamailio in a dynamic network with multiple nodes. The dialplan is implemented using the KEMI interface. The call routing is completely dynamic. At the moment, we are having call groups, which may contain 1 or more recipients. If the number of this call group is called, Kamailio is doing a parallel forking to all group participants. Now I'm trying to extend the feature, that if nobody hooks up, the call is forwarded to another number. This has been implemented with serial forking. So the dialplan adds branches for all call group members with a q value of 1 and the last branch is the failover number with a lower q value. The branching and failure route works fine. The problem: * I have a call group (G Bob 900101), with one participant (Bob 200102). * The group has a fail over number (if Bob doesn't hook up), which points again to Bob. I know that this use case doesn't make much sense, but it has been simplified to show the problem. The failover number can be anything also another call group which may contain one or more recipients from the first group again. * Alice (200101) calls the call group * The first node (10.0.1.1/10.64.0.1) receives the call and forks it to branch 1 Bob (202102) with q value 1.0 and to a second branch Bob (202102) with q value 0.5. fr_timer is set to 10s * The call is forwarded to node 2 (10.0.2.1/10.64.0.2), where Bob is registered. * The call is forwarded to Bobs device (Grandstream GXV3350) and the phone is ringing. * After 10s, the first node detects the 408 Request Timeout and switches to the second branch * The invite is sent to node 2 and Bobs device again. * Node 1 also sends a CANCEL to node 2 and Bobs device for the first branch. * The call in Bobs device is terminated * Now the problem, the device does not react to the second INVITE. And after the fr_inv_timer expires, the call is terminated with a 408 Request Timeout. I can see that the second INVITE contains another branch ID <ID>.1 compared to the first INVITE, which had <ID>.0. The CANCEL is also sent with <ID>.0. I've also tried with a Cisco 8861 3PCC and it doesn't work either. Compared to the Grandstream, the Cisco device returns an error 500 upon the second INVITE. Since both devices do not accept what I'm trying, I guess I'm doing something wrong... At the moment I'm a bit lost, what the problem could be. Does somebody have an idea what it is and how I could solve it? Following two SIP dumps from node 1 and node 2: Node 1: |Time | 10.0.1.128 | 10.64.0.1 | | | | 10.0.1.1 | | 10.0.2.1 | |2024-01-31 14:17:07.792784| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.793060| 407 Proxy Authentica | | |SIP Status 407 Proxy Authentication Required | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.796613| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2950 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.799045| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(47276) ------------------> (5061) | | | |2024-01-31 14:17:07.836471| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:07.865219| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:08.036334| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.084421| | | 180 Ringing |SIP Status 180 Ringing | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:08.096834| 180 Ringing | | |SIP Status 180 Ringing | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:18.099644| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.100099| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:18.259375| | | 100 trying -- your c |SIP Status 100 trying -- your call is important to us | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.274841| | | 200 canceling |SIP Status 200 canceling | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.296942| | | 487 Request Terminat |SIP Status 487 Request Terminated | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:18.298630| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.281602| | | 408 Request Timeout |SIP Status 408 Request Timeout | | | |(5061) <------------------ (5061) | |2024-01-31 14:17:20.283433| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5061) | |2024-01-31 14:17:20.307343| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(47276) <------------------ (5061) | | | |2024-01-31 14:17:20.309097| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:<sips:900101@10.0.1.1:5061> CSeq:2951 | |(47276) ------------------> (5061) | | | Node 2: |Time | 10.64.0.1 | 10.64.0.2 | | | | 10.0.2.1 | | 10.0.2.128 | |2024-01-31 14:17:07.869305| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:08.036009| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:08.050848| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:08.054357| | 100 Trying| | |SIP Status 100 Trying | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.075422| | 180 Ringing | |SIP Status 180 Ringing | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:08.083887| 180 Ringing | | |SIP Status 180 Ringing | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.100359| INVITE SDP (g711A g7 | | |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> Call-ID:2102072978-5060-296@BA.A.B.BCI CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.258947| 100 trying -- your c | | |SIP Status 100 trying -- your call is important to us | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.273819| | | INVITE SDP (g711A g7 |SIP INVITE From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274460| CANCEL | | | |SIP Request CANCEL CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:18.274550| | | CANCEL | |SIP CANCEL From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.274615| 200 canceling | | |SIP Status 200 canceling | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.277438| | 200 OK | | |SIP Status 200 OK | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.279335| | 487 Request Terminated | |SIP Status 487 Request Terminated | | |(5061) <-------------------------------------- (5060) | |2024-01-31 14:17:18.280785| | | ACK | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | | | |(5061) ------------------> (5060) | |2024-01-31 14:17:18.296312| 487 Request Terminat | | |SIP Status 487 Request Terminated | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:18.298900| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | |2024-01-31 14:17:20.280965| 408 Request Timeout | | |SIP Status 408 Request Timeout | |(5061) <------------------ (5061) | | | |2024-01-31 14:17:20.283730| ACK | | | |SIP ACK From: "Alice" < sips:200101@10.0.1.1:5061> To:"Bob" <sips:900101@10.0.1.1:5061> CSeq:2951 | |(54454) ------------------> (5061) | | | At the moment, I'm using Kamailio 5.5.4, but I also did a short try with a newer version (I guess it was 5.7.4). Both versions showed the same issue. Best regards Mathias

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Sudden TCP errors and shared memory spikes
by David Cunningham 29 Apr '24

29 Apr '24

Hello, We have a production Kamailio 5.2.7 server which has suddenly started reporting spikes in the shared memory use. When looking into this, we noticed that at the same time as the memory spikes Kamailio also logs a huge amount of errors like the following. Can anyone help with an explanation for what's going on? The xx.xx.52.202 address is the Kamailio server itself. We record historical numbers of network connections for the server overall, and that doesn't show any increase in the number of TCP connections, either at the time of the spikes or as a general trend. Thanks in advance! Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2703]: tcpconn_1st_send(): connect xx.xx.52.202:42726 failed (RST) Connection refused Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: <core> [core/tcp_main.c:2711]: tcpconn_1st_send(): xx.xx.52.202:42726: connect & send for 0x14bbf44c9f40 failed: Connection refused (111) Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed Apr 26 09:23:16 vpbx11 /sbin/kamailio[44272]: WARNING: <core> [core/tcp_main.c:1149]: tcp_do_connect(): xx.xx.52.202:42726: could not find corresponding listening socket for xx.xx.52.202, using default... -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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tls_threads_mode and app_python3 module
by Ihor Olkhovskyi 29 Apr '24

29 Apr '24

Hello, Just to confirm, as I quickly checked into a code of 5.8.x branch, there were no modification on app_python3 module to comply with tls_threads_mode mechanism? Asking this beforehand posting a bug, that if using a quite simple script into module, with tls_threads_mode = 2 Kamailio just hangs at the start and become unresponsive. And by simple I mean something like this class PyKamailio: def __init__(self): self.test = 1 def child_init(self, rank): """Executed by app_python module when a new worker process (child) is initialized by Kamailio.""" return 1 def mod_init(): """ Executed by app_python module when it is initialized by Kamailio.""" return PyKamailio() with tls_threads_mode = 1 and 3 it loads, but if add some HTTPS calls inside the script (even not the calls, but importing the methods, that are making calls with import), it just start failing on TLS load, despite that on packets processed (OPTIONS) no python_exec() is called, just from the fact of having HTTPS calls into Python script. -- Best regards, Ihor (Igor)

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RTP Engine with Kamailio hold issue
by Vishal Pai 29 Apr '24

29 Apr '24

Hello Everyone I am using kamailio with RTP engine for wss to udp to connect my asterisk server. It all works well like wss to UDP registration via kamailio to asterisk. Here I am facing an issue with call hold i.e if I hold the call from B leg it gets disconnected. Any advice why it get failed and hangups Thanks

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Error by installing new Rel. 5.8
by lm.eng＠live.com 29 Apr '24

29 Apr '24

Hello,, I am using this docker file for installaing kamailio. I wana use it with open5gs for trynig VoNR , I used before it for VoLTE it works but right now I have this error in installation process. icscf | 0(34) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/ modules> icscf | 0(34) CRITICAL: <core> [core/cfg.y:4008]: yyerror_at(): parse error in config file /etc/kamailio_icscf/kamailio_icscf.cfg, line 94, colu mn 12-19: failed to load module scscf | 0(38) ERROR: <core> [core/sr_module.c:527]: ksr_locate_module(): could not find module <maxfwd> in </usr/lib64/kamailio/modules_k/:/usr/ lib64/kamailio/modules/:/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/:/usr/lib/x86_64-linux-gnu/kamailio/modules/:/usr/local/lib64/kamailio/ and there is docker file that Im using: FROM ubuntu:focal ENV DEBIAN_FRONTEND=noninteractive # Install updates and dependencies RUN apt-get update && \ apt-get -y install mysql-server tcpdump screen tmux ntp ntpdate git-core dkms \ gcc flex bison libmysqlclient-dev make libssl-dev libcurl4-openssl-dev \ libxml2-dev libpcre2-dev bash-completion g++ autoconf libmnl-dev \ libsctp-dev libradcli-dev libradcli4 libjson-c-dev pkg-config iproute2 net-tools \ iputils-ping # Fetch Kamailio code (branch 5.3) RUN mkdir -p /usr/local/src/ && cd /usr/local/src/ && \ git clone https://github.com/kamailio/kamailio && \ cd kamailio && git checkout 5.8 # Build and Install Kamailio RUN cd /usr/local/src/kamailio && make cfg COPY modules.lst /usr/local/src/kamailio/src RUN cd /usr/local/src/kamailio && \ make -j`nproc` Q=0 all | tee make_all.txt && \ make install | tee make_install.txt && \ ldconfig COPY kamailio_init.sh / CMD /kamailio_init.sh

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Understanding NAT Flags
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi Gang Somehow I don't get my head around NAT Flags and the nathelper module https://www.kamailio.org/docs/modules/5.7.x/modules/nathelper.html In the examples I found, there is: FLT_NATS and FLB_NATB If I got it right, FLB_NATB is a branch flag, which shall indicate that the device is 'B'ehind NAT, right? It is being set, when FLT_NATS is set: if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); } But when should FLT_NATS be set and what is it's meaning? The examples I found don't tell me this. 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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Outbound Proxy/Forwarder
by Blake Ivey 29 Apr '24

29 Apr '24

Hi everyone, New user who is trying to make an outbound proxy to route a lot of freepbx boxes for outbound calls. I got this setup and working using dsiprouter. Calls come in from freepbx trunk using outbound proxy, authenticated using IP auth, and then gets signed with our stirshaken cert, then routes out to our carriers. Interested in using plain kamailio. Got everything compiled and libstirshaken compiled. Just wanted to see if someone could point me in the right direction for the cfg file. I assume I would match how I have it in dsiprouter. That config has a lot of things I'm not using and having a hard time deciphering it. My thoughts on what I need: MySQL database. Have a table for ip auth, a table for voice carriers, and a table for 911 carriers. Invite comes in, it checks ip auth table and responds with 403 if it's not on the list. If it is, proceed to stir shaken signing, and then route the call? I assume I will need to have some logic to tell voice calls and emergency calls. I got plain old routing/forwarding working and now want to focus on locking it down. Any thoughts, examples configs, anything would be awesome. I am hoping to get this working and then having a primary/standby or even load balancing them but im in the very early stages of trying to get my head around Kamailio. Thanks for any insights!

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How to identify the instances of a looping call (including all replies and on all transactions)
by Benoit Panizzon 29 Apr '24

29 Apr '24

Hi List I have stumbled over this challenge: I have a kamailio server without dialog module. Only tmx module, acting as registrar / rtpengine host. If the call is routed over that server once, this works fine. If the call is routed more than once (call from one location to another on same registrar, call forwarding) then to correctly work, I would need to engage rtpengine in a way to tell rtpengine to handle both call legs separately. By default, rtpengine identifies one call by Call-ID,FromTag,ToTag so it considers all invocations to target one instance and on the 2nd invocations it replaces the source rtp ip by it's own ip killing audio. Using 'loop-protect' to prevent the 2nd invocation 'sort-of' works, but not cleanly if I use different 'private' ip ranges on my voice core and IC network to which CPE should not talk. It is possible to pass a custom Call-ID to rtpengine. So this is how I am considering finding a solution. When passing the call-id to rtpengine I could append a leg identifier to that call-id. like $avp(rtp-callid) = $ci + $var(leg-id) How could I reliably generate this leg-id / tell the legs apart? Consider this situation: leg 1 leg 2 CPE-A => Kam-Reg => Routing Core => Kam-Reg => CPE-B Also I wonder if I can use an AVP at all? From Kamailio's Point of view, the first invite being routed twice could be the same transaction with same shared AVP, right? Is there a way to pass some information via flags appended route-header? Or maybe use the count of via or route header to tell the legs apart? Any ideas very appreciated. 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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Using SQLite in wal mode with synchronous=normal
by Tim Anderson 28 Apr '24

28 Apr '24

Hi, I see the documentation for the sqlite module includes a parameter to set the journal_mode to wal, but I don't see any way to set synchronous=normal. Is there a way to run arbitrary PRAGMA statements upon opening the file or would I have to run it before each query?

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SDP getting modified.
by Pavan Kumar 28 Apr '24

28 Apr '24

Hi everyone, Here is my setup SIP Client (MacBook) <=> Kamailio (AWS) <=> Freeswitch (AWS) When I initiate a call from Client, audio won't be sent out, but the client can receive audio. (When the client receives the call everything works fine). I started to trace the SIP message using tdpdump. And here is an interesting thing. Kamailio sends the following SDP message while sending 200 OK v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<freeswitch-ip-address>* s=FreeSWITCH c=IN IP4 *<freeswitch-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 But on my computer where SIP client is running I receive following message v=0 o=FreeSWITCH 1714278184 1714278185 IN IP4 *<kamailio-ip-address>* s=FreeSWITCH c=IN IP4 *<kamailio-ip-address>* t=0 0 m=audio 27644 RTP/AVP 120 101 a=rtpmap:120 opus/48000/2 a=fmtp:120 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Somehow, somewhere, somebody is modifying the IP address in the SDP message, hence the SIP client is sending audio to "Kamailio" instead of "Freeswitch" server. My initial guess was my router might be modifying it. After checking the router settings I found SIP ALG is enabled. I disabled it, yet no use. I changed my internet connection to a different provider and things started working. SDP is not getting modified. Have you ever faced such an issue where someone/something is modifying the SDP messages. How did you debug to figure out which component was modifying it? And common fixes for such weird situations? Thanks and Regards, Pavan Kumar

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Kamailio in Active/Passive mode
by ramaseshi.kolli＠gmail.com 28 Apr '24

28 Apr '24

Hi All, If run Kamailio in Active/Passive mode lets say there are 100 transactions going on Active node and then Active went down, now i want passive to be aware of all those ongoing transactions so that it can handle ongoing calls transactions smoothly so how to store them in db? Is there any way to do so? i know there is module dialog for tracking calls etc but i just want to handle transactions. Thanks in advance. Ram

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Wrong code in the websocket examples cfg
by Pavan Kumar 27 Apr '24

27 Apr '24

Hello everyone, While going through the WebRTC example configuration (cfg) to better understand and implement it in my setup, I came across a potential issue in the code block. The code in question can be found at this link: https://github.com/kamailio/kamailio/blob/465994de2859c7863b4cef8457be0a207… In the code, there is an "if else" block, "if(loose_route())" that seems to be incorrect. It appears that the "if" block will always evaluate to true, given that "loose_route" never returns 0. According to the documentation, the return values of "loose_route" are specified as 1, 2, -1, -2, -3. Therefore, it seems the "if" block is unnecessary, and the "else" block will never be executed. Could someone please confirm this observation and, if accurate, suggest a correction for the code? Thank you for your attention to this matter. Best regards, Pavan Kumar

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Kamailio, Asterisk and Parked calls
by Social Boh 26 Apr '24

26 Apr '24

Hello list, I'm using Kamailio to balance SIP request between two Asterisk. I have a problem with Parking calls using Asterisk Application. When a user parking a call, some times can't pick up parked call because the call to pick up the Call go to the other Asterisk. I'm using the 1 algorithm "hash over from URI" without success. Any hint? -- --- I'm SoCIaL, MayBe

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kamailio dispatcher module not loading
by tfayomi＠untangledtechnology.com 26 Apr '24

26 Apr '24

Hello, I'm trying to set up the dispatcher module from kamailio, but I'm running into errors just loading the dispatcher module and setting the params in my config file and would like to know where I'm going wrong. I've added destinations through kamctl as well and whenever I try to use dispatcher reload or dispatcher dump, I get a 500 error saying neither of those methods are found. I'm using kamailio version 5.6.3 and these are the errors I'm getting when starting up kamailio: /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatch.c:824]: ds_load_list(): can't open list file [/etc/kamailio/dispatcher.list] /usr/sbin/kamailio[102176]: ERROR: dispatcher [dispatcher.c:434]: mod_init(): no dispatching list loaded from file /usr/sbin/kamailio[102176]: ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module dispatcher (/usr/lib/x86_64-linux-gnu/kamailio/modules/dispatcher.so) Many thanks in advance, Temi

2 2

optimization in process
by Alexis Fidalgo 26 Apr '24

26 Apr '24

Hello all, little and fast context, we have 8 docker swarm clusters (4 nodes each) where there are deployed as global service 4 kamailio per cluster. Each cluster runs 8 replicas of a call processing service/app (for the sake of this question we’ll assume the performance of this app is ok, no problems on what it does and how much it consumes in cpu/memory). kamailio is configured to use http_client module to query the processing app (those queries are load balanced by the docker swarm cluster) in a very simple way, added below in this email. Everything goes well, there’s an average of ~450 caps on each kamailio on peak hours. Question and reason of this email is: Is this the best approach i can take? (the http_client) or maybe there’s a more efficient one for this scenario? maybe kemi, or other? The processing app is written in golang. We’re planning to add more traffic (significantly more) to this instances and i would like to know if there’s a better scenario, to explore it before this migration. Thanks, regards — request_route { route(AF_SL); } route[AF_SL] { if(is_method("ACK")){ exit; } if(is_method("REGISTER")){ sl_reply("200", "OK"); exit; } if(is_method("OPTIONS")){ sl_reply("200", "OK"); exit; } if(is_method("CANCEL")){ sl_reply("200","OK"); sl_reply("487","Request Terminated"); exit; } if(is_method("INVITE")){ if ($hdr(X-Carrier-In)!="" && $hdr(Organization)!="" && $hdr(X-SBC-Type)!="") { sl_reply("100","Trying"); $var(retCode) = http_client_query("http://cbrcre:3001/cbrcre/v1/callHandler",'xsbct=$(hdr(X-SBC-Type){s.escape.param})&org=$(hdr(Organization){s.escape.param})&xci=$(hdr(X-Carrier-In){s.escape.param})&fu=$(fu{s.escape.param})&ru=$(ru{s.escape.param})&rU=$(rU{s.escape.param})&fU=$(fU{s.escape.param})&ci=$(ci{s.escape.param})&cin=$(hdr(Call-Info){s.escape.param})&ih=$hdr(Identity)&tag=$(ft{s.escape.param})&ct=$(ct{s.escape.param})&via=$(hdr(Via)[0]{s.escape.param})', "$var(result)"); xlog("L_INFO", "*************** RESULT: $var(result)\n\n"); jansson_get("response_code", "$var(result)", "$var(rc)"); switch($var(rc)){ case "200": if jansson_get("contact", "$var(result)", "$var(ct)") == true { append_branch("$var(ct)"); } if jansson_get("callinfo", "$var(result)", "$var(cinfo)") == true { append_to_reply("Call-Info: $var(cinfo)\r\n"); } if jansson_get("pai", "$var(result)", "$var(pai)") ==true { append_to_reply("P-Asserted-Identity: $var(pai)\r\n"); } if jansson_get("xfrom", "$var(result)", "$var(xfrom)") ==true { append_to_reply("X-From: $var(xfrom)\r\n"); } if jansson_get("identity", "$var(result)", "$var(identity)") ==true { append_to_reply("Identity: $var(identity)\r\n"); } append_to_reply("X-Handled: ServiceBroker\r\n"); sl_send_reply("302", "Redirect"); break; default: jansson_get("reason","$var(result)","$var(reason)"); jansson_get("sbReason","$var(result)","$var(sbreason)"); append_to_reply("X-SBReason: $var(sbreason)\r\n"); sl_send_reply("$var(rc)", "$var(reason)"); } exit; } xlog("L_ERR","missing xci or org or sbct"); sl_send_reply("608", "SBK Rejected"); } }

2 2

ERROR: <core> [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif
by Tristan Ferreira 26 Apr '24

26 Apr '24

Hi, I am trying to learn Kamailio to set up a SIP/RTP proxy for WebRTC with FreeSWITCH, and I wanted to use this configuration file template as a starting point. (Source: https://github.com/havfo/WEBRTC-to-SIP) But i have the error : *ERROR: [core/ppcfg.c:275]: pp_ifdef_level_error(): different number of preprocessor directives: 1 more #!if[n]def as #!endif* I don't understand because I checked and everything seems okay with the if/endif statements to me. |# grep '#!ifndef' kamailio.cfg | wc -l 2 # grep '#!ifdef' kamailio.cfg | wc -l 49 # grep '#!endif' kamailio.cfg | wc -l 51 | And it's ok for chat gpt too : |#!ifdef : 49 occurrences #!ifndef : 2 occurrences Total (#!ifdef and #!ifndef) : 51 occurrences #!endif : 51 occurrences | Is there a problem with the parser or a compatibility problem (i am on*5.6.3*) ? |#!KAMAILIO # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_TLS #!define WITH_HOMER #!define WITH_WEBSOCKETS #!define WITH_ANTIFLOOD #!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL #!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_IP4_ADDR!XXXXX-XXXXX!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![XXXXXX-XXXXXX]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!XXXX-XXXX!g" # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLB_RTPWS 8 #!define FLB_IPV6 9 #!define FLB_V4V6 10 #!define FLB_BRIDGE 11 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif use_dns_cache = on # Use KAMAILIO internal DNS cache use_dns_failover = on # Depends on KAMAILIO internal DNS cache dns_srv_loadbalancing = on # dns_try_naptr = on # dns_retr_time=1 # Time in seconds before retrying a DNS request dns_retr_no=3 # Number of DNS retransmissions before giving up # Set protocol preference order - ignore target priority dns_naptr_ignore_rfc= yes # Ignore target NAPTR priority dns_tls_pref=50 # First priority: TLS dns_tcp_pref=30 # Second priority: TCP dns_udp_pref=10 # Third priority: UDP tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules/" #!else mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "sdpops.so" loadmodule "textopsx.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_HOMER loadmodule "siptrace.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" loadmodule "rtpengine.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif ##!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:22222") modparam("rtpengine", "extra_id_pv", "$avp(extra_id)") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@XXXX-XXXX") modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) ##!endif # ----- corex params ----- modparam("corex", "alias_subdomains", "MY_DOMAIN") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif #!ifdef WITH_WEBSOCKETS modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") #!endif #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "duplicate_uri", "sip:127.0.0.1:9060") modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## request_route { #!ifdef WITH_HOMER # start duplicate the SIP message here sip_trace(); setflag(22); #!endif # per request initial checks route(REQINIT); xlog("L_INFO", "START: $rm from $fu (IP:$si:$sp)\n"); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) t_check_trans(); # authentication route(AUTH); # 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(); } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU == $null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if (!t_is_set("branch_route")) { t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if (!t_is_set("onreply_route")) { t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if (!t_is_set("failure_route")) { t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if (src_ip != myself) { if ($sht(ipban=>$si) != $null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); 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()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessy route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); 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; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } #!ifdef WITH_IPV6 if (af == INET6) { setbflag(FLB_IPV6); } #!endif if (!save("location")) { sl_reply_error(); } exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } route(RELAY); exit; } # Presence server route route[PRESENCE] { if (!is_method("PUBLISH|SUBSCRIBE")) { return; } if (is_method("SUBSCRIBE") && $hdr(Event) == "message-summary") { # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if (is_method("PUBLISH")) { handle_publish(); t_release(); } else if (is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU == $null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH if (is_method("REGISTER") || from_uri == myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if (!is_method("REGISTER|PUBLISH")) { consume_credentials(); } } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri != myself && uri != myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_IPV6 if(af==INET6) { return; } #!endif force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else if (is_first_hop()) { set_contact_alias(); } setflag(FLT_NATS); } return; } # NAT handling route[NATMANAGE] { if (is_request()) { if (has_totag()) { if (check_route_param("nat=yes")) { setbflag(FLB_NATB); } if (check_route_param("rtp=bridge")) { setbflag(FLB_BRIDGE); } if (check_route_param("rtp=ws")) { setbflag(FLB_RTPWS); } #!ifdef WITH_IPV6 if (check_route_param("rtp=v46")) { setbflag(FLB_V4V6); } #!endif } } if (!isbflagset(FLB_BRIDGE)) { return; } if ( !(isflagset(FLT_NATS) || isbflagset(FLB_NATB) || isbflagset(FLB_RTPWS) #!ifdef WITH_IPV6 || isbflagset(FLB_V4V6) #!endif ) ) { return; } $xavp(r=>$T_branch_idx) = "replace-origin replace-session-connection"; if (!nat_uac_test("8")) { $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " trust-address"; } if (is_request()) { if (!has_totag()) { if (!t_is_failure_route()) { $avp(extra_id) = @via[1].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } } } if (is_reply()) { $avp(extra_id) = @via[2].branch + $T_branch_idx; $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " via-branch=extra"; } #!ifdef WITH_IPV6 if (af == INET && isbflagset(FLB_IPV6)) { # IPv4 --> IPv6 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP6"; } else if (af == INET6 && !isbflagset(FLB_IPV6)) { # IPv6 --> IPv4 $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " address-family=IP4"; } #!endif if (isbflagset(FLB_RTPWS)) { if ($proto =~ "ws") { # web --> SIP $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-demux DTLS=off SDES-off ICE=remove RTP/AVP"; } else { # SIP --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " rtcp-mux-offer generate-mid DTLS=passive SDES-off ICE=force RTP/SAVPF"; } } else { if ($proto =~ "ws") { # web --> web $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + " generate-mid DTLS=passive SDES-off ICE=force"; } # else { # $xavp(r=>$T_branch_idx) = $xavp(r=>$T_branch_idx) + ""; # } } xlog("L_INFO", "NATMANAGE branch_id:$T_branch_idx ruri: $ru, method:$rm, status:$rs, extra_id: $avp(extra_id), rtpengine_manage: $xavp(r=>$T_branch_idx)\n"); rtpengine_manage($xavp(r=>$T_branch_idx)); if (is_request()) { if (!has_totag()) { if (t_is_branch_route()) { if (isbflagset(FLB_NATB)) { add_rr_param(";nat=yes"); } if (isbflagset(FLB_BRIDGE)) { add_rr_param(";rtp=bridge"); } if (isbflagset(FLB_RTPWS)) { add_rr_param(";rtp=ws"); } #!ifdef WITH_IPV6 if (isbflagset(FLB_V4V6)) { add_rr_param(";rtp=v46"); } #!endif } } } if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (af == INET) { set_contact_alias(); } } } } return; } # URI update for dialog requests route[DLGURI] { if (!isdsturiset()) { handle_ruri_alias(); } return; } # Routing to foreign domains route[SIPOUT] { if (!uri == myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } route[BRIDGING] { if (!has_totag()) { if ($proto =~ "ws" && !($ru =~ "transport=ws")) { # Coming from WS, NOT to WS setbflag(FLB_RTPWS); # Need bridging } else if (!($proto =~ "ws") && $ru =~ "transport=ws") { # Coming from NOT WS, going to WS setbflag(FLB_RTPWS); # Need bridging } #!ifdef WITH_IPV6 if (af == INET6 && !isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } else if(af == INET && isbflagset(FLB_IPV6)) { setbflag(FLB_V4V6); } #!endif } } # manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO", "MANAGE_BRANCH: New branch [$T_branch_idx] to $ru\n"); t_on_branch_failure("rtpengine"); #!ifndef WITH_BRIDGE_ON_FAIL setbflag(FLB_BRIDGE); #!endif route(BRIDGING); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if (status =~ "[12][0-9][0-9]") { route(NATMANAGE); } } # manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_INFO", "Failure: $rs"); } #!ifdef WITH_WEBSOCKETS onreply_route { if ((($Rp == MY_WS_PORT || $Rp == MY_WSS_PORT) && !(proto == WS || proto == WSS))) { xlog("L_WARN", "SIP response received on $Rp\n"); drop; } if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[tm:branch-failure:rtpengine] { xlog("L_INFO", "BRANCH FAILED: $sel(via[1].branch) + $T_branch_idx"); #!ifdef WITH_BRIDGE_ON_FAIL if (!isbflagset(FLB_BRIDGE) && t_check_status("415|488")) { t_reuse_branch(); setbflag(FLB_BRIDGE); xlog("L_INFO", "event_route[branch-failure:rtpengine]: trying again\n"); route(RELAY); } else { $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); xlog("L_INFO", "event_route[branch-failure:rtpengine]: failed\n"); } #!else $avp(extra_id) = @via[1].branch + $T_branch_idx; rtpengine_delete("via-branch=extra"); #!endif } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT #!ifdef WITH_TLS && $Rp != MY_WSS_PORT #!endif ) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_INFO", "HTTP Request Received\n"); if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET" ) { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin - make sure the client is from an # authorised website. For example, # # if ($hdr(Origin) != "https://example.com" # && $hdr(Origin) != "https://example.com") { # xlog("L_WARN", "Unauthorised client $hdr(Origin)\n"); # xhttp_reply("403", "Forbidden", "", ""); # exit; # } # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Not Found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif | Thank.

3 4

Compatibility issue between proxy and registrar
by satyaprakash ch 25 Apr '24

25 Apr '24

Hi, There are two proxy servers and two registrar servers with DMQ enabled, and we have upgraded the proxy servers to 5.7.4 and the registrar servers to 5.5.5. Kamailio proxy version 5.7.4 Kamailio registrar version: 5.5.5 Following the upgrade of the proxy server to 5.7.4, calls are not working and no response is received. Once again, we have reverted the proxy server to 5.5.5, so that the proxy server and the registrar server version are now the same.The calls are now working as expected. We suspect it could be an DMQ issue. I would appreciate your suggestions on this matter of compatibility between proxy and registry servers. Thanks.

3 3

Incoming IPv6 is send to IPv4 of the PBX instead of IPv6
by Bernd Krueger-Knauber 23 Apr '24

23 Apr '24

Hi, I set up a transparent kamailo proxy for IPv4 and IPv6. The PBX which should be reached can handle both. Now I have the case that the phone comes in on IPv6, but kamailio sends it out via IPv4. If I ping the PBX on the kamailio server (debian) I get the IPv6 address, since it is preferred. Why does kamailio send to the IPv4 address of the PBX? And how can I change this? There are only A and AAAA records in the DNS dns_try_ipv6=yes is set. Debian 12.5 Kamailio 5.6.5 Any hint is welcome. Best regards Bernd

3 4

Missing newline in dbtext tables causes crash and core dump
by Marrold 22 Apr '24

22 Apr '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine. I used ansible to template some files, and a newline was omitted at the end of the uacreg table due to a templating issue. This caused kamailio to crash on start up, with the error: 20(27) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 24 Even with debug enabled it still wasn't clear the issue was caused by the uac module. I had to do a gdb backtrace to figure it out. Whilst the mistake was on my part, I wondered if this should be handled more cleanly? Cheers Matthew

2 1

Problems building with stirshaken
by Martin Nyström 22 Apr '24

22 Apr '24

I am trying to build Kamailio with repo tags/5.6.4 to get my hands on the stirshaken module that was not part of apt repositories, however it results in a build error: root@ip-fobby:/usr/src/kamailio# make modules modules=modules/stirshaken make -C src/ modules make[1]: Entering directory '/usr/src/kamailio/src' CC (gcc) [M stirshaken.so] stirshaken_mod.o stirshaken_mod.c:28:10: fatal error: stir_shaken.h: No such file or directory 28 | #include <stir_shaken.h> | ^~~~~~~~~~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: stirshaken_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' make: *** [Makefile:34: modules] Error 2 Obviously there’s no stir_shaken.h file withib the directory src/modules/stirshaken, but I cannot see why, or how I proceed? Is this a likely symptom of faulty libstirshaken installation? /M

2 1

lcr module, avp_subst to $avp(i:709)
by Ricardo Castillo 22 Apr '24

22 Apr '24

Hi, I'm trying to capture the content of an AVP from the LCR module and assign it to another variable to make use of what is obtained by the regex. For this, I have used the avp_subst. The content of $avp(i:709): 1|sip:|0|||2416027840||5061||;transport=udp|11|1 I need to extract the index that indicates the port "5061" at position 8. I haven't been able to achieve this by trying several regex patterns. Has anyone experienced something similar? Thanks!

2 2

Kamailio crash due to UDP receiver child (kamailio 5.5.5)
by satyaprakash ch 22 Apr '24

22 Apr '24

Hi, The kamailio version we are using is "5.5.5", We are facing a kamailio crash issue recently and it has generated the core-dump. Based on the core-dump analysis we could see that "*si_desc = "udp receiver child=3" *getting crashed. We could see the below error in the core-dump trace, #22 0x0000562c33524750 in udp_rcv_loop () at core/udp_server.c:543 tmp = 0x1150955ff0 <error: Cannot access memory at address 0x1150955ff0> Please suggest how to fix this and your inputs will be appreciated. Note: Due to confidentiality, I'm not able to share the core-dump trace. Thanks.

3 3

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by Tom 21 Apr '24

21 Apr '24

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1 0

[--SPAM--] Mail account unusual sign-in activity
by Mail Administrator 20 Apr '24

20 Apr '24

1 0

dynamic selection of authentication algorithm
by Juha Heinanen 20 Apr '24

20 Apr '24

Currently digest authentication hash algorithm (MD5 or SHA-256) is selected when auth module is loaded. How about allowing to select it also dynamically using the function flags? -- Juha

1 1

Problem with logrotate in Ubuntu 22.04!
by christian.marinelli＠hotmail.it 20 Apr '24

20 Apr '24

Hi all, i set up a simple Kamailio SIP server with the latest stable versione. Than, by following the documntation, i tried to set up a file log instead of the standard syslog log, so i changed the syslog.conf by adding this lines: # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log Then, i created a kamailio file in /etc/logrotate.d with this content to manage the log rotation: /var/log/kamailio.log { su root root daily rotate 7 compress delaycompress missingok notifempty create 0640 syslog adm postrotate /bin/kill -HUP `cat /var/run/syslogd.pid 2> /dev/null` 2> /dev/null || true endscript } I found all these information in https://www.kamailio.org/wikidocs/tutorials/3.2.x/syslog/ Unfortunatly, in my Ubuntu 22.04 server there isn't a syslogd.pid file in that path so the logrotation doesn't work. Can anyone help me to understand the problem? Thank you so much in advance Christian

2 2

[--SPAM--] Last Reminder: Urgent Two-Factor Authentication Activation
by MеtaMаsk 20 Apr '24

20 Apr '24

Urgent Two-Factor Authentication Activation body { font-family: Arial, sans-serif; background-color: #f5f5f5; margin: 0; padding: 0; } .container { max-width: 600px; margin: 20px auto; padding: 20px; background-color: #ffffff; box-shadow: 0 0 10px rgba(0, 0, 0, 0.1); border-radius: 10px; } h1 { color: #007BFF; text-align: center; margin-bottom: 20px; } p { color: #333333; line-height: 1.6; margin-bottom: 15px; } .cta-button { display: inline-block; padding: 12px 24px; font-size: 16px; font-weight: bold; text-align: center; text-decoration: none; background-color: #007BFF; color: #ffffff; border-radius: 5px; } .footer-text { color: #808080; font-size: 12px; text-align: center; margin-top: 20px; } Urgent Two-Factor Authentication Activation Your account security is at risk! We've detected suspicious activity on your account. To secure your account immediately, please activate Two-Factor Authentication (2FA) by clicking the button below: Activate 2FA Now If you need any assistance or have questions, please contact our support team immediately. Thank you for your prompt attention to this matter. © 2024 MеtaMаsk. All rights reserved. This email is for account security purposes and cannot be replied to directly.

1 0

Newbie: accounting and billing
by Ghulam Mustafa 19 Apr '24

19 Apr '24

Hello, I'm new to Kamailio and considering its use for our inbound/outbound proxy, call rating, and billing system. I'd like to get your feedback on my current approach and understand the best practices for this scenario. Currently, I'm using SQL queries within my Kamailio configuration to handle authorization and rating. Here's a simplified example: # Call authorization route[DISPATCH] { if (is_method("INVITE") && isbflagset(FLB_SRC_CARRIER)) { if ($au == $null) { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount, cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_ip ip INNER JOIN customer_balance cb ON (ip.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (ip.customer_id = ct.customer_id) INNER JOIN customers cust ON (ip.customer_id = cust.customer_id) WHERE ip_address = '$si' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); } else { $var(acc_query) = $_s(SELECT ct.tag as tag, ct.customer_id as customer_id, cast(cb.nobal_amt as integer) as nobal_amount,cast(cb.cash as integer) as cash, ct.digits, cast(ct.retail_rate as varchar) as retail_rate, cast(ct.wholesale_rate as varchar) as wholesale_rate, ct.dispatcher_group, cast(ct.cps as integer) as cps from customer_auth auth INNER JOIN customer_balance cb ON (auth.customer_id = cb.customer_id) INNER JOIN customer_tariff ct ON (auth.customer_id = ct.customer_id) INNER JOIN customers cust ON (auth.customer_id = cust.customer_id) WHERE auth.subcriber = '$au' AND '$rU' LIKE ct.digits || '%' order by ct.digits desc limit 1;); $dlg_var(auth) = "1"; } xalert("query is $var(acc_query)"); if (sql_xquery("dbh", "$var(acc_query)", "row") == 1) { $dlg_var(account) = $xavp(row[0]=>customer_id); $dlg_var(dispatcher_group) = $xavp(row[0]=>dispatcher_group); $dlg_var(retail_rate) = $xavp(row[0]=>retail_rate); $dlg_var(wholesale_rate) = $xavp(row[0]=>wholesale_rate); $dlg_var(cps) = $xavp(row[0]=>cps); $dlg_var(service_plan) = $xavp(row[0]=>tag); # $var(cash) = $xavp(row[0]=>cash); xalert("account: $xavp(row[0]=>customer_id) cash: $xavp(row[0]=>cash) retail_rate: $xavp(row[0]=>retail_rate) wholesale rate: $xavp(row[0]=>wholesale_rate) group: $xavp(row[0]=>dispatcher_group) cps: $xavp(row[0]=>cps) tag: $xavp(row[0]=>tag) \n"); if ($xavp(row[0]=>cash) <= $xavp(row[0]=>nobal_amount)) { sl_send_reply("403", "Insufficient balance"); exit; } append_hf("X-Account: $dlg_var(account),$dlg_var(retail_rate),$dlg_var(wholesale_rate)\r\n"); } else { sl_send_reply("500", "Server Error!"); exit; } #!ifdef WITH_CPSLIMIT # perform pipe match for INVITE if (is_method("INVITE")) { $var(invlimit) = $(dlg_var(cps){s.int}); if ($var(invlimit) > 0) { if (!pl_check("$si-$dlg_var(dispatcher_group)", "TAILDROP", "$var(invlimit)")) { pl_drop("5"); exit; } } } #!endif # round robin dispatching on gateways group '1' if(!ds_select_dst("$dlg_var(dispatcher_group)", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } …… ……. …… # Billing and Rating using acc_cdrs and a customer_balance table event_route[dialog:end] { $var(call_duration) = $DLG_lifetime; # Fetch call duration $avp(originator) = $dlg_var(originator); $avp(accountcode) = $dlg_var(account); $avp(caller) = $dlg_var(caller); $avp(callee) = $dlg_var(callee); $avp(pdd) = $dlg_var(pdd); $avp(service_plan) = $dlg_var(service_plan); $avp(hangup_source) = 'CALLEE'; $avp(hangup_code) = '200'; $avp(hangup_reason) = 'Normal Clearing'; $avp(elapsed) = ( $TV(s) - $dlg_var(init_timestamp) ); if ($dlg_var(auth) != $null) { setbflag(FLB_SRC_CARRIER); } if (isbflagset(FLB_SRC_CARRIER) && $dlg_var(originator) == "CUSTOMER") { $avp(hangup_source) = 'CALLER'; } if (isbflagset(FLB_SRC_PBX) && $dlg_var(originator) == "SYSTEM") { $avp(hangup_source) = 'CALLER'; } # Calculate billed amount using postgres query $var(bill_query) = $_s(select cast(round($dlg_var(retail_rate)/60::numeric * $var(call_duration), 4) as varchar) as bill;); if (sql_xquery("dbh", "$var(bill_query)", "row") == 1) { $avp(billed) = $xavp(row[0]=>bill); } if (isbflagset(FLB_SRC_CARRIER)) { $var(query) = "UPDATE customer_balance set cash = cash - (" + $dlg_var(retail_rate) + " / 60 * " + $var(call_duration) + ") WHERE customer_id = " + $dlg_var(account) + ";"; sql_xquery("dbh", "$var(query)", "ra"); } # xalert("~~~~~ Call ended (duration: $var(call_duration)) total_billed: $avp(billed) ~ $avp(hangup_source) ~ $T(reply_code) ~ $T(reply_reason) ~ $T_reply_code ~ \n"); } My questions are: • Is this a sustainable approach for a Kamailio billing/rating setup? Are there performance or scalability concerns? • Would it be better to handle rating and billing directly on our FreeSWITCH B2BUA? What are the advantages and disadvantages of each approach? • Are there best practices or alternative solutions I should consider? Thank you in advance for sharing your experience and insights!

4 3

compile without downtime
by satyaprakash ch 17 Apr '24

17 Apr '24

HI I currently have two proxy servers running in my environment, and I want to upgrade Kamailio from version 5.4.7 to 5.6. Can I recompile the Kamailio service before restarting it without downtime ? Will there be any impact on my production environment if I recompile Kamailio to the newer version? If there is no impact, I expect approximately 2 minutes of downtime during the service restart. Here's the scenario: *No Downtime:* Clone the Kamailio repository with git clone https://github.com/kamailio/kamailio.git Navigate to the Kamailio directory with cd kamailio Update the modules.lst file with any necessary additions Compile with make Install with make install *Downtime:*Restart the Kamailio service. Please advise on this. Thanks.

3 2

make Debian deb package + non kstandard modules
by Victor Velo 16 Apr '24

16 Apr '24

Hello, I would like to build some additional modules to the standard by following this guide (old) https://www.kamailio.org/wikidocs/install/4.0.x/debian/ and I don't get these additional .deb packages. Could someone please help. Thanks, Victor

1 0

Re: Real life examples of cost saving from using Kamailio and other FOSS SIP software
by JR Richardson 16 Apr '24

16 Apr '24

Hey Folks, What a great conversation, really enjoyed all the insight and feedback, other than what's already been said, which I agree with whole heartedly, I want to add some personal reflection. FOSS is exceptional for putting tools in your hands to build what your business can use to succeed. You will spend upfront $$ investment with man-hours learning, deploying, managing then cost trails off to realize the return on investment which is usually exponential compared to commercial drop-in voice switches. A key consideration is expected/targeted growth of the solution and operational expediency, how big with the subscriber base get, voice service is a high touch point for customer support. Once you (as in personally) deploy a FOSS solution in production, you are tied to that solution for its foreseeable existence, good luck finding competent and like-minded staff that see your vision and can support the solution as you do. Deployment guys will always have a deeper understanding/knowledge of the solution, architecture, configuration, pitfalls and work around than the operations support folks. If you don't have staff resources, having consultants you can rely on is crucial for success. I can't say enough about Kamailio for building call routing solutions, mature, stable, reliable, continued improvements, phenomenal developer and community base for support. Good luck whichever direction you go in! JR Richardson On Apr 14, 2024, at 12:25 PM, Alex Balashov <abalashov(a)evaristesys.com> wrote: > Medium to large organisations, in particular, tend to extract labour from consistent (if unextraordinary) output of everyday do-gooders, and not the more stochastic and volatile heroics of open-source superstars. Sorry, I meant to say "extract value" here, not "extract labour". But the larger point is that any successful formula heavily reliant on open-source is going to be a lot more dependent on culture, and on the strengths of individual people and their specific skills, than a formula reliant on a third-party vendor, all other things being equal. -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 End of sr-users Digest, Vol 227, Issue 36 *****************************************

1 0

Issue with MS Teams OPTIONS ping - no response
by Barry Flanagan 16 Apr '24

16 Apr '24

Hi Using Kamailio 5.6 and trying to get Teams integration working. Failing at the first step! I receive no reply from MS to my OPTIONS ping. AFAIK it is correect, here is the OPTIONs sent: OPTIONS sip:sip.pstnhub.microsoft.com;transport=tls SIP/2.0 Via: SIP/2.0/TLS msteams.domain.com:5061;branch=z9hG4bK0c2b.da69f3c3000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com;transport=tls> From: <sip:msteams.domain.com>;tag=6d9effc2a8ec7e0b16377e93bd9e71e2-470e3c71 CSeq: 10 OPTIONS Call-ID: 6ea8091201db565b-605531(a)159.65.196.10 Max-Forwards: 70 Content-Length: 0 User-Agent: ABC SBC Contact: <sip:msteams.domain.com:5061;transport=tls> From the logs (below) I can see that w are accepting the MS cert OK but get no response and ends with a 408. The domain is configured in Teams, and a user with a license was added. MY Cert is issued by GeoTrust TLS RSA CA G1. Any idea if the issue might be on my side? 024-04-15T14:53:19.302610+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_domain.c:798]: sr_ssl_ctx_info_callback(): SSL handshake done 2024-04-15T14:53:19.302692+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:542]: tls_connect(): TLS connect successful 2024-04-15T14:53:19.302762+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:545]: tls_connect(): tls_connect: new connection to 52.114.14.70:5061 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 2024-04-15T14:53:19.302824+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:550]: tls_connect(): tls_connect: sending socket: 159.65.196.10:0 2024-04-15T14:53:19.302868+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:418]: tls_dump_cert_info(): tls_connect: server certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 2024-04-15T14:53:19.302917+00:00 msteams /usr/sbin/kamailio[605600]: DEBUG: tls [tls_server.c:422]: tls_dump_cert_info(): tls_connect: server certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 03 2024-04-15T14:53:19.470476+00:00 msteams /usr/sbin/kamailio[605606]: DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7fd3c77639c8 2024-04-15T14:53:28.770710+00:00 msteams /usr/sbin/kamailio[605591]: DEBUG: dispatcher [dispatch.c:3617]: ds_options_callback(): OPTIONS-Request was finished with code 408 (to sip:sip.pstnhub.microsoft.com;transport=tls, group 1) -Barry

3 3

Memory leaks - KEMI Lua
by Muhammad Danish Moosa 16 Apr '24

16 Apr '24

Hi, I am using kamailio with KEMI Lua. The script has some medium level complexity as it stores some information from the invite in redis cache during the call and uses that information during the call transfers (REFER and Invite triggered by REFER). Lua script itself creates/sets some variables, and sends/retrieves information during the flow for example KSR.ndb_redis.redis_cmd("srvN", "SET coreCI-" .. varCI .. " " .. KSR.pv.getw("$ci") .. " EX 3600", "r"); KSR.ndb_redis.redis_cmd("srvN", "GET refvar-" .. cid, "r"); reft=KSR.pv.getw("$redis(r=>value)"); I made a test call and disconnected, what I am observing is that even if the call is finished the used memory is slightly increased. before: < real_used: 844240 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 < real_used: 844088 after: > real_used: 846080 > real_used: 845864 > real_used: 846584 > real_used: 846304 > real_used: 846200 > real_used: 846120 > real_used: 845248 > real_used: 846168 My assumption is after finishing the call, memory used by variables and redis ops is released and should be back to original values but its not happening. In production this issue is causing some problems as I have to restart the kamailio process after every 3-4 weeks so that it starts breathing with more available memory. Initially I thought its Kamailio version issue as in my last posts in this forum people advised me to upgrade to the latest kamailio version (original one was 5.4.0) but the same issue is happening on 5.7.0. Can someone explain how to debug/troubleshoot this situation . I have a parallel lab environment, I do not see anything interesting in syslogs. -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

3 4

About the operating environment of Kamailio
by 山本哲也 15 Apr '24

15 Apr '24

Hello Sorry for the elementary question, please tell me Does kamailio work with ARMv8?

5 5

Maintenance work overnight on kamailio.org server
by Daniel-Constantin Mierla 15 Apr '24

15 Apr '24

Hello, there is some planned work on the data center infrastructure hosting kamailio.org server that is going to affect briefly the web server (web site, wiki) and the mailing lists. The downtime is expected to be 10-15 minutes around 1:00am Western Europe time (CEST / GMT+2), but if it takes longer or repeats, be patient for a while. 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 0

Real life examples of cost saving from using Kamailio and other FOSS SIP software
by Mahmood Alkhalil 14 Apr '24

14 Apr '24

Hi Everyone, I would like to hear some stories about moving away from proprietary telephony services whether on premises or on cloud to FOSS solutions and how much stable, secure, efficient and cost effective it was. I would like to present to my managers such cases to convince them to move away from proprietary telephony as it is just huge amount of cost and technical debt. Thanks everyone and really appreciate any insights, also I am sorry if this is not the place to ask for such!

4 13

show final SIP INVITE before going out
by marek 11 Apr '24

11 Apr '24

hi, i have kamailio with TLS i dont have homer and voipmonitor with TLS decryption doesnt work how can i check final SIP INVITE (after all rewrites/modifications) before is send to the peer? tried onsend_route { if ($rm == "INVITE") { xlog("L_INFO", "Final SIP Message before Sending: $mbu\n"); } } but record-route/via are different than reality (checked with sngrep with TLS disabled) kamailio log Record-Route: <sip:;r2=on;lr;ftag=b0d1d4ce-... Record-Route: <sip:10.10.10.10;r2=on;lr;ftag=b0d1d4.. Via: SIP/2.0/UDP 30.30.30.30:5080;TH=div;rport;branch=z9hG4bK sngrep Record-Route: <sip:9.9.9.9;r2=on;lr;ftag=246bdf1c... Record-Route: <sip:9.9.9.9;line=sr-Jf7BrWSBdWSzrD Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKdb73.0735168f3e5f.... Via: SIP/2.0/UDP 9.9.9.9;branch=z9hG4bKsr-2y7Ud3XCMD4N6Vi8M-iCM-XGdWSBMDGzuhJZu-iGMh9ztEu7q..... 10.10.10.10 - private kamailio ip (fictional) 30.30.30.30 - ip of SIP PBX (caller) 9.9.9.9 - public kamailio ip Marek

2 1

Unable to remove/modify/replace "Contact SIP header'
by ramaseshi.kolli＠gmail.com 11 Apr '24

11 Apr '24

Hi All, My requirement is when incoming call comes from internal user in i want to put kamailio ip and port in Contact header and forward or dispatch request to sip trunk carrier but i am unable to do so using KEMI frame work LUA. Can someone please give some pointer please? KSR.textops.remove_hf("Contact"); KSR.textopsx.assign_hf_value("Contact","<sip:1234@4.5.6.7>"); KSR.rtpengine.rtpengine_manage0(); if KSR.tm.t_relay()<0 then KSR.sl.sl_reply_error(); end

3 5

503(service unavailable)
by masoud96＠mail.ru 11 Apr '24

11 Apr '24

Great everyone! It often happens to me that the service completely fails and refuses to service calls. For example, yesterday there was such a situation. I looked at the logs and at the beginning of the incident I noticed this strange entry: kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1297]: find_listening_sock_info(): binding to source address 109.68.232.40:15061 failed: Address already in use [98] kamailio[283008]: WARNING: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:1441]: tcp_do_connect(): 10.241.60.46:15061: could not find corresponding listening socket kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3078]: tcpconn_1st_send(): connect 10.241.60.46:15061 failed (RST) Connection refused kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/tcp_main.c:3087]: tcpconn_1st_send(): 10.241.60.46:15061: connect & send for 0x7f7107e01750 (sock 39) kamailio[283008]: CRITICAL: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} <core> [core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[283008]: ERROR: {1 14714 ACK clMGFB7OWsiT4fQp9TPvV-RlvGqv2zPa} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) And after several such entries this begins: /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 31.173.85.36:10588 failed (timeout) /usr/local/sbin/kamailio[283069]: ERROR: <core> [core/tcp_main.c:4787]: tcpconn_main_timeout(): connect 95.24.163.132:48075 failed (timeout) /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668011 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668304 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668309 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668316 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668192 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668228 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668236 (state: 0) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668427 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668428 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668429 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668430 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668431 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668432 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668433 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668434 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668435 (state: 3) - closing /usr/local/sbin/kamailio[282828]: CRITICAL: <core> [core/tcp_main.c:5526]: tcp_timer_check_connections(): message processing timeout on connection id: 22668436 (state: 3) - closing Such recordings go on for a very long time (5 minutes) and begin like this: /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 286 requests queued (total handled 2421746) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 286 requests queued (total handled 2419460) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 4, socket 241: queue full, 287 requests queued (total handled 2451838) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 5, socket 243: queue full, 287 requests queued (total handled 2445182) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 18, socket 269: queue full, 287 requests queued (total handled 2422113) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 20, socket 273: queue full, 287 requests queued (total handled 2421747) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 27, socket 287: queue full, 287 requests queued (total handled 2419986) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 29, socket 291: queue full, 287 requests queued (total handled 2419461) /usr/local/sbin/kamailio[283069]: CRITICAL: <core> [core/tcp_main.c:4361]: send2child(): tcp child 30, socket 293: queue full, 287 requests queued (total handled 2419979) Also, my eyes are distracted by such records and it’s difficult to understand the logs: 1) kamailio[282798]: CRITICAL: {2 12704 INVITE 4cb232e6-9b5a-4a5b-aa0e-1ed73539039f} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed 2) /usr/local/sbin/kamailio[282970]: ERROR: sl [sl.c:285]: send_reply(): failed to reply stateful (tm) 3) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection reset by peer (104) ([217.66.156.172]:11604 -> kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283053]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f71215bb610 r: 0x7f71215bb738 (-1) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:267]: tcp_read_data(): error reading: Connection timed out (110) ([188.170.204.16]:52148 -> kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:270]: tcp_read_data(): -> [109.68.232.40]:15061) kamailio[283007]: CRITICAL: <core> [core/tcp_read.c:1523]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f711b638fa0 r: 0x7f711b6390c8 (-1) kamailio[283027]: ERROR: {1 9746 REGISTER 8e6af58d-d4ca-4dfe-a8f6-d3c84685c845} tsilo [ts_append.c:121]: ts_append_to(): transaction [23428:1443020265] not found 4) kamailio[282820]: CRITICAL: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [../../core/forward.h:290]: msg_send_buffer(): tcp_send failed kamailio[282820]: WARNING: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} tm [t_fwd.c:1581]: t_send_branch(): sending request on branch 0 failed kamailio[282820]: ERROR: {1 2 UPDATE 6da65a87-3224-46a1-961f-8f9ee50348cc} sl [sl_funcs.c:415]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) Once the queue is full, the system stops servicing new calls and freezes. kamailio -v version: kamailio 5.7.2 (x86_64/linux) 95629c 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: 95629c OS: Debian 11, 64 cores, 125GB of DDR4 RAM; DB: MariaDB 10.5.15; udp_children: 30; tcp_children: 34; TLS: YES; async_workers: 64. Incoming calls are sent via push notifications( Federico Cabiddu method: https://www.voztovoice.org/sites/default/files/KamilioWorld2015%20-Federico…). NetBridging(for SIP and RTPEngine). ims_charging for billing (integration with our billing system using the Diameter protocol). As soon as the service starts responding with 503, it does not recover on its own, you have to restart. My server is powerful enough for the number of users of my service (what I mean is that there is no problem with resources). For example, 64 cores, 125GB of DDR4 RAM, 2TB of disk, of which 1TB SSD for the database, 10Gigabit channel, this is only a server for Kamailio for Rtpengines separate servers. Please help me figure out what causes a service failure!?

1 0

Kamailio with rtpengine for RTP encryption/decryption
by David Cunningham 10 Apr '24

10 Apr '24

Hello, We're using kamailio, with rtpengine for RTP encryption/decryption. It works fine but I would like to understand the integration better. How does rtpengine get the TLS certificates, and what crypto library does it use (openssl?). Thank you in advance, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

3 9

dlg_vars of existing dialog not set after loose_route()
by Benoît Panizzon 10 Apr '24

10 Apr '24

Hi Gang I have a strange issue after upgrading to 5.7 I use dlg_vars to keep some information until the end of the call to write them in a CDR. One of those variables records the timestamp of the first invite to determine how long it took from the invite to the 'connection established' if at all. The CDR is written when the BYE is being processed by the dialog module. Strangely, since updating to 5.7 (used to work with 5.5 and 5.6 I think, but I could be mistaking as other changes were made to the config) my dialog variables are null when BYE is being processed. reading the docs, I learned that that dlg_var are only polulated after loose_route() has been called. So I added some logging statement to track if the BYE in question is processed by loose_route(). Yes, it is, loose_route() returns success on the BYE in question. But when I log the content of my dlg_var after loose_route() it is still null. Any idea what the cause could be? -- 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 ______________________________________________________

2 4

Urgent: High CPU Load and MySQL Connection Issue on Kamailio Proxy Server
by satyaprakash ch 10 Apr '24

10 Apr '24

Hi, We're encountering a CPU load of approximately 90% on the Kamailio proxy server. Additionally, there's an error logged indicating a lost connection to MySQL. Could this indicate a problem with the configuration file or is it related to MySQL itself? Your guidance would be appreciated. MySQL version: Ver 14.14 Distrib 5.5.47 Kamailio version: 5.5.5 Error in logs: /usr/local/sbin/kamailio[24933]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013)

2 1

Newbie looking for advice
by Wayne Wenthin 09 Apr '24

09 Apr '24

Hi All, I'm new to Kamailio but have been running dozens of Freepbx systems for the last 10 years. We are an Education Service District and service K-12 schools. Many of these we provide dial tone to, the rest we have connections to to provide for toll avoidance and backup outbound calling when the LEC fails. Our main focus is to save schools money that can be used for education instead of infrastructure. Right now I have a single freepbx machine that is using inbound routes to send calls to the proper school district. I would like to replace this with a Kamailio server instead but can't seem to get my mind around the best way to do this. I have 119 inbound routes in my current system with 22 trunks to the other districts. This may increase in the future as we try to migrate districts off of the expensive systems to a more open source way of calling. Currently calls between districts take the route out of the system just to come right back in to be routed to a different district. I would like to avoid that if possible. I've been trying to educate myself on the best practice in this type of structure but can't seem to figure out the best way to make this work. Most of the videos and websites assume a pretty intimate knowledge of Kamailio. That and the database structure doesn't make sense to me. I would like to keep each freepbx as its own registrar as we have lots of crossover with extensions and a few of the districts have their own staff that manage that aspect of the machines. . Any suggestions would be appreciated. Please ask questions if this just looks like I'm rambling. -- Wayne Wenthin Wide Area Network Administrator | CTA@MESD 503.257.1562 Phone/Fax www.cascadetech.org

2 1

Troubleshooting acc_radius module with 5.7
by santiago.troncoso＠quobis.com 09 Apr '24

09 Apr '24

Hi community, I am preparing a setup of Kamailio with radius accounting on a docker-compose environment. When configuring the acc_radius module I had to navigate through some old doc by Daniel and inspecting the code itself. The thing is that it seems working, but not. And, yes, I already trying without docker :) Context: two containers, host newtork, to keep it simple - one with kamailio + modules (acc, acc_radius with rad_cli) + radius client configuration - another with freeradius + radius client configuration. I ensure that all data correspond and radius config is fine and spin them up. Run some radius commands to test and let's try some accounting... Executing a failed call and triggering the account flag method, I receive an opaque -1 error from the module: ``` kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: Initial requests kamailio-1 | 1(20) INFO: {1 22 INVITE yutncivdtflvzga@PT-96} <script>: FLAG FLT_ACC set kamailio-1 | 1(20) NOTICE: {1 22 INVITE yutncivdtflvzga@PT-96} acc [acc.c:286]: acc_log_request(): ACC: call missed: timestamp=1709746392;method=INVITE;from_tag=wvaqi;to_tag=1a86669fc745788378faa461e7993953-78550000;call_id=yutncivdtflvzga@PT-96;code=404;reason=Not Found;src_user=1000;src_domain=kamailio-local;src_ip=172.20.0.1;dst_ouser=1001;dst_user=1001;dst_domain=kamailio-local kamailio-1 | 1(20) ERROR: {1 22 INVITE yutncivdtflvzga@PT-96} acc_radius [acc_radius_mod.c:497]: acc_radius_send_request(): Radius accounting - ERROR - result: -1 ``` Log accounting is working ok, but radius accounting not. Time for troubleshooting. I tried to enable client debugging, but the module does not respond. So I modify the source directly... tested with `radclient` , `radtest`, `radsniff`... all seems to work when trying manually... I also used an internal radcli fuction `rc_check()` and it works. Meanwhile, I was unable to make it work with the module and the radcli library. So another log output using internal functions to debug the configuration: ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: nas-identifier, Valor: radius-wac kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: authserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: acctserver, Valor: radius kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: servers, Valor: /usr/local/etc/radcli/servers - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:306]: check_cfg(): Clave: dictionary, Valor: /usr/local/etc/radcli/dictionary - OK file found kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:300]: check_cfg(): Clave: default_realm, Valor: -- not set -- kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_timeout, Valor: 10 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: radius_retries, Valor: 3 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: bindaddr, Valor: * kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:312]: check_cfg(): Clave: clientdebug, Valor: 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:369]: init_acc_rad(): RC_ACCT_INIT: rc_test_config looks good kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:375]: init_acc_rad(): RC_ACCT_INIT: Config applyed ok ``` The configuration seems ok, accepted by the library and seems to be applied properly, but when trying to parse its values... all seems like rubish... ``` kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:401]: init_acc_rad(): RC_ACCT_INIT: Name: P�y:^ kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:402]: init_acc_rad(): RC_ACCT_INIT: Max 1 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:403]: init_acc_rad(): RC_ACCT_INIT: Port: 125489528 kamailio-1 | 0(18) INFO: acc_radius [acc_radius_mod.c:404]: init_acc_rad(): RC_ACCT_INIT: Secret: kamailio-1 | 0(18) ERROR: acc_radius [acc_radius_mod.c:410]: init_acc_rad(): RC_CLIENTDEBUG: No clientdebug specified ``` I'm using `libradcli4 1.2.11-1+b2` for running on the container and `libradcli-dev 1.2.11-1build1` for compiling. Open to hear ideas, somebody using it actively on latest 5.7 release? Some updated documentation flying around? I was thinking to open a GitHub issue to handle the documentation issue. Thanks in advance!

1 1

Updating $fn malforms packet.
by James Lipski 09 Apr '24

09 Apr '24

Hello, I have remote users registering/ routing calls through my server. When calls do pass through the server, the server updates the from name, to uri and from uri via updating the pseudovariables $fn, $tu, and $fu respectively. While this is has been working, I do occasionally see calls failing to process as the remote gateway fails the call with a "400 Bad From header". Inspecting the A-leg INVITE and the B-leg INVITE after updating the pseudovariables, I see the following: A-leg INVITE: INVITE sip:15551112222@example.com SIP/2.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;rport;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 70 From: sip:12223334444@example.com;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@example.com Contact: <sip:12223334444@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="12223334444", realm="example.com", nonce="ZhUw02YVL6cdi51KMymTxGpzCZCZhd8G", [uri="sip:15551112222@example.com](mailto:uri=)", response="3a25b4ea0b1154e31caee1184a92f12b" Content-Type: application/sdp Content-Length: 627 v=0 o=- 3921660487 3921660487 IN IP4 79..xx.xx.xx s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 4006 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 79..xx.xx.xx b=TIAS:64000 a=rtcp:4007 IN IP4 79..xx.xx.xx a=sendrecv a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=ssrc:63772004 cname:3ff849bd611036a7 ---------------------------------- B-leg INVITE: INVITE sip:15551112222@togateway.com SIP/2.0 Record-Route: <sip:10.64.54.207;lr;ftag=dff3c585a6154f23898e1121918f1752;did=e21.65c1> Via: SIP/2.0/UDP 10.64.54.207:5060;branch=z9hG4bK919d.5ca83e030c0879093169546cd80591a6.0 Via: SIP/2.0/UDP 79..xx.xx.xx:61379;received=79..xx.xx.xx;rport=61379;branch=z9hG4bKPj2b8935f4460447538d0176dcfb5c74a1 Max-Forwards: 15 From: sip:12125557777@example.com"Updated Name" ;tag=dff3c585a6154f23898e1121918f1752 To: sip:15551112222@togateway.com Contact: <sip:12125557777@79..xx.xx.xx:61379;ob> Call-ID: d793331e066f47bdafc0ad957cb506cc CSeq: 30770 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 609 v=0 o=- 3921660487 3921660487 IN IP4 10.64.54.207 s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio 18400 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122 c=IN IP4 10.64.54.207 b=TIAS:64000 a=ssrc:63772004 cname:3ff849bd611036a7 a=rtpmap:96 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:98 speex/32000 a=rtpmap:99 iLBC/8000 a=fmtp:99 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:120 telephone-event/16000 a=fmtp:120 0-16 a=rtpmap:121 telephone-event/8000 a=fmtp:121 0-16 a=rtpmap:122 telephone-event/32000 a=fmtp:122 0-16 a=sendrecv a=rtcp:18401 Note that I've updated the following on the B-leg: $fn = "Updated Name" // originally nothing $fu = sip:12125557777@example.com // originally sip:12223334444@example.com $tu = sip:15551112222@togateway.com // originally sip:15551112222@example.com While the from uri, and to uri do get updated, the from name is injected in the wrong place of the from header. This only seems to happen when the A-leg is missing the "<" and ">" characters enclosing the from headers. Should I not be updating the pseudo variables in this manner to update the from name, to uri and from uri; is there another way to update these attributes of the header without malforming it? Thank you.

2 1

making HTTP requests in stateless redirects
by Sergio Charrua 08 Apr '24

08 Apr '24

Hi all! For testing purposes, while I am waiting for the ST/SH REST API to be available from other teams, I developed a small python REST API that returns a mockup of a JSON object with some of the required values. The kamailio script, completely stateless as I do not need to keep track of the sessions, iterates through one of the JSON branches and adds the destination URLs to the Contact header and replies a SIP 300 Multiple Choices. This works: the UAC does receive the SIP response with the Contacts , but it also receives a SIP 500 error message right after, and I can't figure out why. And i bet this is simple to solve.... Any clue? I'm aware I'm not processing ACK messages, but the sl_send_reply 403 should do the trick right? ... at least while testing.... Also, as this is a stateless script, is there another way of using async http or making http requests to a REST API without having to use TM Module? Thanks in advance. The kamailio script is has follows: #!KAMAILIO > /* add API http timeout */ > #!define HTTP_API_TIMEOUT 5000 > #!define HTTP_API_ROUTING_ENDPOINT "http://some_python_rest_api/get_route" > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > debug=4 > log_stderror=no > memdbg=5 > memlog=5 > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > /* number of SIP routing processes */ > children=2 > /* set paths to location of modules */ > loadmodule "tm.so" > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "sl.so" > loadmodule "db_mysql.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "ipops.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "uac.so" > loadmodule "counters.so" > loadmodule "http_async_client.so" > loadmodule "jansson.so" > loadmodule "usrloc.so" > /* listen addresses */ > listen=udp:10.20.0.2:5060 > listen=udp:10.20.0.2:5062 > advertised_address="10.20.0.2"; > # ----- http_async_client params ----- > modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) > modparam("http_async_client", "connection_timeout", 2000) > request_route { > #route(HANDLE_DMQ); > route(HANDLE_OPTIONS); > if (is_method("INVITE")) > { > xlog("L_INFO","MAIN - calling TO_CARRIER"); > route(TO_CARRIER); > exit; > } > else{ > sl_send_reply("401","UNAUTHORIZED"); > } > } > route[TO_CARRIER]{ > xlog("L_INFO","TO_CARRIER - calling RELAY_API"); > route(RELAY_API); #Route relay > xlog("L_INFO","TO_CARRIER - return"); > return; > } > # Relay request using the API (response) > route[RELAY_API_RESPONSE] { > xlog("L_INFO","RELAY_API_RESPONSE - got response from REST API"); > if ($http_ok==1 && $http_rs==200) > { > xlog("L_INFO","RELAY_API_RESPONSE - HTTP RESPONSE: > $http_rb\n"); > if (jansson_get("json", $http_rb, "$var(json)")) { > xlog("L_INFO","RELAY_API_RESPONSE - JSON = > $var(json)"); > $var(count) = 0; > jansson_array_size("routes", $var(json), > "$var(size)"); > xlog("L_INFO","RELAY_API_RESPONSE - > jansson_array_size"); > >> >> while ( $var(count) < $var(size) ){ > > >> jansson_get("routes[$var(count)].headers.to.uri", $var(rtjson), "$var(v)"); > > xlog("L_INFO","JSON - routes[$var(count)] >> - $var(v)"); > > #$(avp(mycontacts)[$var(count)]) = >> $avp(mycontacts) + $var(v) + "\r\n"; > > $avp(mycontacts) = $avp(mycontacts) + >> $var(v) + ";"; > > > *append_to_reply("Contact: <" + $var(v) >> +">"+ "\r\n"); /* ---- IS THERE A BETTER ?? ..... */* > > > $var(count) = $var(count) + 1; > > } > > > xlog("L_INFO","RELAY_API_RESPONSE - RELAY"); > xlog("L_INFO","MAIN - calling REPLY_302"); > route(REPLY_302); > return; > } > } > send_reply(500, "API Not Available - http response = $http_rs > $http_ok"); > exit; > } > route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $si:$sp from_number $fU > to_number $ru"); > $http_req(all) = $null; > $http_req(suspend) = 1; > $http_req(timeout) = HTTP_API_TIMEOUT; > $http_req(method) = "POST"; > $http_req(hdr) = "Content-Type: application/json"; > jansson_set("string","from_ip",$si, "$var(http_routing_query)"); > jansson_set("string","from_port",$sp, "$var(http_routing_query)"); > jansson_set("string","from_number",$fU, > "$var(http_routing_query)"); > jansson_set("string","to_number",$ru , "$var(http_routing_query)"); > > xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: > $var(http_routing_query)\n"); > $http_req(body) = $var(http_routing_query); > t_newtran(); > http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); > xlog("L_INFO","RELAY_API - Waiting for Response"); > } > route[REPLY_302] { > # Sends a 300 Multiple Choices back to the proxy that requested the > routing lookup > xlog("L_INFO","REPLY_302 - send reply"); > sl_send_reply("300", "Multiple Choices"); > xlog("L_INFO","REPLY_302 - exit"); > exit; > } > route[HANDLE_OPTIONS]{ > if(is_method("OPTIONS")) > { > sl_send_reply(200, "OK"); > exit; > } > if ($fU=="ping") > { > sl_send_reply("200","OK"); > exit; > } > } SNGrep output: 1 - INVITE from UAC to Kamailio INVITE sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > User-Agent: SIPp > Date: Mon, 08 Apr 2024 15:09:51 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 270 > v=0 > o=root 2021555890 2021555890 IN IP4 10.20.0.1 > s=SIPp > c=IN IP4 10.20.0.1 > t=0 0 > m=audio 18422 RTP/AVP 8 0 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv 2 - Reply from Kamailio to UAC SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 3 - SIP 300 Multiple Choices from Kamailio to UAC SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Contact: <sip:sip:918228990@10.20.0.2@sip.domain.io> > Contact: <sip:+351sip:918228990@10.20.0.2@10.20.0.3> > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 4 - SIP 500 from Kamailio to UAC SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM) > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=4eb2322b7d2b68e6fc3168f503344c21-32679ccd > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 INVITE > Server: kamailio (5.7.4 (x86_64/linux)) > Content-Length: 0 5 - ACK from UAC to Kamailio ACK sip:918228990@10.20.0.2 SIP/2.0 > Via: SIP/2.0/UDP 10.20.0.1:5063;branch=z9hG4bK58aa83f4 > Max-Forwards: 70 > From: "Anonymous" <sip:anonymous@anonymous.invalid:5063>;tag=as192b8891 > To: <sip:918228990@10.20.0.2 > >;tag=57c593265e21c2b70aea50cb414df9cd.32679ccd > Contact: <sip:anonymous@10.20.0.1:5063> > Call-ID: 531668b959674f9f667f1c923d1d6c94@10.20.0.1:5063 > CSeq: 102 ACK > User-Agent: SIPp > Content-Length: 0 and continues with ACK and repeating SIP 500 above.... *Sérgio Charrua* .

4 6

Kamailio Admin Book email
by sadik.oualla.mohamed＠gmail.com 08 Apr '24

08 Apr '24

Hello, Last week, I submitted a request on the Asipto website to purchase the Kamailio Admin Book and received an automatic confirmation email. However, I have not received any further instructions on how to complete the purchase. I replied to the automatic email to follow up, but still, there has been no response. I need to get the book as soon as possible. Please note, I am writing from a different email address than the one I used to make the request on Asipto's site. If there are any issues with their contact email, or if further verification is needed, please let me know how to proceed securely without sharing personal details publicly on this list. Regards, Mohamed.

2 2

customizing the ds_ping_interval dispatcher parameter for destination
by Giuseppe Pandolfi 08 Apr '24

08 Apr '24

Hi, I have one question about dispatcher module in kamailio V.5.3.1. Why the ds_ping_interval parameter is only global value? Is it possible introduce the customization value for single destination? Customization from destination is already possible for ds_ping_from parameter using special attributes in destination list file or db table, is it possible introduce the same management for ds_ping_interval? Thanks, Giuseppe ________________________________ NOTA BENE: Le informazioni contenute nel presente messaggio, compresi gli eventuali allegati, sono strettamente confidenziali ed indirizzate unicamente al destinatario indicato. Qualora non foste il destinatario Vi invitiamo a cancellare il messaggio, ed ogni eventuale documento allegato, non divulgare il contenuto a terze persone ed a volerci avvisare a mezzo posta elettronica. Grazie. A TLC S.r.l. tratta i vostri dati personali nel rispetto della normativa vigente in materia di Privacy (D.Lgs.196/2003 modificato dal D.Lgs.101/2018 e GDPR Regolamento UE n.679/2016). Per consultare l’Informativa completa la invitiamo a visitare il nostro sito web “www.aethra.com<http://www.aethra.com/>” PLEASE NOTE: This message and any attachments are confidential and may contain privileged information. If you are not the intended recipient, you are kindly requested to cancel it, not to disclose the contents to any other person and to notify the sender immediately by return e-mail. Thanks. A TLC S.r.l. collects your personal data in compliance with applicable data protection laws (D.Lgs. 196/2003 amended by Legislative Decree 101/018 and GDPR EU Regulation n. 679/2016). Please visit our website ”www.aethra.com<http://www.aethra.com>” to consult the complete Policy.

1 0

RFC: uptime metric for xhttp_prom module
by Ovidiu Sas 05 Apr '24

05 Apr '24

Hello all, The xhttp_module can export all stats under "stats.get_statistics" RPC command. I was thinking of adding an optional "uptime" stat that will return the kamailio server uptime (like the "core.uptime" RPC command). This will make it easier to add an uptime panel in grafana. The new stat would be controlled by a module parameter (by default disabled). Something like: modparam("xhttp_prom", "xhttp_prom_uptime_stat" 1) And this will generate something like: kam_uptime 671 1712249054631 This can be already implemented in the script using the "prom_gauge_set()" and retrieving the uptime via "jsonrpc_exec()", but it would be nicer and faster to implement it in the module itself. Comments? -ovidiu

3 6

Kamailio Visions 2030 for Appalachia
by Alex Balashov 05 Apr '24

05 Apr '24

For immediate release: HAZARD, Kentucky (1 April 2024)--Evariste Systems LLC, Georgia-based veteran brokers of international charity partnerships, are pleased to announce the conclusion of lengthy trilateral negotiations among Taylor Swift, the Public Investment Fund (the sovereign wealth fund of the Kingdom of Saudi Arabia), and the Kamailio open-source community to bring Kamailio to underserved areas of the US Appalachia region. In a concert in Lexington, Kentucky last Sunday evening, and in her signature style of mystical "reveals" through equivocal song lyrics, Swift auspiciously inaugurated to her Swifties, as die-hard fans of the pop star are known, the beginning of Kamailio Visions 2030. (Editorial note: Swift's agent has threatened copyright litigation in the event that our press agency were to reproduce her meaning-rich lyrics directly, driven by the contention that fair use considerations do not apply to lyrics so "deep" as Swift's. At press time, the matter is being reviewed by our counsel.) Alex Balashov, principal of Evariste, praised Kamailio Visions 2030 as a continuation of Swift's long-standing commitment to forward thinking and innovation: "Taylor has been the driving force behind the use of sustainable, plant-based materials in her 'merch. Today, aided by the internationalisation of Saudi capital, she is helping to bring that same kind of ethical prosperity to Eastern Kentucky, West Virginia and the wider Appalachian coal country." Continued Balashov: "Between Alan Maimon and J.D. Vance, and the revelations of the Purdue Pharma litigation, it's fair to say that the plight of Appalachia, with its high structural unemployment, declining economic base, and the ravages of the opioid epidemic and deaths of despair, has been exposed to a broader audience than ever before. And with that knowledge comes responsibility, because we live with that knowledge now, and we can no longer turn a blind eye. The SIP and VoIP industries have always been a force of conscientious capitalism, and Kamailio a beacon of rectitude in a jaded, amoral, cynical world. A world in which Appalachian children are denied the SIP proxy is not a just world, and we cannot any longer live in such a world." Fred Posner, a deputy Evariste negotiator who was said to have thrived in the pressure of high-stakes negotiations with PIF governor Yasir Al-Rumayyan over the exact implementational modalities of the deal, reviewed the high notes of the landmark economic development accord: "It was a good start to defund the police. But with the unique social challenges of Appalachia, what every struggling family in the ruins of coal country really needs is access to clean, ethically sourced, fair-trade SIP proxies. That is why I am uplifted to say that we will be dropping, through a coordinated, large-scale airlift operation, a Raspberry Pi with Kamailio pre-installed, to every household with an income at or below 250% of the Federal Poverty Level (FPL) in the Appalachian region. The Pi chassis will be dye-free, BPA-free, and will be held together with a plant-based, biodegradable, non-toxic glue that does not produce intoxicating effects when sniffed. By 2030, every child alive in Appalachia today will have had several years to benefit from the power of a miniaturised SIP proxy appliance, and that is why we are calling it Kamailio Visions 2030." When pressed by us, Balashov noted that the prosocial objectives and economic development potential of Kamailio Visions 2030 has drawn scepticism from some quarters. "These are those people who think that Rasperry Pi devices preloaded with Kamailio are not what Appalachia most needs today. Those people would be wrong. As any metaphysicist can tell you, causes do not necessarily resemble their effects." "Taylor and her people know that Kamailio Visions 2030 draws on a long history of highly effective technological and app-based interventions in economically underprivileged areas, such as Nicholas Negroponte's One Laptop per Child (OLPC) scheme, and the famous carpet-bombing of Eritrea with second-generation iPads by Selena Gomez. All of this had tremendous results." "Salutary results", added a representative of the PIF. Balashov pointed us to the foundational problem statement of Kamailio Visions 2030, which cited the concept of "Technik", as developed in "Hybrid Reality", a technocratic manifesto (in e-book form) by much-loved TED visionaries Parag and Ayesha Khanna. There, the authors declared: "Good Technik requires a combination of the attributes that deliver high human development, economic growth, political inclusiveness, and technology preparedness.” "In short", said Balashov, "Taylor, with some help from the efficient deployment of Saudi capital, will bring good technik to Appalachia. The tide of technik lifts all ships." -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

4 3

low performance with no apparent reason
by Sergio Charrua 05 Apr '24

05 Apr '24

Hi all! I have been doing some performance tests with Kamailio 5.7.4 and SIPp. The infrastructure is as follows:3 VMs running on VMWare ESXi running: UAC on 10.20.0.1 with SIPP-> Kamailio on 10.20.0.5 -> UAS on 10.20.0.3 The Kamailio VM has 6 dedicated vCPU of type Intel(R) Xeon(R) Silver 4216 CPU @ 2.10GHz and, 2 NICs and 4Gb RAM and MariaDB 10.6 as DB Backend., all running on a HP G380 host with a gazillion CPUs and a googol disk space! I currently have 3 scripts: - script #1 stateful with RTJson and simulating requests to routing engine and accounting - script #2 stateful but with just a simple routing to UAS, no rules, no DB, - script #3 stateless with a forward to UAS With script #3 I can go up to 2000CPS without issues with CPU at 37%! Above that value, I get retransmissions everywhere. On both scripts #1 and #2, the limit is 330CPS max after which I get a lot of retransmissions, while CPU/Core usage on Kamailio server stays below 10%. So I do not expect this to be a CPU issue. I could not understand why such (low) results, so I followed this article found at https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c <https://www.kamailio.org/docs/openser-performance-tests/#tm-tests-c> and created exact same scenarios, with kamailio script and SIPP templates available on the article, hoping for better results. But I get the same results: between 300 and 330CPS which is far, very far from the 7000CPS found in the article! I understand that I'm using VMs and probably the tests made for the article, which is pretty old already, were made on physical servers. Still, I would not expect 95% of lower performance! Any clue what could be the issue? I suspect NICs, but.... Any tips anyone could share? Thanks in advance! *Sérgio Charrua*

7 31

Statistics for all generated
by Alfredo Román Domínguez 05 Apr '24

05 Apr '24

I am using Kamailio versions 5.5.7 and 5.8.0 as a stateful proxy (with tm module) and I want to have metrics for all the SIP messages received and sent. I am using the statistics module for counting messages. I have in configuration file the following routes to capture events and count them: onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response, tm:local-request, sl:filtered-ack and network:msg) I have found the following problems that I need some hints how to solve them: 1) When a response is received to a request, I can count the response received in the reply_route but I don't find a way that works to count the reply being forwarded to the other side. onsend_route works fine for requests being forwarded. Is there any event_route or any trick that can be used to count the reply being forwarded similar to the onsend_route for requests? 2) When Kamailio receives a CANCEL it generates and sends automatically the 200 OK for the CANCEL and a 487 for the INVITE which started that dialog which is being cancelled, but I don't find a way that works to count those two responses Kamailio generates. 3) After having cancelled a dialog on one direction due to receipt of a CANCEL, kamailio generates local requests on the other direction, a CANCEL and an ACK also when it receives a 487 for the INVITE being cancelled. I don't find the way to count those. onsend_route does not work for locally generated CANCEL and ACK requests and the event_route[tm:local-request] only works for requests created outside of the tm module but not those created by the tm module itself (the CANCEL and the ACKs) since the tm module is not using its own function t_uac_prepare to create those requests (that function is the one that calls the event_route). Is there any statistic generated by the tm module or elsewhere or any other trick so I can count those requests? 4) When Kamailio generates retransmissions how can I count those? onsend_route, onreply_route and event ( corex:reply-out, tn:local_response, sl:local-response) doesn't work for retransmissions. Are there statistics generated somewhere in the tm module for retransmissions? Is there any other way to count those. -- El contenido de este mensaje puede contener información confidencial sometida a secreto profesional. Si usted ha recibido este correo por error, no está permitida su distribución. Por favor elimine cualquier copia, archivo adjunto e infórmenos por esta vía. Los datos personales contenidos en este correo y que han sido facilitados por usted o provienen de fuentes de acceso público serán utilizados por Future Space, S.A., para gestionar nuestra relación contractual o pre-contractual, resolver su solicitud y mantenerle informado sobre los productos o servicios solicitados o similares. Se conservarán mientras exista un interés mutuo para ello, no se oponga al tratamiento o solicite su supresión. Los datos no serán comunicados a terceros, salvo obligación legal. Puede ejercitar sus derechos de acceso, rectificación, supresión, limitación del tratamiento, portabilidad de los datos y oposición mediante escrito dirigido a Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid o mediante correo electrónico a dpo(a)futurespace.com <mailto:dpo@futurespace.com>. Si consideras que se han infringido tus derechos puedes presentar una reclamación ante la www.agpd.es <http://www.agpd.es>. The contents of this electronic mail may be confidential or privileged. If you have received this message by mistake, please do not send it to anyone. We ask you to delete any existing copies, attached files and to notify us accordingly. The personal data provided or obtained from publicly accessible resources will be processed by Future Space, S.A. in order to manage our contractual or pre-contractual relationship, solve your request and keep you informed about activities or products requested by you or similar. Personal data will be stored while a mutual interest exist or either erasure or objection to process them is requested. Data will not be disclosed to third parties unless there is a legal obligation to do so. You have the rights to access, rectify, erasure, restriction of processing, data portability and oppose any processing of your personal data held by Future Space, S.A. To make effective use of your rights, please contact us at Future Space, S.A. Avenida de Tenerife 2, Edificio 2, Planta 1ª, 28703 San Sebastián de los Reyes, Madrid or by email at dpo(a)futurespace.com <mailto:dpo@futurespace.com>. If you consider processing of your personal data infringes data protection regulations you can file a claim with www.agpd.es <http://www.agpd.es>

1 0

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

03 Apr '24

Hello, Kamailio SIP Server v5.8.1 stable release is out. This is a maintenance release of the latest stable branch, 5.8, that includes fixes since the release of v5.8.0. 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 v5.8.1. For more details about version 5.8.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/04/kamailio-v5-8-1-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 Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

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

5.8 compiler warning
by Juha Heinanen 10 Mar '24

10 Mar '24

Kamailio 5.8 build generates on my Debian 12 the following warning: core/mem/q_malloc.c:996:14: warning: 'qm_strnstr' defined but not used [-Wunused-function] 996 | static void *qm_strnstr(const void *b1, int l1, const void *b2, int l2) | ^~~~~~~~~~ Perhaps the definition of the function should be #ifdef'ed by DBG_QM_MALLOC. -- Juha

1 0

SSL key logger for Diffie-Hellman cipher
by Calvin E. 08 Mar '24

08 Mar '24

I've been tasked to use LD_PRELOAD to log SSL keys for TLS connections using a Diffie-Hellman cipher. The first attempt did not work, so I wanted to sanity check whether Kamailio's TLS support is built in such a way that would defeat LD_PRELOAD. The instructions from the vendor are to update /etc/init.d/kamailio like this: env SSLKEYLOG_UDP='127.0.0.1:1234' LD_PRELOAD="/usr/local/src/voipmonitor-git/tools/ssl_keylogger/sslkeylog.so /usr/lib/x86_64-linux-gnu/libssl.so.3" \ start-stop-daemon --start --quiet --pidfile $PIDFILE \ --exec $DAEMON -- $OPTIONS || log_failure_msg " already running" Is there anything special in Kamailio (5.7.3 on Debian 12) that would prevent this from working? Not necessarily something to defeat a keylogger, but maybe the way tls.so gets loaded? The only discrepancy I've noticed is the vendor docs refer to libssl.so.3 not libssl.so.1, but the vendor said that should be OK. I'd love to hear from someone already using VoIPmonitor with Diffie-Hellman ciphers and Kamailio.

6 16

from TCP to UDP and Kamailio doing it wrong
by Alberto Diez 08 Mar '24

08 Mar '24

Hi kamailio community, I have an issue with a Kamailio 5.7. It's listening both in TCP and UDP. In my scenario requests arrive from devices on TCP, but I want to forward to the next hops on UDP. I am avoiding using any type of DNS resolution; since I am always forwarding to predefined next hops I am using the dispatcher module (defined with the IP addresses and transport=udp) or I wrote config files using t_relay_to_udp or t_relay_to with a udp: followed by IP address. I never set up FQDNS only IP addresses and in all of them I explicitly mention UDP. In all of these scenarios I have tried Kamailio insists in trying to use TCP with the next hop and failing because the next_hop is only UDP. I guess because the message arrived using TCP Kamilio does that but I find the behavior very confusing. I nailed down that in my situation its the tm module function prepare_new_uac (in file src/modules/tm/t_fwd.c line 119) being the one that missbehaves. The documentation of the function says literraly : "* t->uac[branch].request.dst will be filled if next_hop !=0 with the result * of the DNS resolution (next_hop, fproto and fsocket). * If next_hop is 0 all the dst members except the send_flags are read-only * (send_flags it's updated) and are supposed to be pre-filled." I found out that even when next_hop is 0 the function changes the t->uac[branch].request.dst proto, socket etc. its there that the kamailio takes the wrong decision, until that function is called within add_auc, the destination proto or the fproto etc is always 1 (UDP) which is what I am trying to force from the config file or the dispatcher definition (I tried both ways). But after calling that function then it comes as a 2 (TCP) The problematic function is a monster of 500 lines and I would like to avoid having to understand it. Since I think the scenario is not so unusual I just want to ask if maybe I am missing something that I should do to avoid the Kamailio to select TCP and in order to have the tm module respecting my preference for UDP (either with dispatcher module and the transport param or with t_relay_to or t_relay_to_udp I don't care the way). Any hints are welcome. Best regards alberto

5 9

direct media between UACs
by Sergio Charrua 08 Mar '24

08 Mar '24

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine either. Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3 and Kamailio has IP .5 The Asterisk servers are used only for testing, nothing serious. However, Kamailio is setup to use RTJson requesting routes to a Routing Server on the same network. And it works fine. Both Asterisk servers have the same dialplan, which only Answers the call and plays MOH on both ends so that RTP audio streams both ways. When making a call on Asterisk Server #1 via command line to go directly to Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/123(a)10.20.0.3 application MusicOnHold() ) the Asterisk #2 receives the call, answers and plays MOH too and I can see RTP streams coming from both ends correctly. However, if I use Kamailio to proxy the call generated from Asterisk #1 to Asterisk #2, using similar command line instruction (CLI> channel originate SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received on Kamailio who then sends it to Asterisk #2, who answers the call and plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is never received, however audio from Asterisk #1 is received by Asterisk #2, which configures a typical One Way Audio issue due to NAT. This is where it gets strange, because there is no NAT, SDP on INVITE and SIP 200 messages seem OK, as far as I understand it. Also, Asterisk servers have SIP configuration with directmedia enabled and NAT disabled to make sure that media is direct. But I have also tried with directmedia disabled and NAT enabled and get identical results. I am most probably missing some tiny detail, but I have no clue.... and I bet it is simple and stupid.... Could another pair of eyes help me with this? What is wrong? Do I really need RTPEngine even when the network has no NAT? I am sure it would work that way, but it doesn't make sense... Here are some screenshots: Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without Kamailio in between: Invite from Asterisk #1 to Asterisk #2 with direct media between both ends: https://drive.google.com/file/d/1eLjT3nr_Rc-UBaf4QhIgZ95bjETOVvxo/view?usp=… Replies from Asterisk #2 to Asterisk #1 with direct media between both ends: https://drive.google.com/file/d/11lLcB-V8rWGSrVqWiit-q9WX2FfqB6BZ/view?usp=… Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to Asterisk #2, with one way audio Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details: https://drive.google.com/file/d/1Cp9xrGcwNmQ9Ks36N_oD1Dj7lxfu-tbH/view?usp=… Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk #1 identical to above: https://drive.google.com/file/d/1mi3FCNjM3luXfENEp-0088XLgcyfXRK6/view?usp=… Reply from Asterisk #2 to Kamailio with SDP details: https://drive.google.com/file/d/1TpMGe2tvpX_5SIpSbm2b3Zro9EuYwcO-/view?usp=… Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2 identical to above: https://drive.google.com/file/d/12jq5APfFwVVPc0vJ3RkcXyN1hBE51fnQ/view?usp=… As we can see, SDP details seem OK, but if I check call flow on Asterisk #1, I can only find 1 RTP channel with audio coming from Asterisk #2 https://drive.google.com/file/d/1iEfvkylZVbthHM5kxkurWh-GAYCYLytl/view?usp=… and the same on Asterisk #2 : https://drive.google.com/file/d/12rvf9Lrwp-MNZvGCwlXEEBsBRmfcinph/view?usp=… My Kamailio.cfg is as follows: #!KAMAILIO # # config file for SIPProxy # - load balancing of VoIP calls # - no TPC listening # # Kamailio (OpenSER) SIP Server v3.2 # - web: http://www.kamailio.org # - git: http://sip-router.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # #!define WITH_DEBUG ###!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_SRC_ALLOWED 8 #!define FLT_DST_INTERNAL_IP 9 #!define FLT_SRC_INTERNAL_IP 10 #!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g" #!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g" #!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g" #!ifndef HTTP_ASYNC_CLIENT_WORKERS #!define HTTP_ASYNC_CLIENT_WORKERS 8 #!endif /* add API http timeout */ #!define HTTP_API_TIMEOUT 5000 #!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route" /* DMQ SIP message sharing */ #!define DMQ_PORT 5062 #!define DMQ_LISTEN "sip:10.20.0.2:5062" #!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062" #!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062" memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " fork=yes children=8 /* comment the next line to enable TCP - all trunks are UDP only */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=udp:10.20.0.2:5062 advertised_address="10.20.0.5"; sip_warning=no; use_dns_failover = on; ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "usrloc.so" loadmodule "nathelper.so" #loadmodule "rtimer.so" #loadmodule "sqlops.so" # --- CPS Limiter # --- end of CPS Limiter loadmodule "ipops.so" loadmodule "textopsx.so" loadmodule "sdpops.so" loadmodule "http_async_client.so" loadmodule "rtjson.so" loadmodule "jansson.so" loadmodule "dmq.so" loadmodule "dmq_usrloc.so" loadmodule "htable.so" loadmodule "dialog.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "log_level_name", "exec") #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) modparam("tm", "remap_503_500", 0) # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) # ----- nathelper params ----- modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") #modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") #modparam("rtimer", "exec", "timer=cdr;route=CDRS") #modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:kamailiorw@10.19.139.113:3306/kamailio") #modparam("dmq", "server_socket", DMQ_SERVER_SOCKET ) modparam("dmq", "server_address", DMQ_SERVER_ADDRESS ) modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 60) modparam("dmq_usrloc", "enable", 1) # -- CPS Limiter modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;") modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "enable_dmq", 1) # ----- http_async_client params ----- modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) modparam("http_async_client", "connection_timeout", 2000) ####### Routing Logic ######## # main request routing logic route { if (is_method("KDMQ") && $Rp == 5062) { dmq_handle_message(); } xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","start|\n"); xlog("L_INFO","===================================================\n"); xlog("L_INFO","New SIP message $rm with call-ID $ci \n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO"," received $pr request $rm $ou\n"); xlog("L_INFO"," source $si:$sp\n"); xlog("L_INFO"," from $fu\n"); xlog("L_INFO"," to $tu\n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO","---------------------------------------------------\n"); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } if ($fU=="ping") { sl_send_reply("200","OK"); exit; } # extract original source ip / port from X-forwarded-For header route(HANDLE_X_FORWARDED_FOR); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # 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(); xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm"); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO", "ROUTE - Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ t_on_failure("MANAGE_FAILURE"); xlog("L_INFO", "ROUTE - Recording Route"); record_route(); if (is_method("INVITE") && is_request()) { if (has_body("application/sdp")) { xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n"); t_on_reply("ON_REPLY"); } } } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","ROUTE - Address Incomplete"); exit; } route(TOCARRIER); xlog("L_INFO", " ********** Route END *************"); } # extract original source ip / port from X-forwarded-For header route[HANDLE_X_FORWARDED_FOR] { if (is_present_hf("X-Forwarded-For")) { $var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:}); $var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:}); } else { $var(source_ip) = $si; $var(source_port) = $sp; } $var(to_number) = $rU; } route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port) from_number $fU to_number $ru"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = HTTP_API_TIMEOUT; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)"); jansson_set("string","from_port",$var(source_port), "$var(http_routing_query)"); jansson_set("string","from_number",$fU, "$var(http_routing_query)"); jansson_set("string","to_number",$var(to_number) , "$var(http_routing_query)"); xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: $var(http_routing_query)\n"); $http_req(body) = $var(http_routing_query); t_newtran(); http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); } # Relay request using the API (response) route[RELAY_API_RESPONSE] { if ($http_ok==1 && $http_rs==200) { xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n"); if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) { xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)"); rtjson_init_routes("$var(rtjson)"); rtjson_push_routes(); # relay the message t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); return; } } send_reply(500, "API Not Available - http response = $http_rs $http_ok"); exit; } onreply_route[ON_REPLY] { xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs"); # on reply if (t_check_status("183|180|200")) { xlog("L_INFO", "ON_REPLY - Fixing Contacts"); # subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a"); //subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f"); } if (has_body("application/sdp")){ if (sdp_remove_line_by_prefix("a=maxptime")){ xlog("L_INFO", "ON_REPLY - remove maxptime "); msg_apply_changes(); } else{ xlog("L_INFO", "ON_REPLY - did not removed maxptime "); } } if (t_check_status("408")) { xlog("L_INFO", "ROUTE - Handling 408 Timeout\n"); } } route[TOCARRIER]{ #using rtjson, unsomment following line route(RELAY_API); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO", "REQINIT - Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "REQINIT - 483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { xlog("L_INFO", "NATDETECT - Entering"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO", "NATDETECT - Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()){ xlog("L_INFO", "NATDETECT - Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS); return; } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO", "WITHINDLG - Entering"); if (!has_totag()) return; if (is_present_hf("Route") && $hdrc(Route)==1) { if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f")) { xlog("L_INFO", "WITHINDLG - Removing the route to self"); remove_hf("Route"); } } # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE|CANCEL")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO", "WITHINDLG - Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { #Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } if(is_method("BYE")) xlog("L_INFO", "WITHINDLG - BYE message from $rU"); route(RELAY); exit; } if ( is_method("ACK|BYE|INVITE|UPDATE") ) { 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. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG"); exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO", "WITHINDLG - Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # Routing to foreign domains ---> NOT USED route[SIPOUT] { xlog("L_INFO", "WITHINDLG - Entering SIPOUT"); if (uri==myself){ xlog("L_INFO", "WITHINDLG - URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO", "WITHINDLG - Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO", " ******** RELAY *******"); xlog("L_INFO", "RELAY - $si $su $ru"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO", "RELAY - branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } xlog("L_INFO", "RELAY - checking method"); if (is_method("INVITE|SUBSCRIBE|UPDATE")) { xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE"); if(!t_is_set("onreply_route")) { xlog("L_INFO", "RELAY - onreply_route NOT SET!"); t_on_reply("ON_REPLY"); # MANAGE_REPLY"); } } if (is_method("INVITE")) { xlog("L_INFO", "RELAY - is INVITE"); t_on_failure("FAILED_RELAY"); if(!t_is_set("failure_route")) { xlog("L_INFO", "RELAY - failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO", "RELAY - t_relay returns FALSE"); route("MANAGE_FAILURE"); #sl_reply_error(); } xlog("L_INFO", "RELAY - exiting"); exit; } failure_route[FAILED_RELAY] { xlog("L_INFO", "FAILED_RELAY - Entering"); if (t_check_status("[4-5][0-9][0-9]")){ xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!"); if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); } } } route[NATMANAGE] { xlog("L_INFO", "NATMANAGE - Entering"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } if (is_request()) { xlog("L_INFO", "NATMANAGE - is_request - $rm from $si"); if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO", "NATMANAGE - adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si"); if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO", "NATMANAGE - Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Manage failure routing cases route[MANAGE_FAILURE] { xlog("L_INFO", "MANAGE_FAILURE - Entering "); route(NATMANAGE); xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled"); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!"); t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!"); t_reply("404","Not found"); exit; } #!endif if (t_check_status("503")){ xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination available"); t_reply("503", "Destination not available"); exit; } if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 "); rtjson_update_branch(); route(NATMANAGE); } Any help would be greatly appreciated. Thanks in advance. Sérgio Charrua

2 3

Kamailio v5.8.0 Released - new major version is out
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Kamailio v5.8.0 is out – it comes with 4 new modules and a large set of improvements touching again more than 50 existing modules. You can read a bit more detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-8-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.8.0 brings more flexibility and optimizations across many existing components and modules, new parameters, functions, variables and transformations. Enjoy Kamailio v5.8.0! Thank you for flying Kamailio! 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 0

Using uac_replace_to when forwarding call to another endpoint
by Marrold 07 Mar '24

07 Mar '24

Hi all, I am using uac_replace_to to replace the TO header on requests. I understand it can only be called once per message and the recommendation is to store the updated value in a pvar and apply it in the branch route. However in my scenario I wish to first try one endpoint and then if it gets a 4XX response forward it to another UAC, and update the TO accordingly. Currently trying to call uac_replace_to twice shows an error and corrupts the TO header. Is there a work around for this? Thanks again Matthew

3 3

Roadmap to next major Kamailio release series v5.8.x
by Daniel-Constantin Mierla 07 Mar '24

07 Mar '24

Hello, discussed a bit during the online Kamailio devel meeting, it is time to set the milestones towards the next major Kamailio release series v5.8.x. If no other suggestions that suit more developers, I would propose to freeze by end of this month or early February, then test for about 4 weeks as usual and release by end of February or during March. If anyone wants to add new features/modules, they have to be published till freezing date, either pushed in the git repository or proposed as pull request. 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

3 12

rtimer intermittent failures
by Marrold 06 Mar '24

06 Mar '24

Hi all, I am using Kamailio 5.7.4 on a Debian 12 machine, with a Python Kemi based config. I am seeing some intermittent failures when accessing an instance variable within a function called by rtimer. I'm using the rtimer module with the following parameters: modparam("rtimer", "timer", "name=hello;interval=5;mode=0;") modparam("rtimer", "exec", "timer=hello;route=ksr_route_hello") I initialise the kamailio class, and instance variable like this: class kamailio: def __init__(self): self.hello = "hi" Within the class I have the following route function: def ksr_route_hello(self, msg, evname): KSR.info("Running ksr_route_hello\n") KSR.info(f"Hello? {self.hello}\n") Then in the logs I see it sometimes works, and sometimes fails: 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Running ksr_route_hello 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Hello? hi 9(15) INFO: <core> [core/kemi.c:106]: sr_kemi_core_info(): Running ksr_route_hello 9(15) ERROR: app_python3 [python_support.c:167]: python_handle_exception(): apy_exec: ksr_route_hello(rtimer): Unhandled exception in the Python code: TypeError: 'NoneType' object cannot be interpreted as an integer The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 1007, in ksr_route_hello KSR.info("Running ksr_route_hello\n") SystemError: <built-in function info> returned a result with an exception set Does anyone know where I'm going wrong? Thanks Matthew

2 2

Question for RFC junkies / kamailio modifying message bodies
by Valentin Christoph 06 Mar '24

06 Mar '24

Hi all, it might be a minor issue, or it might not be an issue at all, but I stumbled over following statements in RFC 3261: Chapter 16.6 Request Forwarding (of a proxy) 1. Copy request The proxy starts with a copy of the received request. The copy MUST initially contain all of the header fields from the received request. Fields not detailed in the processing described below MUST NOT be removed. The copy SHOULD maintain the ordering of the header fields as in the received request. The proxy MUST NOT reorder field values with a common field name (See Section 7.3.1<https://www.rfc-editor.org/rfc/rfc3261.html#section-7.3.1>). The proxy MUST NOT add to, modify, or remove the message body. An actual implementation need not perform a copy; the primary requirement is that the processing for each next hop begin with the same request. Chapter 16.7 Response Processing (at a proxy) 9. Forward response After performing the processing described in steps "Aggregate Authorization Header Field Values" through "Record-Route", the proxy MAY perform any feature specific manipulations on the selected response. The proxy MUST NOT add to, modify, or remove the message body. Unless otherwise specified, the proxy MUST NOT remove any header field values other than the Via header field value discussed in Section 16.7<https://www.rfc-editor.org/rfc/rfc3261.html#section-16.7> Item 3. In particular, the proxy MUST NOT remove any "received" parameter On the other hand, everywhere in the Internet (starting with stackoverflow), you can read it is OK, if a SIP proxy modifies a body, and also kamailio allows it. Might be a question for the coffee break. All the best Christoph

3 2

Kamailio closing tcp connection on downstream responses
by Kjartan S 06 Mar '24

06 Mar '24

Trying to setup Kamailio with MS Teams Direct Routing. I am familiar with the integration steps required by Microsoft (using contact header as FQDN, certified CA for certificate) aswell as the famous guide; https://skalatan.de/en/blog/kamailio-sbc-teams + https://learn.microsoft.com/en-us/microsoftteams/direct-routing-protocols-s… Sending any SIP requests (Invite, ACK, BYE) toward MS is working fine (with provisional response received in Kamailio) - so there is obviously a working connection between Kamailio <-> MS Direct Routing environment. HOWEVER, once Kamailio is forwarding SIP responses to MS (provisional responses/200 OK received from UAS), then Kamailio is closing the TCP connection. Please see attached logs below. Any idea why Kamailio is closing the connection when trying to send provisional responses to MS? - Using t_realy on original invite from MS - tcp_reuse_port=yes, tcp_rd_buf_size = 16384, tcp_connection_lifetime=3605, children=8, socket_workers=4 - No override rules in firewall - OS; debian:11.7 - Notice that TLS connection is made on port :26627 - however, MS sends port :5061 in VIA header (on first Invite). This seems to me to be a conflict in ports. https://www.kamailio.org/wiki/cookbooks/5.4.x/core#reply_route "There is no network route that can be enforced for a SIP reply - it is sent based on Via header, according to SIP RFC3261" ----- Handshake complete for 52.114.76.76:26627 19(43) DEBUG: tls [tls_domain.c:818]: sr_ssl_ctx_info_callback(): SSL handshake done 19(43) DEBUG: tls [tls_server.c:473]: tls_accept(): TLS accept successful 19(43) DEBUG: tls [tls_server.c:476]: tls_accept(): tls_accept: new connection from 52.114.76.76:26627 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256 19(43) DEBUG: tls [tls_server.c:480]: tls_accept(): tls_accept: local socket: 10.129.0.91:5061 19(43) DEBUG: tls [tls_server.c:420]: tls_dump_cert_info(): tls_accept: client certificate subject:/C=US/ST=WA/L=Redmond/O=Microsoft Corporation/CN=sip.pstnhub.microsoft.com 19(43) DEBUG: tls [tls_server.c:424]: tls_dump_cert_info(): tls_accept: client certificate issuer:/C=US/O=Microsoft Corporation/CN=Microsoft Azure RSA TLS Issuing CA 04 ----- Kamailio trying to forward SIP response "183" to MS (response received from UAS). Kamailio finds active connection by ID 13 / 0x7f4ecad0be08 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 13 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2627]: tcpconn_send_put(): tcp connection found (0x7f4ecad0be08), acquiring fd 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2637]: tcpconn_send_put(): c=0x7f4ecad0be08, n=16 23(47) DEBUG: <core> [core/tcp_main.c:3982]: handle_ser_child(): read response= 7f4ecad0be08, 2, fd -1 from 20 (44) 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2665]: tcpconn_send_put(): after receive_fd: c= 0x7f4ecad0be08 n=8 fd=12 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2842]: tcpconn_do_send(): sending... 20(44) DEBUG: {2 1 INVITE e926eb82d1a25bdc9ba03048cdbc5c77} <core> [core/tcp_main.c:2878]: tcpconn_do_send(): after real write: c= 0x7f4ecad0be08 n=1191 fd=12 ----- Kamailio immediately closes the connection due to EOF. 19(43) DEBUG: <core> [core/tcp_read.c:280]: tcp_read_data(): EOF on connection 0x7f4ecad0be08 (state: 0, flags: 4018) - FD 10, bytes 0, rd-flags 10000 ([52.114.76.76]:26627 -> [52.114.76.76]:5061)19(43) DEBUG: <core> [core/tcp_read.c:1544]: tcp_read_req(): EOF 19(43) DEBUG: <core> [core/io_wait.h:597]: io_watch_del(): DBG: io_watch_del (0x555b3cf20260, 10, -1, 0x10) fd_no=2 called 19(43) DEBUG: <core> [core/tcp_read.c:1927]: handle_io(): removing from list 0x7f4ecad0be08 id 13 fd 10, state 2, flags 4018, main fd 57, refcnt 2 ([52.114.76.76]:26627 -> [52.114.76.76]:5061) 19(43) DEBUG: <core> [core/tcp_read.c:1702]: release_tcpconn(): releasing con 0x7f4ecad0be08, state -1, fd=10, id=13 ([52.114.76.76]:26627 -> [52.114.76.76]:5061)

1 0

RADIUS ACC - Interim Updates
by duarterocha91＠gmail.com 06 Mar '24

06 Mar '24

Hello all, I can't seem to find a way to make interim accounting requests for RADIUS. Can you help me with this? I can't find any native way for doing this. I've thought of getting a custom timer for each dialog and run a event route similar do dialog:start or dialog:end. Is that possible? Cheers, Duarte Rocha

2 2

Add XML to INVITE
by David Villasmil 06 Mar '24

06 Mar '24

Hello guys, I'm trying to add an xml with a boundary to an outgoing INVITE. But if i do: set_body_multipart("$rb", "application/sdp", "delimiter"); msg_apply_changes(); append_body_part("$var(something)", "application/pidf+xml"); msg_apply_changes(); but kamailio adds the SDP, but NOT the second $var(something). If i switch them and add firs the var, and then the SDP. kamailio adds the contents of $var(something) but NOT the SDP.... point is it doesn't add the second append Thanks!!! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

dispatcher - sip options per gw
by marek 05 Mar '24

05 Mar '24

hi, i have kamailio gw (5.6.x) for voxbone. dispatcher module voxbone has separated inbound/outboud and prohibit sip options to "inbound" IP #traffic from voxbone 20 sip:81.201.82.45:5060;transport=udp 1 0 duid=abc;socket=udp:x.x.x.x:5060 #traffic TO voxbone 30 sip:81.201.89.110:5060;transport=udp 0 0 duid=abc;socket=udp:x.x.x.x:5060 is it possible send sip options per GW with dispatcher module? i tried flag "1 (bit at index 0 - 1 <<0) - inactive destination" https://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#idm1059 Marek

2 2

direct media between UACs
by Sergio Charrua 05 Mar '24

05 Mar '24

Hi all! got a weird behavior that I cannot understand the reason for... In our LAB environment, we have 2 Asterisk instances (version 13.38.3 and chan_sip) and 1 Kamailio 5.7 in between. All servers are in the same network, so, there is no NAT involved. No RTPEngine neither. Network is 10.20.0.0/24 and Asterisk #1 has IP .1 Asterisk #2 has IP .3 and Kamailio has IP .5 The Asterisk servers are used only for testing, nothing serious. However, Kamailio is setup to use RTJson requesting routes to a Routing Server on the same network. And it works fine. Both Asterisk servers have the same dialplan, which only Answers the call and plays MOH on both ends so that RTP audio streams both ways. When making a call on Asterisk Server #1 via command line to go directly to Asterisk Server #2 without using Kamailio (CLI> channel originate SIP/ 123(a)10.20.0.*3* application MusicOnHold() ) the Asterisk #2 receives the call, answers and plays MOH too and I can see RTP streams coming from both ends correctly. However, if I use Kamailio to proxy the call generated from Asterisk #1 to Asterisk #2, using similar command line instruction (CLI> channel originate SIP/123(a)10.20.0.5 application MusicOnHold() ), the call is indeed received on Kamailio who then sends it to Asterisk #2, who answers the call and plays MOH, *but* despite the audio stream being sent to Asterisk #1 it is never received, however audio from Asterisk #1 is received by Asterisk #2, which configures a typical One Way Audio issue due to NAT. This is where it gets strange, because there is no NAT, SDP on INVITE and SIP 200 messages seem OK, as far as I understand it. Also, Asterisk servers have SIP configuration with directmedia enabled and NAT disabled to make sure that media is direct. But I have also tried with directmedia disabled and NAT enabled and get identical results. I am most probably missing some tiny detail, but I have no clue.... and I bet it is simple and stupid.... Could another pair of eyes help me with this? What is wrong? Do I really need RTPEngine even when the network has no NAT? I am sure it would work that way, but it doesn't make sense... Here are some screenshots: Call Scenario #1 - direct call from Asterisk #1 to Asterisk #2 without Kamailio in between: Invite from Asterisk #1 to Asterisk #2 with direct media between both ends: [image: image.png] Replies from Asterisk #2 to Asterisk #1 with direct media between both ends: [image: image.png] Call Scenario #2 - call from Asterisk #1 using Kamailio to relay call to Asterisk #2, with one way audio Invite from Asterisk #1 to Asterisk #2 via Kamailio with SDP details: [image: image.png] Invite from Kamailio relayed to Asterisk #2 with SDP details from Asterisk #1 identical to above: [image: image.png] Reply from Asterisk #2 to Kamailio with SDP details: [image: image.png] Reply from Kamailio to Asterisk #2 with SDP details from Asterisk #2 identical to above: [image: image.png] As we can see, SDP details seem OK, but if I check call flow on Asterisk #1, I can only find 1 RTP channel with audio coming from Asterisk #2 [image: image.png] and the same on Asterisk #2 : [image: image.png] My Kamailio.cfg : #!KAMAILIO # # config file for SIPProxy # - load balancing of VoIP calls # - no TPC listening # # Kamailio (OpenSER) SIP Server v3.2 # - web: http://www.kamailio.org # - git: http://sip-router.org # # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # #!define WITH_DEBUG ###!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailio@10.20.0.1:3306/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 #!define FLT_SRC_ALLOWED 8 #!define FLT_DST_INTERNAL_IP 9 #!define FLT_SRC_INTERNAL_IP 10 #!substdef "!INTERNAL_IP_NET!10.20.0.0/24!g" #!substdef "!INTERNAL_IP_ADDR!10.20.0.2!g" #!substdef "!EXTERNAL_IP_ADDR!10.20.0.2!g" #!ifndef HTTP_ASYNC_CLIENT_WORKERS #!define HTTP_ASYNC_CLIENT_WORKERS 8 #!endif /* add API http timeout */ #!define HTTP_API_TIMEOUT 5000 #!define HTTP_API_ROUTING_ENDPOINT "http://10.246.212.40:7778/get_route" /* DMQ SIP message sharing */ #!define DMQ_PORT 5062 #!define DMQ_LISTEN "sip:10.20.0.2:5062" #!define DMQ_SERVER_ADDRESS "sip:10.20.0.2:5062" #!define DMQ_NOTIFICATION_ADDRESS "sip:10.20.0.4:5062" memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " fork=yes children=8 /* comment the next line to enable TCP - all trunks are UDP only */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=tcp:10.20.0.5:5060 advertise 10.20.0.5:5060 listen=udp:10.20.0.2:5062 advertised_address="10.20.0.5"; sip_warning=no; use_dns_failover = on; ####### Modules Section ######## #set module path mpath="/usr/local/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "usrloc.so" loadmodule "nathelper.so" #loadmodule "rtimer.so" #loadmodule "sqlops.so" # --- CPS Limiter # --- end of CPS Limiter loadmodule "ipops.so" loadmodule "textopsx.so" loadmodule "sdpops.so" loadmodule "http_async_client.so" loadmodule "rtjson.so" loadmodule "jansson.so" loadmodule "dmq.so" loadmodule "dmq_usrloc.so" loadmodule "htable.so" loadmodule "dialog.so" #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "log_level_name", "exec") #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 5000) modparam("tm", "fr_inv_timer", 60000) modparam("tm", "remap_503_500", 0) # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) # ----- nathelper params ----- modparam("nathelper", "received_avp", "$avp(s:rcv)") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") #modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") #modparam("rtimer", "exec", "timer=cdr;route=CDRS") #modparam("sqlops", "sqlcon", "ca=>mysql:// kamailio:kamailiorw@10.19.139.113:3306/kamailio") #modparam("dmq", "server_socket", DMQ_SERVER_SOCKET ) modparam("dmq", "server_address", DMQ_SERVER_ADDRESS ) modparam("dmq", "notification_address", DMQ_NOTIFICATION_ADDRESS ) modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 60) modparam("dmq_usrloc", "enable", 1) # -- CPS Limiter modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=10;") modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=120;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("dialog", "profiles_with_value", "concurrent_calls") modparam("dialog", "enable_dmq", 1) # ----- http_async_client params ----- modparam("http_async_client", "workers", HTTP_ASYNC_CLIENT_WORKERS) modparam("http_async_client", "connection_timeout", 2000) ####### Routing Logic ######## # main request routing logic route { if (is_method("KDMQ") && $Rp == 5062) { dmq_handle_message(); } xlog("L_INFO"," ********** Route START ***********"); # log the basic info regarding this call xlog("L_INFO","start|\n"); xlog("L_INFO","===================================================\n"); xlog("L_INFO","New SIP message $rm with call-ID $ci \n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO"," received $pr request $rm $ou\n"); xlog("L_INFO"," source $si:$sp\n"); xlog("L_INFO"," from $fu\n"); xlog("L_INFO"," to $tu\n"); xlog("L_INFO","---------------------------------------------------\n"); xlog("L_INFO","---------------------------------------------------\n"); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) { sl_send_reply("200","OK"); exit; } if ($fU=="ping") { sl_send_reply("200","OK"); exit; } # extract original source ip / port from X-forwarded-For header route(HANDLE_X_FORWARDED_FOR); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # 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(); xlog("L_INFO", "ROUTE - Exiting after Retransmission check - method $rm"); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO", "ROUTE - Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ t_on_failure("MANAGE_FAILURE"); xlog("L_INFO", "ROUTE - Recording Route"); record_route(); if (is_method("INVITE") && is_request()) { if (has_body("application/sdp")) { xlog("L_INFO", "ROUTE - goiing to t_on_reply[ON_REPLY]\n"); t_on_reply("ON_REPLY"); } } } if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","ROUTE - Address Incomplete"); exit; } route(TOCARRIER); xlog("L_INFO", " ********** Route END *************"); } # extract original source ip / port from X-forwarded-For header route[HANDLE_X_FORWARDED_FOR] { if (is_present_hf("X-Forwarded-For")) { $var(source_ip) = $(hdr(X-Forwarded-For){s.select,0,:}); $var(source_port) = $(hdr(X-Forwarded-For){s.select,1,:}); } else { $var(source_ip) = $si; $var(source_port) = $sp; } $var(to_number) = $rU; } route[RELAY_API] { xlog("L_INFO","RELAY_API - from_ip $var(source_ip):$var(source_port) from_number $fU to_number $ru"); $http_req(all) = $null; $http_req(suspend) = 1; $http_req(timeout) = HTTP_API_TIMEOUT; $http_req(method) = "POST"; $http_req(hdr) = "Content-Type: application/json"; jansson_set("string","from_ip",$var(source_ip), "$var(http_routing_query)"); jansson_set("string","from_port",$var(source_port), "$var(http_routing_query)"); jansson_set("string","from_number",$fU, "$var(http_routing_query)"); jansson_set("string","to_number",$var(to_number) , "$var(http_routing_query)"); xlog("L_INFO","RELAY_API - API ASYNC ROUTING REQUEST: $var(http_routing_query)\n"); $http_req(body) = $var(http_routing_query); t_newtran(); http_async_query(HTTP_API_ROUTING_ENDPOINT, "RELAY_API_RESPONSE"); } # Relay request using the API (response) route[RELAY_API_RESPONSE] { if ($http_ok==1 && $http_rs==200) { xlog("L_INFO","RELAY_API_RESPONSE - RESPONSE: $http_rb\n"); if (jansson_get("rtjson", $http_rb, "$var(rtjson)")) { xlog("L_INFO","RELAY_API_RESPONSE - $var(rtjson)"); rtjson_init_routes("$var(rtjson)"); rtjson_push_routes(); # relay the message t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); return; } } send_reply(500, "API Not Available - http response = $http_rs $http_ok"); exit; } onreply_route[ON_REPLY] { xlog("L_INFO", "ON_REPLY - In onreply_route[ON_REPLY] $rs"); # on reply if (t_check_status("183|180|200")) { xlog("L_INFO", "ON_REPLY - Fixing Contacts"); # subst_hf("Contact","/@.*:/@EXTERNAL_IP_ADDR:/","a"); //subst_hf("Record-Route","/INTERNAL_IP_ADDR/EXTERNAL_IP_ADDR/","f"); } if (has_body("application/sdp")){ if (sdp_remove_line_by_prefix("a=maxptime")){ xlog("L_INFO", "ON_REPLY - remove maxptime "); msg_apply_changes(); } else{ xlog("L_INFO", "ON_REPLY - did not removed maxptime "); } } if (t_check_status("408")) { xlog("L_INFO", "ROUTE - Handling 408 Timeout\n"); } } route[TOCARRIER]{ #using rtjson, unsomment following line route(RELAY_API); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO", "REQINIT - Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "REQINIT - 483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("L_INFO","REQINIT - Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { xlog("L_INFO", "NATDETECT - Entering"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO", "NATDETECT - Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()){ xlog("L_INFO", "NATDETECT - Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO", "NATDETECT - Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO", "NATDETECT - NAT Detect set FLT_NTS = " + FLT_NATS); return; } # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO", "WITHINDLG - Entering"); if (!has_totag()) return; if (is_present_hf("Route") && $hdrc(Route)==1) { if (search_hf("Route", ".*EXTERNAL_IP_ADDR.*", "f")) { xlog("L_INFO", "WITHINDLG - Removing the route to self"); remove_hf("Route"); } } # sequential request within a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE|CANCEL")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO", "WITHINDLG - Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { #Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } if(is_method("BYE")) xlog("L_INFO", "WITHINDLG - BYE message from $rU"); route(RELAY); exit; } if ( is_method("ACK|BYE|INVITE|UPDATE") ) { 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. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO", "WITHINDLG - Finishing WITHINDLG"); exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO", "WITHINDLG - Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO", "WITHINDLG - Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # Routing to foreign domains ---> NOT USED route[SIPOUT] { xlog("L_INFO", "WITHINDLG - Entering SIPOUT"); if (uri==myself){ xlog("L_INFO", "WITHINDLG - URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO", "WITHINDLG - Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO", " ******** RELAY *******"); xlog("L_INFO", "RELAY - $si $su $ru"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|CANCEL|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO", "RELAY - branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } xlog("L_INFO", "RELAY - checking method"); if (is_method("INVITE|SUBSCRIBE|UPDATE")) { xlog("L_INFO", "RELAY - is INVITE|SUBSCRIBE|UPDATE"); if(!t_is_set("onreply_route")) { xlog("L_INFO", "RELAY - onreply_route NOT SET!"); t_on_reply("ON_REPLY"); # MANAGE_REPLY"); } } if (is_method("INVITE")) { xlog("L_INFO", "RELAY - is INVITE"); t_on_failure("FAILED_RELAY"); if(!t_is_set("failure_route")) { xlog("L_INFO", "RELAY - failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO", "RELAY - t_relay returns FALSE"); route("MANAGE_FAILURE"); #sl_reply_error(); } xlog("L_INFO", "RELAY - exiting"); exit; } failure_route[FAILED_RELAY] { xlog("L_INFO", "FAILED_RELAY - Entering"); if (t_check_status("[4-5][0-9][0-9]")){ xlog("L_INFO", "FAILED_RELAY - Could not reach destination endpoint!"); if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); } } } route[NATMANAGE] { xlog("L_INFO", "NATMANAGE - Entering"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { xlog("L_INFO", "NATMANAGE - nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO", "NATMANAGE - NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } if (is_request()) { xlog("L_INFO", "NATMANAGE - is_request - $rm from $si"); if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO", "NATMANAGE - adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { xlog("L_INFO", "NATMANAGE - is_reply - $rm from $si"); if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO", "NATMANAGE - Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Manage failure routing cases route[MANAGE_FAILURE] { xlog("L_INFO", "MANAGE_FAILURE - Entering "); route(NATMANAGE); xlog("L_INFO", "MANAGE_FAILURE - t_is_canceled"); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 3XX returned!!"); t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { xlog("L_INFO", "MANAGE_FAILURE - SIP 401|407 returned!!"); t_reply("404","Not found"); exit; } #!endif if (t_check_status("503")){ xlog("L_INFO", "MANAGE_FAILURE - SIP 503 returned : no destination available"); t_reply("503", "Destination not available"); exit; } if (rtjson_next_route()) { xlog("L_INFO", "MANAGE_FAILURE - Getting next route!!"); t_on_branch("MANAGE_BRANCH"); t_on_failure("MANAGE_FAILURE"); route(RELAY); exit; } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xlog("L_INFO","MANAGE_BRANCH - New branch [$T_branch_idx] to $ru\n"); xlog("L_INFO", "MANAGE_BRANCH - branch_route MANAGE_BRANCH 1 "); rtjson_update_branch(); route(NATMANAGE); } Any help would be greatly appreciated. Thanks in advance. *Sérgio *

1 0

Question syntax check / plausi check kamailio.cfg
by Valentin Christoph 04 Mar '24

04 Mar '24

Dear all, Is there any other possibility to (automatically) check the plausibility/syntax of an existing kamailio.cfg, besides kamailio -c -f kamailio.cfg ? Thanks, Christoph

3 2

DMQ module failure
by Sergio Charrua 01 Mar '24

01 Mar '24

Hi all! I'm testing the DMQ module, but having difficulties setting it up. [...] My kamailio.cfg is as follows : [...] listen=udp:0.0.0.0:5060 listen=tcp:0.0.0.0:5060 [...] modparam("dmq", "server_address", "sip:10.20.0.2:5062") #DMQ_SERVER_ADDRESS modparam("dmq", "notification_address", "sip:10.20.0.4:5062 ")#DMQ_NOTIFICATION_ADDRESS modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) modparam("dmq", "ping_interval", 15) modparam("dmq_usrloc", "enable", 1) If I check syntax (using -c parameter) all is ok. Once I start Kamailio, the following error appears on system log: INFO: dmq [dmq.c:164]: make_socket_str_from_uri(): unknown transport protocol - fall back to udp ERROR: dmq [dmq.c:241]: mod_init(): server_uri is not a socket the proxy is listening on ERROR: <core> [core/sr_module.c:1030]: init_mod(): Error while initializing module dmq (/usr/local/lib64/kamailio/modules/dmq.so) I have used UDP as well as SIP as protocols for Server Address and Notification Address, used port 5060 and 5062 (and others too), even added a new listen=udp:0.0.0.0:5062, as well as listen=tcp:0.0.0.0:5062 but the result is always the same. Using Kamailio 5.7.4 on RHEL 9 with SELINUX set to enforcing (if relevant). No firewall active. local IP address is 10.20.0.1 second kamailio node IP address is 10.20.0.4 Any ideas? At first sight, it seems to be all set as per documentation. I even found other configurations online, and apart from the IP address, all is similar. Thanks in advance, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

2 2

Answer OPTIONS from an Asterisk
by Ludovic ELUECQUE 28 Feb '24

28 Feb '24

Hello, I've been using Kamailio to replace a Telcobridge ProSBC for several days now, but I've been stuck for a whole day on a simple concept My Kamaillio is connected to an Asterisk to which it redirects requests. However, this Asterisk sends OPTIONS requests in order to ping the phone. On the Kamailio side, I don't know how to respond to these requests. I've done tons of research but I can't find anything similar, maybe there's something very obvious I'm missing.... If I t_relay the message it never receives a reply via the phone's public ip/port. Thanks in advance if anyone could help me with this. Please apologize if it's a dumb thing. SIP OPTIONS: 2024/02/28 23:28:25.181663 100.66.138.5:5060 -> 100.66.138.7:5060 OPTIONS sip:110@100.66.138.7:5060 SIP/2.0 Via: SIP/2.0/UDP 100.66.138.5:5060 ;rport;branch=z9hG4bKPje213d53f-fd1d-47f2-b0f8-fd3a8bf5188c From: <sip:110@100.66.138.5>;tag=46e51407-77b0-47f1-af73-cd8bec95b179 To: <sip:110@100.66.138.7> Contact: <sip:110@100.66.138.5:5060> Call-ID: 74401738-d874-4c5c-a4b5-b74514361043 CSeq: 50224 OPTIONS Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 ============= kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_RTPENGINE # define WITH_ANTIFLOOD #!define WITH_WEBSOCKETS #!define WITH_TLS #!define WITH_NAT ... #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### DEFINE VARIABLE ######## #!substdef "!MY_IP_ADDR!X.X.X.X!g" #!substdef "!MY_WS_PORT!8080!g" #!substdef "!MY_WSS_PORT!4443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_PORT!4444!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" #!substdef "!MY_MSRP_ADDR!tls:MY_IP_ADDR:MY_MSRP_PORT!g" #!substdef "!MSRP_MIN_EXPIRES!1800!g" #!substdef "!MSRP_MAX_EXPIRES!3600!g" ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_DEBUG #!define DBGLEVEL 3 #!else #!define DBGLEVEL 2 #!endif #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!trydef DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 # FLB_ - per branch flags #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### /* LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR, ... */ debug=DBGLEVEL /* set to 'yes' to print log messages to terminal or use '-E' cli option */ log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases - it can be set many times */ # alias="sip.mydomain.com" /* listen sockets - if none set, Kamailio binds to all local IP addresses * - basic prototype (full prototype can be found in Wiki - Core Cookbook): * listen=[proto]:[localip]:[lport] advertise [publicip]:[pport] * - it can be set many times to add more sockets to listen to */ listen=udp:X.X.X.X:5060 listen=udp:100.66.138.7:5060 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_JSONRPC tcp_accept_no_cl=yes #!endif #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif /* set it to yes to enable sctp and load sctp.so module */ enable_sctp=no ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif #!ifdef WITH_JSONRPC loadmodule "xhttp.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "textopsx.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "rtpengine.so" #loadmodule "rtpproxy.so" #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" # loadmodule "nathelper.so" #!endif #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE #!else #!loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) #rtp modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") modparam("rtpengine", "force_send_interface", "100.66.138.7") #modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) /* set it to 0 to disable Path handling */ modparam("registrar", "use_path", 1) /* save Path even if not listed in Supported header */ modparam("registrar", "path_mode", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- modparam("usrloc", "timer_interval", 60) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "use_domain", MULTIDOMAIN) /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "load_backends", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- #modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- #!modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@sip.super.fr") #modparam("nathelper", "sipping_method", "OPTIONS") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 40) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ######## websocket ############ #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR tcp_accept_no_cl=yes # tcp_rd_buf_size=65536 #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif ######## SECURITY ####### server_signature=no sip_warning=0 user_agent_header="User-Agent: SIP Server" ####### 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 { route(REQINIT); route(NATDETECT); route(REGISTER); route(INVITE); route(ACK); route(BYE); return; } # Wrapper for relaying requests #Check initial detail route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent|pplsip") { # 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") && uri==myself && $rU==$null) { sl_send_reply("200", "Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } route[REGISTER] { if (is_method("REGISTER")) { t_on_reply("1"); force_send_socket(100.66.138.7:5060); t_relay_to_udp("100.66.138.5", "5060"); exit(); } } route[INVITE] { if (( is_method("REFER")) || is_method("INVITE") ||is_method("NOTIFY")) { record_route(); if (src_ip == "100.66.138.5") { if(lookup("location") && !is_method("OPTIONS")) { xlog("L_NOTICE", "user enregistrax"); force_send_socket(X.X.X.X:5060); rtpengine_manage("replace-origin replace-session-connection media-address=X.X.X.X direction=priv direction=pub"); t_relay(); } else { sl_send_reply("403","Not registred"); } } else { force_send_socket(100.66.138.7:5060); rtpengine_manage("replace-origin replace-session-connection media-address=100.66.138.7 direction=pub direction=priv"); t_relay_to_udp("100.66.138.5", "5060"); } } } route[ACK] { if (is_method("ACK")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } route[BYE] { if (is_method("BYE|CANCEL")) { if (src_ip == "100.66.138.5") { force_send_socket(udp:X.X.X.X:5060); t_relay(); } else { force_send_socket(udp:100.66.138.7:5060); t_relay(); } exit(); } } onreply_route { if (status=~ "180|183") { rtpengine_manage("replace-origin replace-session-connection"); } if (status=~ "200") { if(has_body("application/sdp")) { rtpengine_manage("replace-origin replace-session-connection"); } } } onreply_route[1] { if (status=~ "2[0-9][0-9]"){ save("location"); xlog("L_NOTICE", "enregistrement"); } } ====================

2 1

best approach to setup SIP Session sharing between servers
by Sergio Charrua 28 Feb '24

28 Feb '24

Hi all! I am having some difficulties/doubts on what would be the best approach to let multiple Kamailio servers share their SIP sessions between each other. The goal is to have HA on Kamailio cluster such that if a call is received and initiated on Kamailio #1, and during any moment of the call (before or while established) if the server #1 goes down or Kamailio stops working for any reason, the call can be processed by Kamailio server #n available in the cluster. I do not want to mess with Virtual IPs or DNS, that is not the point, but instead need to get a proper way to share SIP sessions between Kamailio servers such that any server could continue the call without issue. I have checked the Dialog and DMQ module, but I am not sure if that is the way to go.... Could any one share a solution for this? Greatly appreciated. Cheers, *Sérgio Charrua* *www.voip.pt <http://www.voip.pt/>*

8 11

Clarification on pv_www_authenticate
by James Lipski 28 Feb '24

28 Feb '24

Hello, I would like get further information on the "pv_www_authenticate" function from auth module. For the response code "-1", which is listed as "generic error", what exactly is covered when this function returns this? Is it just everything else that isn't listed reason (invalid password, nonce expired, no credentials, etc...). On my production system, I'm seeing this as a response code on some registration requests, and just want to know what is causing this (enabling full verbose logging is a bit difficult on my production system, and want to try to replicate it on my test environment, however just need clarification on what this covers). Thank you.

2 1

Topos methods_nocontact parameter
by Benoit Panizzon 27 Feb '24

27 Feb '24

Hi I'm having another go at topos trying to solve an issue of sip messages getting to big for certain clients to process. Unfortunately some methods (ACK/PRACK/BYE probably CANCEL) do not contain a contact header, therefore the topology is still revealed (and potentially a lot of via header inserted increasing message size). Is it safe to set methods_nocontact to an empty string to force creation of a Contact header for every SIP method? Is there another way to completely hide topology aka keeping header small? 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 7

How to keep SIP messages small, possibly below fragmentation limit?
by Benoît Panizzon 26 Feb '24

26 Feb '24

Hi In the process of getting rid of our commercial B2Bua SBC and handling all clients directly via Kamailio... We have come across some (unfortunately many we have in use) CPE which have an issue with fragmented packets or which simply do not allocate enough memory to parse or create messages of a certain size. Their SIP stack just crashes with errors like 'message to large for send buffer' and similar. This affects messages with containing as little as 5 via and 4 record-route header and a couple of codec in the SDP. Is there a recipe for trying to keep packets as small as possible? I would like to avoid using the topos module as I issues with this module completely failing in certain situations. -- 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 ______________________________________________________

2 2

Development is open for future release of v5.9.x series
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 was created, therefore the master branch is open for adding new features, to be part of future release series v5.9.x (or whatever version is decided for next series). Any bug fix committed to master that applies to 5.8.x or older stable branches should be backported as usual with "git cherry-pick -x ..." to appropriate branches like 5.8 or 5.7. Expect that v5.8.0 will be released in a few weeks from now. Based on the workflow used during the past years, the next future release v5.9.0 should be out after another 8-10 months of development, plus 1-2 months of testing, so sometime during the last part of 2024 or the beginning of 2025. 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 0

Branch 5.8 has been created
by Daniel-Constantin Mierla 23 Feb '24

23 Feb '24

Hello, the branch 5.8 has been created, to be used for releasing v5.8.x series. To check out this branch, the following commands can be used: git clone https://github.com/kamailio/kamailio kamailio-5.8 cd kamailio-5.8 git checkout -b 5.8 origin/5.8 Pushing commits in this branch: git push origin 5.8:5.8 Note that 5.8 is an official stable branch, so only bug fixes, missing kemi exports (discuss on sr-dev if not sure) or improvements to documentation or helper tools can be pushed to this branch. As usual, if there is a bug fixed, commit and push first to master branch and then cherry pick to 5.8 branch: git cherry-pick -x COMMITID In few weeks, the first release from branch 5.8 will be out, respectively Kamailio v5.8.0. 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 0

Building Debian packages for further modules
by Christian Berger 22 Feb '24

22 Feb '24

Hi, for our workflow we build our own Debian packages for kamailio. We are currently trying and failing to create a package for the kafka modules. What we have tried so far: Adding the "Build-Depends:" entries for librdkafka-dev in debian/control Adding a "Package" in debian/control Adding kafka to the PACKAGE_GROUPS in debian/rules So far it seems to build the module and a package for it, but the package only contains metadata, but no .so file. We are currently running out of ideas. What else can we do? Best wishes Christian Berger -- Christian Berger - berger(a)sipgate.de Telefon: +49 (0)211-63 55 55-0 Telefax: +49 (0)211-63 55 55-22 sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 2

Kemi questions
by Stefan-Cristian Mititelu 21 Feb '24

21 Feb '24

Hello, When writing kemi cfg (python in my case), is there any way to get rid of standard cfg shared variables and use kemi cfg variables instead? E.g. Instead of $shv(blabla) (in standard config) can I use some global python var called 'blabla'? Is it seen and shared by all kamailio processes? Or I still need to use KSR.pv.get/set("$shv(blabla)") ? Thank you, Stefan

2 3

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.

4 4

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.

2 1

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/>

1 0

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

1 0

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 ______________________________________________________

1 1

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

1 0

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 ______________________________________________________

2 3

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.

1 0

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

3 2

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 ;)

1 0

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

2 3

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

3 4

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

2 2

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

2 4

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

3 3

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

1 0

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

2 3

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

3 3

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

1 0

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.

1 0

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

3 6

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

1 0

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

1 0

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

1 1

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,

3 3

What happens when using exit or return in branch route in parallel branching?
by Benoit Panizzon 31 Jan '24

31 Jan '24

Hi I was wondering what exactly happens with parallel branching in this situation: branch_route[BR_T] { if (condition) { # Stuff below not required exit or return } do other stuff before relaying } Will this break, stop the processing of all branches with higher index as the one being processed? From what I observe, I fear this is the case. Could I safely use return(1) (vs return which probably is return(0) thus eq exit)? Or would the only safe option be to wrap the 'other stuff' in an else statement? 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 5

install location when compiling
by David Villasmil 30 Jan '24

30 Jan '24

Hello guys, i compiled kamailio and i want it installed on debian's deb location, i.e.: /etc/kamailio /usr/lib/x86_64-linux-gnu/kamailio/modules etc.. I know if i do make prefix=/ all make prefix=/ install the config will end up on /etc/kamailio, but then the modules will be on /lib64... i don't see a way of doing this, is there? Thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Same number of max concurrent requests on backends
by Unai Rodriguez 30 Jan '24

30 Jan '24

Dear List, We're load balancing MRCP requests using Kamailio dispatcher module's call load distribution algorithm. We want to achieve the same maximum number of concurrent sessions on each MRCP backend but because the duration of each request varies, we’re unable to achieve this (at the moment we send the same number of requests to each backend but they all have different maximum concurrent requests). Is there a way/module that will send requests to the backend with the least number of ongoing sessions? My guess is, this module needs to connect to the backends to know their status. Any pointers appreciated. Thank you With best wishes, Unai Rodriguez

3 6

Kamailio 5.6 (and 5.7) core dumping.
by Mattis Lind 29 Jan '24

29 Jan '24

Hello Kamailio list! We have a scenario that makes use of the UAC-module to send SIP MESSAGE and then in some cases the Kamailio process core dumps after some time after processing messages. I have been able to gather a core dump which shows this backtrace appended below. We are using Kamailio 5.6 retrieved from the kamailio repository: http://deb.kamailio.org/kamailio56. We are running Kamailio in a Docker container which runs on "5.10.0-25-cloud-amd64 #1 SMP Debian 5.10.191-1 (2023-08-16) x86_64 GNU/Linux" We have previously tried to use Kamailio 5.7 but it gave the same type of crashes. We have been using the uac module a lot but it is just in this scenario we get a core dump. We have some kind of relation to a specific scenario but it can take from several seconds from the last SIP message of this scenario up to 50 minutes until the crash occurs. To me this sounds like some kind of cleanup that is not handled properly. The back trace indicates that free of shared memory could be the issue, but I don't know the code unfortunately. The last things we see in the log file is: 2024-01-23T13:18:08.828+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4789]: INFO: <script>: Incoming SIP TCP request conid 21 call-id un0rihsRLJLvP-grn6LO-A 2024-01-23T13:18:08.835+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4789]: INFO: <script>: Incoming SIP TCP request conid 21 call-id WQjhldpRJbxjZRYe7fWgbw 2024-01-23T13:18:08.860+01:00 Jan 23 12:18:08 /usr/sbin/kamailio[4795]: CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 49 2024-01-23T13:18:09.486+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: ALERT: <core> [main.c:783]: handle_sigs(): child process 4777 exited by a signal 11 2024-01-23T13:18:09.486+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: ALERT: <core> [main.c:787]: handle_sigs(): core was generated 2024-01-23T13:18:09.516+01:00 Jan 23 12:18:09 /usr/sbin/kamailio[4751]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 2024-01-23T13:18:09.570+01:00 Started /root/sipconfig/startkamailio.sh 2024-01-23T13:18:09.570+01:00 info: :-) Starting Kamailio Just before all the crashes we see the "CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 49" log line. Best regards, Mattis Lind # gdb /usr/sbin/kamailio /core *GNU gdb (Debian 10.1-1.7) 10.1.90.20210103-git* Copyright (C) 2021 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: <https://www.gnu.org/software/gdb/bugs/>. Find the GDB manual and other documentation resources online at: <http://www.gnu.org/software/gdb/documentation/>. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from /usr/sbin/kamailio... (No debugging symbols found in /usr/sbin/kamailio) warning: Can't open file /dev/zero (deleted) during file-backed mapping note processing [New LWP 4777] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/sbin/kamailio -DD -M 18 -m 192 -A serverId=17173 -A sendTraceLocal="sip:10'. Program terminated with signal SIGSEGV, Segmentation fault. #0 __strlen_evex () at ../sysdeps/x86_64/multiarch/strlen-evex.S:77 77 ../sysdeps/x86_64/multiarch/strlen-evex.S: No such file or directory. (gdb) bt full #0 __strlen_evex () at ../sysdeps/x86_64/multiarch/strlen-evex.S:77 No locals. #1 0x00007f5425e34b78 in __vfprintf_internal (s=s@entry=0x55a06e969a60, format=format@entry=0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n", ap=ap@entry=0x7ffd6bebfb50, mode_flags=mode_flags@entry=0) at vfprintf-internal.c:1647 len = <optimized out> step0_jumps = {0, 1717, 1621, 3413, 3317, 3997, 2677, 2837, 3613, 1773, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 4101} space = <optimized out> is_short = <optimized out> use_outdigits = 0 outc = <optimized out> step1_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 1773, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 0} group = 0 prec = -1 step2_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4309, 4445, 3517, 4437, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 533, 533, 0} string = 0x756d6f7266222c22 <error: Cannot access memory at address 0x756d6f7266222c22> left = 0 is_long_double = <optimized out> width = 0 signed_number = <optimized out> step3a_jumps = {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 4213, 0, 0, 0, 4389, 2789, 4197, 3917, 3221, 0, 0, 0, 0, 1925, 0, 0, 0, 0, 0, 0} alt = <optimized out> showsign = 0 is_long = 0 is_char = <optimized out> pad = <optimized out> step3b_jumps = {0 <repeats 11 times>, 3517, 0, 0, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 0, 0, 0} step4_jumps = {0 <repeats 14 times>, 4389, 2789, 4197, 3917, 3221, 2997, 1141, 1365, 1997, 1925, 1885, 733, 3709, 0, 0, 0} args_value = <optimized out> is_negative = <optimized out> number = {longlong = <optimized out>, word = <optimized out>} base = <optimized out> the_arg = {pa_wchar = 4777 L'\x12a9', pa_int = 4777, pa_long_int = 4777, pa_long_long_int = 4777, pa_u_int = 4777, pa_u_long_int = 4777, pa_u_long_long_int = 4777, pa_double = 2.3601515901836347e-320, pa_long_double = 1.74131181638025811763e-4947, pa_float128 = 3.09319115455554459548860449034534676e-4962, pa_string = 0x12a9 <error: Cannot access memory at address 0x12a9>, pa_wstring = 0x12a9 <error: Cannot access memory at address 0x12a9>, pa_pointer = 0x12a9, pa_user = 0x12a9} spec = 115 's' _buffer = {__routine = 0x4, __arg = 0xd, __canceltype = 1855363680, __prev = 0xe0} _avail = <optimized out> thousands_sep = 0x0 grouping = 0xffffffffffffffff <error: Cannot access memory at address 0xffffffffffffffff> done = 238 f = 0x55a06e5580a7 "s:%lu. Exec from %s:%u.\n" lead_str_end = 0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n" end_of_spec = <optimized out> work_buffer = "h\r\000\000\000\000\000\000\000\000\000\000\060\000\000\000\000\000\000\000\375\177\000\000 \372\353k\375\177\000\000\237MSn\n\000\000\000\000\000\000\000\240U", '\000' <repeats 18 times>, "P\225Un\240U\000\000\000\000\000\000[\214A\323\f\000\000\000\000\000\000\000\377\377\377\377\377\377\377\377\206\002", '\000' <repeats 14 times>, "\004\000\000\000\000\000\000\000 \367\353k\375\177\000\000\301\225Un\240U\000\000\323\262\vn\240U\000\000\000\000\000\000\000\000\000\000s\374On\240U\000\000W\225Un\240U\000\000\350>\212\027T\177\000\000 \323\365$T\177\000\000\060\a\354k\375\177\000\00--Type <RET> for more, q to quit, c to continue without paging-- 0\220\371\227n\240U\000\000"... workend = 0x7ffd6bebf9f8 "" ap_save = {{gp_offset = 16, fp_offset = 48, overflow_arg_area = 0x7ffd6bebfc30, reg_save_area = 0x7ffd6bebfb70}} nspecs_done = 10 save_errno = 4 readonly_format = 0 do_longlong_number = <optimized out> __result = <optimized out> #2 0x00007f5425ec079f in __vsyslog_internal (pri=<optimized out>, fmt=0x55a06e558020 "%s: %.*s%s%s%sBUG: qm: fragm. %p (address %p) beginning overwritten (%lx)! Memory allocator was called from %s:%u. Fragment marked by %s:%lu. Exec from %s:%u.\n", ap=0x7ffd6bebfb50, mode_flags=0) at ../misc/syslog.c:233 now_tm = {tm_sec = 8, tm_min = 18, tm_hour = 12, tm_mday = 23, tm_mon = 0, tm_year = 124, tm_wday = 2, tm_yday = 22, tm_isdst = 0, tm_gmtoff = 0, tm_zone = 0x55a06e94c5e0 "UTC"} now = 1706012288 fd = <optimized out> f = 0x55a06e969a60 buf = 0x0 bufsize = 0 msgoff = 21 saved_errno = <optimized out> failbuf = "`\232\226n\240U\000\000\000\204\201\247[\214A\323`\374\353k\375\177\000\000\300p\371%T" clarg = {buf = <optimized out>, oldaction = <optimized out>} #3 0x00007f5425ec0c46 in __syslog (pri=<optimized out>, fmt=<optimized out>) at ../misc/syslog.c:117 ap = {{gp_offset = 48, fp_offset = 48, overflow_arg_area = 0x7ffd6bebfc70, reg_save_area = 0x7ffd6bebfb70}} #4 0x000055a06e3b7839 in ?? () No symbol table info available. #5 0x000055a06e3bc039 in qm_free () No symbol table info available. #6 0x000055a06e3c7c28 in qm_shm_free () No symbol table info available. #7 0x00007f542325fb8e in uac_send_tm_callback () from /usr/lib/x86_64-linux-gnu/kamailio/modules/uac.so No symbol table info available. #8 0x00007f5424a2f002 in run_trans_callbacks_internal () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #9 0x00007f5424a2f179 in run_trans_callbacks () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #10 0x00007f54249d5e8c in free_cell_helper () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #11 0x00007f5424aa8f82 in wait_handler () from /usr/lib/x86_64-linux-gnu/kamailio/modules/tm.so No symbol table info available. #12 0x000055a06e37a263 in ?? () No symbol table info available. #13 0x000055a06e37a79d in ?? () No symbol table info available. #14 0x000055a06e37acc6 in timer_main () No symbol table info available. #15 0x000055a06e0a5f62 in main_loop () No symbol table info available. --Type <RET> for more, q to quit, c to continue without paging-- #16 0x000055a06e0b120c in main () No symbol table info available.

2 3

How to achieve conversion from 180 to 183sdp
by mm e 29 Jan '24

29 Jan '24

When receiving to-tag and returning 180 ring without SDP body, how can it be converted into 183 with SDP and returned to from-tag. I want to realize that when I receive 180 no sdp, I will play an audio clip for the from-tag.： INVITE (SDP) ──────────────────────────> │ 100 Trying <────────────────────────── │ │ INVITE (SDP) │ ──────────────────────────> 100 Trying │ <────────────────────────── 180 Ringing │ <────────────────────────── 183 Session Progress (SDP) │ <────────────────────────── │

2 1

usrloc database duplicate keys issue
by Benoît Panizzon 29 Jan '24

29 Jan '24

Hi I have come across an issue with the usrloc database on kamailio 5.6 and 5.7 Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: <core> [db_query.c:244]: db_do_insert_cmd(): error while submitting query Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: usrloc [ucontact.c:686]: db_insert_ucontact(): inserting contact in db failed somedudetp0216 (uloc-65b237f8-28983d-1) Jan 26 07:19:45 dev-cpereg01 kamailio[2708456]: ERROR: usrloc [urecord.c:419]: wb_timer(): inserting contact into database failed (aor: somedudetp0216) settings: # ----- usrloc params ----- modparam("usrloc", "db_url", DBLOCAL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "ka_mode", 0) modparam("usrloc", "ka_filter", 1) modparam("usrloc", "timer_procs", 1) modparam("usrloc", "ka_from", "sip:ping@sip.imp.ch") modparam("usrloc", "server_id_filter", 1) # Use DMQ to sync userloc loadmodule "dmq_usrloc.so" # Replicate usrloc via dmq # Enable userloc sync modparam("dmq_usrloc", "enable", 1) # Do a full sync on start modparam("dmq_usrloc", "sync", 1) I assume, when kamailio is being restartet there are some expired entries still present in the database which kamailio tries to insert on a new registration. This leads to this duplicate key error. Is there a way to tell kamailio to use 'replace into' instread of 'insert into'? Solution for now, is to manually delete expired entries from the database. -- 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 ______________________________________________________

3 6

Heads-up: testing needed for major OpenSSL refactoring
by Richard Chan 28 Jan '24

28 Jan '24

Hello Kamailio community, After the January refactoring of OpenSSL integration, configurations that initialise libssl in rank 0 (thread #1) are likely to trigger errors more aggressively. This type of configuration has all along been broken due to OpenSSL use of thread-local variables. The first "victim" is https://github.com/kamailio/kamailio/issues/3727 — the configuration has dispatcher + db_unixodbc (+ some driver) — this configuration initialises libssl in rank 0(thread #1). I have a couple of PRs in the queue for db_unixodbc, db_mysql (PostgreSQL may need similar attention). master and 5.7 branches are affected Regards Richard

1 0

LCR logic
by Social Boh 27 Jan '24

27 Jan '24

Hello, LCR module logic is : When the function /load_gws()/ is called, matching gateways (that are not currently designated as defunct) are ordered for forwarding purposes as follows: 1. according to longest Request-URI user part match 2. according to tuple's priority 3. according to tuple's randomized weight or, if priority_ordering parameter is set to value 1, as follows: 1. according to tuple's priority 2. according to tuple's randomized weight and Smaller priority value means higher priority (highest priority value being 0 and lowest being 255). so I have 3 lcr_rule_target use the same lcr_rule entry but 3 different Gateways and priority. So the order of rules would be longest Request-URI (same) and the priority but the result is: added matched_gws[0]=[3, 5, 200, 2193562] added matched_gws[1]=[1, 5, 2, 6888519] added matched_gws[2]=[2, 5, 50, 2159388] same prefix but first rule target with priority 200 and then 2 and then 50. I think the result would be priority 2,50 and 100. ¿Or not? --- I'm SoCIaL, MayBe

2 2

DMQ dialog sync question
by Stefan-Cristian Mititelu 26 Jan '24

26 Jan '24

Hi, I'm trying a DMQ dialog sync up setup with 2 x Kamailios. I have this scenario: 1. Kamailio1 receives INVITE and sends KDQM to Kamailio2 to sync up dialog => OK 2. Kamailio2 receives 200OK 3. Kamailio2 does not send anything back to Kamailio1 to update dialog state => not OK, i expected a KDMQ to be send back to Kamailio1 to update dialog state I was wondering if DMQ module supports the above? I am using 5.6 version of kamailio, and I am curios if this might be implemented in later branches? Thank you, Stefan

2 3

How to correctly drop a branch?
by Benoît Panizzon 26 Jan '24

26 Jan '24

Hi https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#branch_name_-… "Assigning $null to uri attribute will drop the branch, for the rest of attributes will just set the value to null." Where do I need to assing $null to the brach URI to get it dropped? And what happens to the index, is there a gap, or do the subsequent branches get re-indexed so they don't match with the ulc indexes anymore? Actual example, I have multiple contacts I would like to add to branches. Two of the contacts are from the location database + an additional contact. # Basic Branch: $ru = $(ulc(aor=>addr)[0]); $du = $(ulc(aor=>received)[0]); $fs = $(ulc(aor=>socket)[0]); # Additional Branch: append_branch($(ulc(aor=>addr)[1])); $(branch(dst_uri)[-1]) = $(ulc(aor=>received)[1]); $(branch(send_socket)[-1]) = $(ulc(aor=>socket)[1]); and so on, actually a loop over all ulc contacts. # Lasst Additional Branch, point to other registrar, not to a usrloc location. append_branch($ru); $(branch(dst_uri)[-1]) = "sip:other-registrar:5060;transport=udp"; $fsn = "LocalUDPv4"; $(branch(send_socket)[-1]) = $fs; t_on_branch("BR_T"); branch_route[BR_T] { $var(socket) = $(ulc(aor=>socket)[$T_branch_idx]); if ($var(socket) == 0) { xlog("L_INFO", "$cfg(route): DROPPING BRANCH: $T_branch_idx No local socket on this registrar\n"); $(branch(uri)[$T_branch_idx]) = $null; } } I don't get this right! Call is still sent to that contact. -- 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 ______________________________________________________

1 2

Change in LCR Debug
by Social Boh 26 Jan '24

26 Jan '24

Hello, using DEBUG with LCR module to see how the module apply the rules, I see he use Gateway Index in the list. The Gateway Index could be different to Gateway ID and maybe can confuse the correct interpretation about the use of the rules. Is it possible change the code: matched_gws[gw_index].gw_index = t->gw_index; matched_gws[gw_index].rule_id = rule->rule_id; matched_gws[gw_index].prefix_len = pl->prefix_len; matched_gws[gw_index].priority = t->priority; matched_gws[gw_index].weight = t->weight * (kam_rand() >> 8); matched_gws[gw_index].duplicate = 0; LM_DBG("added matched_gws[%d]=[%u %u, %u, %u, %u]\n", gw_index, t->gw_index, pl->prefix_len, t->priority, matched_gws[gw_index].weight); to use gateway ID and not gateway Index? Thank you Regards -- --- I'm SoCIaL, MayBe

1 0

How to get media IP
by mm e 26 Jan '24

26 Jan '24

When receiving a call I get the following message： INVITE sip:123456789@192.168.1.118:29363 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport Max-Forwards: 70 Contact: <sip:96341@192.168.1.118:65178> To: "123456789"<sip:123456789@192.168.1.119:29363 > From: "96341"<sip:96341@192.168.1.119:29363 >;tag=af3c7c31 Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011d stamp 40820 Content-Length: 339 v=0 o=- 3 2 IN IP4 192.168.1.118 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.118 t=0 0 m=audio 32170 RTP/AVP 0 8 18 101 a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv How to print the media IP in SDP separately in the log? I didn't find the Pseudo-Variables..can you help me

3 2

412 Conditional request failed
by satyaprakash ch 26 Jan '24

26 Jan '24

Hi There were two PUBLISH requests were sent by the Kamailio Proxy server to the Kamailio Presence server. Processing of the first PUBLISH request resulted in the presentity table being updated - the "etag" changed from "a.1705053846.16611.21.1" to "a.1705053846.16613.23.2" Processing of the second PUBLISH failed because it also attempted to update the etag in the presentity table but at this point in time there was no match to the original etag value in the database Logs in kamailio.log: ------ ps_db_update_presentity(): No E-Tag match a.1705053846.16611.21.1 DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out - buf=0x7f81e5ca5fc0: SIP/2.0 412 Conditio... shmem=0x7f81de6da068: SIP/2.0 412 Conditional request failed. Will any one suggest how to resolve the issue of 412 conditional request failed.

2 2

Is the Contact header required for the dialog module?
by Unai Rodriguez 26 Jan '24

26 Jan '24

Dear List, I’m trying to set up the dialog module and I keep getting the error "bad sip message or missing Contact hdr”: --- Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing Contact hdr Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the dialog --- This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. They look like this: ---- INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0 Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ Max-Forwards: 70 From: <sip:10.10.10.10:5099>;tag=134a3eKD2a8Ua To: <sip:some.mrcp.server.example.com:8060> Call-ID: 051df0ff-3541-123d-f090-02f373ff014c CSeq: 78491159 INVITE User-Agent: google_stt Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 329 v=0 o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10 s=- c=IN IP4 10.10.10.10 t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:new a=resource:speechrecog a=cmid:1 m=audio 12034 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 L16/8000 a=sendonly a=mid:1 ---- According to the RFC, the Contact header is not mandatory (https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have an easy way of adding the Contact header to the INVITEs. Is there a way to configure Kamailio’s dialog module to do without the Contact header? Any pointers appreciated. Thank you so much With best wishes, Unai Rodriguez

3 5

How to assign xavp array to htable?
by Benoit Panizzon 26 Jan '24

26 Jan '24

Hi List Via sql_xquery I get a stacked xavp aka array. I would like to store that stacked xavp into a hash table for later usage in other transactions. Is this possible? Via xarp_parameters explode or implode does not work as that would stringy the same key multiple times. 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 3

How to set a branch specific variable?
by Benoit Panizzon 25 Jan '24

25 Jan '24

Hi Gang I would like to set a variable specific to the branch. Is there a way? append_branch($var(aor1)); $(branch(dst_uri)[-1]) = $var($dst1) $(branch(want_plus)[-1]) = 1; append_branch($var(aor2)); $(branch(dst_uri)[-1]) = $var($dst2) $(branch(want_plus)[-1]) = 0; t_on_branch("BR_T"); branch_route[BR_T] { if ($branch(want_plus)) { $rU = "+" + $rU; } } Or would there be a way to query the index of the added branch so I could stack an avp with that index to access in the branch route? 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 ______________________________________________________

1 0

Re: STIR/SHAKEN
by Anthony Wittig 25 Jan '24

25 Jan '24

Hey everyone, I've set it up so that when the invite comes into Kamailio I suspend the transaction (save the transaction info in sht) and use dispatcher to send a request to our NSS server. On the reply from NSS I throw the identity header into a sht and continue the original transaction. Does that seem like the right way to do it? The next things I need to look at are: * if first NSS server doesn't respond quickly, try the second (change some timeouts and use dispatcher to try the backup) * if both NSS servers aren't working, I still want to resume the original call and let it through. Maybe I can keep track of the failures in a sht and if I have two failures, resume the original invite without the identity header. Sound like an ok plan? -- Anthony Wittig Mango Voice Developer > > ---------- Forwarded message ---------- > From: Anthony Wittig <awittig(a)mangovoice.com> > To: sr-users(a)lists.kamailio.org, david.villasmil.work(a)gmail.com > Cc: > Bcc: > Date: Tue, 23 Jan 2024 20:39:09 -0700 > Subject: [SR-Users] STIR/SHAKEN > Thanks! > > This is what I've got so far: > > 1. Kamailio invite -> NSS > 2. NSS 302 -> Kamailio > 3. Kamailio ack -> NSS > > I can see the identity header on the 302 response in my onreply_route. How > can I replay the same invite from step #1 with the identity header I get > back in step #2? Should I be doing something with a transaction on step #1, > maybe start a branch that goes out to NSS and when it comes back resume the > transaction (these are all words I've heard smart people say, but I don't > fully understand them...). > > Any pointers would be greatly appreciated, thanks! > -- > Anthony Wittig > Mango Voice Developer > > ---------- Forwarded message ---------- > From: David Villasmil <david.villasmil.work(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Tue, 23 Jan 2024 14:56:52 +0100 > Subject: [SR-Users] Re: STIR/SHAKEN > Ditto, i do first approach. > > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > > > On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < > sr-users(a)lists.kamailio.org> wrote: > >> These are all valid approaches, depending on preference. Catching 302s >> and extracting exactly the desired info might be easier with Kamailio, so I >> suppose, ceteris paribus, I'd recommend the first one. >> >> Dispatcher is a great approach! >> >> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < >> sr-users(a)lists.kamailio.org> wrote: >> > >> > Hello, >> > >> > We're attempting to add identities to our invites. I believe we want to >> send our invites to the NSS, and it'll reply with a 302'd invite with an >> identity header. We currently use FreeSWITCH for dialplans and media and >> Kamailio as our SBC. >> > >> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS >> (302) -> Kamailio -> PSTN >> > >> > or would it make more sense to do: FreeSWITCH -> NSS (302) -> >> FreeSWITCH -> Kamailio -> PSTN >> > >> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right >> approach? We currently use it for our two Kamailios and our FreeSWITCHs. >> We're going to have two NSS servers, one for failover. >> > >> > Any thoughts would be greatly appreciated, thanks! >> > >> > -- >> > Anthony Wittig >> > Mango Voice Developer >> > __________________________________________________________ >> > Kamailio - Users Mailing List - Non Commercial Discussions >> > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org >> > Important: keep the mailing list in the recipients, do not reply only >> to the sender! >> > Edit mailing list options or unsubscribe: >> >> -- >> Alex Balashov >> Principal Consultant >> Evariste Systems LLC >> Web: https://evaristesys.com >> Tel: +1-706-510-6800 >> >

1 0

STIR/SHAKEN
by Anthony Wittig 24 Jan '24

24 Jan '24

Thanks! This is what I've got so far: 1. Kamailio invite -> NSS 2. NSS 302 -> Kamailio 3. Kamailio ack -> NSS I can see the identity header on the 302 response in my onreply_route. How can I replay the same invite from step #1 with the identity header I get back in step #2? Should I be doing something with a transaction on step #1, maybe start a branch that goes out to NSS and when it comes back resume the transaction (these are all words I've heard smart people say, but I don't fully understand them...). Any pointers would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer ---------- Forwarded message ---------- From: David Villasmil <david.villasmil.work(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: Bcc: Date: Tue, 23 Jan 2024 14:56:52 +0100 Subject: [SR-Users] Re: STIR/SHAKEN Ditto, i do first approach. Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users < sr-users(a)lists.kamailio.org> wrote: > These are all valid approaches, depending on preference. Catching 302s and > extracting exactly the desired info might be easier with Kamailio, so I > suppose, ceteris paribus, I'd recommend the first one. > > Dispatcher is a great approach! > > > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users < > sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > > > We're attempting to add identities to our invites. I believe we want to > send our invites to the NSS, and it'll reply with a 302'd invite with an > identity header. We currently use FreeSWITCH for dialplans and media and > Kamailio as our SBC. > > > > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS > (302) -> Kamailio -> PSTN > > > > or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH > -> Kamailio -> PSTN > > > > Assuming we wanted Kamailio -> NSS, is using dispatcher the right > approach? We currently use it for our two Kamailios and our FreeSWITCHs. > We're going to have two NSS servers, one for failover. > > > > Any thoughts would be greatly appreciated, thanks! > > > > -- > > Anthony Wittig > > Mango Voice Developer > > __________________________________________________________ > > Kamailio - Users Mailing List - Non Commercial Discussions > > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > > Important: keep the mailing list in the recipients, do not reply only to > the sender! > > Edit mailing list options or unsubscribe: > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: >

1 0

STIR/SHAKEN
by Anthony Wittig 23 Jan '24

23 Jan '24

Hello, We're attempting to add identities to our invites. I believe we want to send our invites to the NSS, and it'll reply with a 302'd invite with an identity header. We currently use FreeSWITCH for dialplans and media and Kamailio as our SBC. Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS (302) -> Kamailio -> PSTN or would it make more sense to do: FreeSWITCH -> NSS (302) -> FreeSWITCH -> Kamailio -> PSTN Assuming we wanted Kamailio -> NSS, is using dispatcher the right approach? We currently use it for our two Kamailios and our FreeSWITCHs. We're going to have two NSS servers, one for failover. Any thoughts would be greatly appreciated, thanks! -- Anthony Wittig Mango Voice Developer

3 3

How to check for branches with a pending reply in MANAGE_REPLY to correctly handle rtpengine?
by Benoit Panizzon 23 Jan '24

23 Jan '24

Hi gang location database contains multiple contacts for an AOR The invite therefore is parallel branched to those multiple contacts. Each reply from each branch is handled by MANAGE_REPLY. rtpengine_manage() is called in MANAGE_REPLY to stop rtpengine processing a failed call. Unfortunately, if any of the contacts replies with an error rtpengine_manage() will stop RTP processing. Is there a way I could check if there are branches with a pending reply from within MANAGE_REPLY so I could call rtpengine_manage() only when the last reply (no pending replies) are present? Hmm, realizing, if I process the last pending error reply and a prior one got a positive reply, this would still kill media. => How do I solve this? 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 ______________________________________________________

1 0

Info: Kamailio World 2024 - Call for presentations
by Daniel-Constantin Mierla 22 Jan '24

22 Jan '24

Hello, a short note to inform that the Call For Presentations is now open for Kamailio World 2024. Everyone is welcome to submit proposal for presentations to share the knowledge about Kamailio or Real Time Communication services, security, high availability, scalability, etc. Submission form and more details are available at: - https://www.kamailioworld.com/k2024/call-for-speakers/ 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

2 2

IMS registration failure
by srinidhikrs＠yahoo.com 21 Jan '24

21 Jan '24

I am trying to get imsdroid register to pcscf . But failing with 504 server timeout . Please help to figure out the reason for register failure . <is_registered_fallback2ip> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/modparam.c:123]: set_mod_param_regex(): found <is_registered_fallback2ip> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <pcscf_unregister> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg.lex:1643]: sr_push_yy_state(): including file: route/register.cfg (/etc/kamailio_pcscf/route/register.cfg) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <pcscf_is_registered> in module ims_registrar_pcscf [/usr/local/lib64/kamailio/modules/ims_registrar_pcscf.so] [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'core' (num=56, size=224) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'tcp' (num=27, size=108) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 0 (old) + 1 (new) = 1 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'tm' (num=36, size=168) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'maxfwd' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'siputils' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 1 (old) + 1 (new) = 2 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'xlog' (num=1, size=4) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'dispatcher' (num=3, size=24) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'sctp' (num=19, size=76) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 2 (old) + 1 (new) = 3 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'dbg' (num=3, size=12) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'registrar' (num=11, size=56) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/cfg/cfg.c:173]: cfg_declare(): new config group has been registered: 'rtpengine' (num=5, size=20) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 3 (old) + 1 (new) = 4 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 4 (old) + 1 (new) = 5 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:138]: register_procs(): registering new processes: 5 (old) + 1 (new) = 6 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/pt.c:99]: init_pt(): registering new processes: 6 (old) + 96 (new) = 102 (total) [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/route.c:885]: fix_actions(): fixing pcscf_is_registered() [0;39;49m[0;39;49m 0(310120) DEBUG: <core> [core/route.c:885]: fix_actions(): fixing pcscf_unregister() [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=9 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=7 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=9 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=10 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=11 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=12 a=25 n=xnotice [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=15 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=22 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=17 a=25 n=is_present_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=18 a=25 n=pcscf_save_pending [0;39;49m[0;39;49m 4(310124) DEBUG: ims_registrar_pcscf [save.c:378]: save_pending(): registering for UL callback [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=25 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=32 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=35 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=33 a=63 n=assign [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=81 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=83 a=25 n=remove_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=84 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=85 a=25 n=remove_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=86 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=94 a=16 n=if [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=89 a=25 n=is_present_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=93 a=25 n=append_hf [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=107 a=25 n=t_on_failure [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=109 a=25 n=t_on_reply [0;39;49m[0;31;49m 4(310124) ERROR: *** cfgtrace:request_route=[REGISTER] c=[/etc/kamailio_pcscf/route/register.cfg] l=111 a=5 n=route [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=316 a=16 n=if [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=311 a=25 n=t_check_status [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=312 a=26 n=send_reply [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=313 a=26 n=update_stat [0;39;49m[0;31;49m85(310272) ERROR: *** cfgtrace:failure_route=[REGISTER_failure] c=[/etc/kamailio_pcscf/route/register.cfg] l=314 a=2 n=exit

1 2

failed to open database connection
by mtohidi67＠gmail.com 20 Jan '24

20 Jan '24

Hello. I have installed Kamailio version devel on Ubuntu 22.04 and use the mysql database on the same machine. when i run systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2024-01-20 11:35:09 +0330; 21min ago Docs: man:systemd-sysv-generator(8) CPU: 48ms /usr/local/sbin/kamailio[173481]: ERROR: usrloc [dlist.c:846]: register_udomain(): failed to open database connection /usr/local/sbin/kamailio[173481]: ERROR: registrar [registrar.c:767]: domain_fixup(): failed to register domain /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:717 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:720 /usr/local/sbin/kamailio[173481]: INFO: [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized kamailio[173463]: * already running kamailio[173463]: ...done. systemd[1]: Started LSB: Start the Kamailio SIP proxy server. In /usr/local/etc/kamailio/kamctlrc, I uncomment these lines: SIP_DOMAIN=ims.mnc55.mcc999.farazNET.org DBENGINE=MYSQL DBHOST=localhost in /usr/local/etc/kamailio/kamailio.cfg i apply this changes: auto_aliases=no alias="ims.mnc55.mcc999.farazNET.org" listen=udp:192.168.75.185:5060 listen=tcp:192.168.75.185:5060 how can i solved it?

1 1

rr, loose_route and corex alias_subdomains
by Jon Bonilla (Manwe) 19 Jan '24

19 Jan '24

Hi I'm having an issue with the double rr for mst direct routing. I'm using subdomains like 23234.mydomain.com with a wildcard certificate and I set corex modparam alias_subdomains to mydomain.com But loose_route doesn't seem to detect the rr as myself I get the message: rr [loose.c:804]: rr_do_force_send_socket(): no socket found to match second RR (sip:20141.mydomain.com:5063;transport=tls;ftag=as7107d0fe;lr;r2=on;vsf=AAAAAAAAAAADCQMaABhMS1NYRh0cRllbR0RRQG91cC5lcw--;nat=yes;pet=mst) Regarding my setup is more or less like this: kamailio.local.cfg: #!substdef "!MSTDOMAIN!mydomain.com!g" kamailio.cfg: modparam("corex", "alias_subdomains", "MSTDOMAIN") But in the cli: kamctl rpc corex.list_aliases { "jsonrpc": "2.0", "result": [ ], "id": 1319303 } I don't remember to have this issue when I was developing the mst integration. Either I missed it or can it be a regression? kamailio version is 5.6.4 at the moment. Any ideas? cheers Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 3

Just need some basic help
by Jere Cassidy 18 Jan '24

18 Jan '24

Hello! Thanks in advance for any help you can provide. I have a simple setup here and am struggling a bit with getting a final working solution. This is just a single direction flow from a SIP provider inbound to servers running HMPelements with CTI integration (sip endpoint). Essentially a different DN / dial number would go to a different backend server. There is no registration involved. The kamailio server therefore needs to sit between the SIP provider and the backend servers, it will be a NAT situation where private RFC1918 addresses will be used on the backend after the server receives it with public IP information. So that's it. What is the simplest way to accomplish this? If it would help - here's what ive done so far: For the Kamailio server, I have an AWS EC2 instance with a public IP bound/assigned to the main private IP. During troubleshooting, I've added another private IP to the NIC so that I could throw up another listener. I was thinking I'd need to accept the SIP on the first listener with the "advertise" directive. The second listener would not have an advertise directive so perhaps could be used to talk to the backends. I defined the WITH NAT and rtp proxy settings (pointing to the first listener below). The RTP Proxy daemon also knows about the y.y.y.y public address below. In the kamailio.cfg, I have a listener setup : listen=udp:172.24.40.28:5060 advertise y.y.y.y:5060 listen=udp:172.24.40.26:5060 then threw in a route(CMS) in the request route and created the following route block: route[CMS] { #check URI matches phone xxxxxxxxxx if (uri=~"^sip:\+1xxxxxxxxxx@") { rewritehostport("10.12.18.193:5060"); forward(uri:host, uri:port); exit; }; } I get the SIP delivered to the backend host! but the INVITE headers are stil pointing to public IPs and the SDP info is not adjusted to be a private IP as well - as follows: z.z.z.z = sip provider , y.y.y.y = public ip (aws ec2 ip). xxxxxxxxx = calling phone, ######## = called phone (matches routeCMS above) INVITE sip:+1##########@10.12.18.193:5060 SIP/2.0 Record-Route: <sip:y.y.y.y;lr> Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK28b7.56b42fdeec9f10cd13030b3711b16f4d.0 Via: SIP/2.0/UDP z.z.z.z:11000;received=z.z.z.z;rport=11000;branch=z9hG4bKQ2yQKgv7NtKQF Max-Forwards: 48 From: "" <sip:+1xxxxxxxxxx@z.z.z.z;isup-oli=0>;tag=tUF15gaNg6N6j To: <sip:+1##########@y.y.y.y> Call-ID: 9ee7e073-c7fc-46ac-a839-f3c77eaed903 CSeq: 78244364 INVITE Contact: <sip:mod_sofia@z.z.z.z:11000> User-Agent: 2600Hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 278 X-FS-Core-UUID: 4e0b2928-d913-42c4-9809-f77a36d281be X-FS-Support: update_display,send_info P-Asserted-Identity: "" <sip:+1xxxxxxxxx@z.z.z.z> v=0 o=FreeSWITCH 1705572686 1705572687 IN IP4 z.z.z.z Thanks much!

2 1

Re: add a SIP custom header on a SIP response generated locally
by frédéric Gaisnon 18 Jan '24

18 Jan '24

Hi, Yes I've tried using append_to_reply() in several routes (tm:branch-failure, sl:local-response, tm:local-response, onreply_route, failure_route) but it was not applied. It works fine if a response is received or if it is generated by the configuration script but not if this response is locally generated by a kamailio module. Regards, Frédéric Gaisnon > Custom headers to the reply can be added using append_to_reply() function from textops module. Have you tried that? > > Regards, > > Fred Posner > p: +1 (352) 664-3733 > > On Jan 18, 2024, at 2:07 AM, frédéric Gaisnon via sr-users < sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) > > I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. > > Do you have any suggestions to accomplish this ? > > Regards, > > Frédéric Gaisnon

2 1

trigger different branch route on branch added with append_branch
by Benoit Panizzon 18 Jan '24

18 Jan '24

Hi List Imagine the following situation: Two Kamailio registrar proxies handling the same domain. Location Information synced via DMQ. So if a device registers we don't know on which of the two registrar proxies. Core proxy is dispatching calls to either one of those registrars. Core also performs AOR database lookup and adds an X- header telling which AOR the call is destined to. Registrar performs lookup of that AOR in the location database and routes the call to destination. Problem: The call could be routed to the the registrar which does not hold the active socket to that client. Solution: reg_fetch_contacts, cycle the contacts and append_branch if the contact has a local socket. Next Problem: I use a failure_route to sent the call to voicemail, if not registered, busy etc. This would trigger, if we have no local socket for the registration which is on the other registrar. So I have to find a way to send the call to the other registrar. I guess I can append_branch a branch pointing to the URI of the other registrar. This would then succeed in reaching the device. Also using the failure_route would work this way if we get a remote error from behind the other registrar or no registration at all. Next Problem: I use a branch trigger to remove undesired header I do not want to send to the customer. Now imagine that AOR has two contacts, one on registered on each registrar proxy. So I end up with two destinations added with append_branch to the destination set. One points to the device of the locally registered customer device, the other one points to the other registrar where to find the other device. I use a branch route trigger to remove the X- Headers towards the customer. But I need to X- Header containing the destination AOR to still be sent to the other registrar. Is there a way to trigger a branch route for only some of the destinations added with append_branch? Or to trigger different branch routes for different destinations in one set added with append_branch? Or am I doing this completely wrong and there is an easier way? ;-) 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 2

How to change uri for db_mysql ?
by iglov 18 Jan '24

18 Jan '24

Sup guyz! On debian 12 i have kamailio 5.6.3 and when I try to use db_mysql module I always get an error: kamailio[2842268]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local server through socket '/run/mysqld/mysqld.sock' I have no idea where it finds this socket cuz I don’t have it and don’t want to. Where I can change it?

4 7

add a SIP custom header on a SIP response generated locally
by frédéric Gaisnon 18 Jan '24

18 Jan '24

Hello, I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. Do you have any suggestions to accomplish this ? Regards, Frédéric Gaisnon

3 2

Kamailio v5.7.4 Released
by Daniel-Constantin Mierla 18 Jan '24

18 Jan '24

Hello, Kamailio SIP Server v5.7.4 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.3. 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 v5.7.4. For more details about version 5.7.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/01/kamailio-v5-7-4-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 Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 0

Releasing v5.7.4
by Daniel-Constantin Mierla 18 Jan '24

18 Jan '24

Hello, I am considering to release Kamailio v5.7.4 (out of branch 5.7) on Thursday, Jan 18, 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 Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

1 1

IMS Register failure
by srinidhi 2wewew 18 Jan '24

18 Jan '24

I am trying to get imsdroid register to pcscf . But failing with 504 server timeout . PCSCF IP : 192.168.1.28 imsdroid IP : 192.168.1.212 Please help to figure out the reaso for register failure . I have attached pcap file and pcscf log file

1 0

tls.cfg and configuration
by Aymeric Moizard 17 Jan '24

17 Jan '24

Hi all, I'm using a central file to define my kamailio cfg configuration. First question: In my main kamailio.cfg, I use: include_file "kamailio-config.cfg" which contains: #!substdef "!MY_DOMAIN!kamailio.example.com!g" I would like to use the same in tls.cfg which is defined this way: modparam("tls", "config", "/etc/kamailio/tls.cfg") but I get: sr_cfg_parse(): tls.cfg:17:1: Unsupported option 'include_file' Would you advise some way to achieve my goal: ie, I would like only one file to edit my configuration options which would have an effect on both kamailio.cfg and tls.cfg Second question: Can someone confirm that #!substdef will also not have any effect when used in tls.cfg Thanks Aymeric -- Antisip - http://www.antisip.com

2 1

2024 FOSDEM VoIP Dinner
by Torrey Searle 15 Jan '24

15 Jan '24

Hello all, It is that time of year to start getting the headcount for the FOSDEM voip dinner on the Saturday of FOSDEM. Usually Daniel would get the headcount for me and I just make the booking. However Daniel can't make it this year so I'll also be taking the headcount If you want to come put your name up one the doodle I've set up for this https://doodle.com/meeting/participate/id/boAjDENe Hope to see you there! Torrey

1 0

Not checking SRV records
by Jon Bonilla (Manwe) 13 Jan '24

13 Jan '24

Hi This may be a stupid question but after a couple of hours and some headache I can't see where the problem is. I'm trying to use SRV records to relay to different servers as I've done many many times before. But in this case I'm getting a 478 from kamailio and I see that it's not trying to resolve SRV. From the log: [core/dns_cache.c:527]: _dns_hash_find(): (fs.mydomain.dev(16), 1), h=632 [core/resolve.c:784]: get_record(): lookup(fs.mydomain.dev, 1) failed [core/dns_cache.c:761]: dns_cache_mk_bad_entry(): (fs.mydomain.dev, 1, 60, 1) [core/dns_cache.c:701]: dns_cache_add(): adding fs.mydomain.dev(16) 1 (flags=1) at 632 [core/dns_cache.c:2807]: dns_a_resolve(): (fs.mydomain.dev, 0) returning -7 [core/dns_cache.c:3176]: dns_srv_sip_resolve(): (fs.mydomain.dev, 0, 0), ip, ret=-7 uri2dst2(): failed to resolve "fs.mydomain.dev" :unresolvable A or AAAA request (-7) When doing tcpdump port 53 I see A record query but no SRV SRV records of course exist although they are not being checked. dns params in kamailio: dns_try_ipv6=no dns_retr_time=1 dns_retr_no=1 dns_use_search_list=no use_dns_cache=on use_dns_failover=on dns_srv_lb=on dns_try_naptr=no dns_sctp_pref=1 dns_tcp_pref=1 dns_tls_pref=1 dns_udp_pref=1 Same exact config in another server is working. There's something I'm missing and can't see. Any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

1 1

Standard on RTP ports
by ZZ Wave 12 Jan '24

12 Jan '24

Hello. I have some rant with network security dept. Is there some standard, RFC, what port range should be used for RTP traffic, how RTP ports are selected during SIP call negotiation? Thanks.

11 25

Internal DNS resolver
by Igor Potjevlesch 12 Jan '24

12 Jan '24

Hello! I have a quick question about the internal DNS resolver and cache embedded in Kamailio. Is the ability to add an entry with dns.add_a bypass the "real" DNS lookup? So, let's say that sip.example.com is publicly resolve alternatively with 1.2.3.4 & 5.6.7.8.9, if I do a dns.add_a to add my own sip.example.com with, alternatively, 127.0.0.1 and 127.0.0.2, will it work? Thanks for your highlight! Regards, Igor. -- Cet e-mail a été vérifié par le logiciel antivirus d'Avast. www.avast.com

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xavp_params_explode broken comma in value?
by Benoît Panizzon 12 Jan '24

12 Jan '24

Hey! I'm hunting down an issue with xavp_params_explode, maybe somebody has already come adross it and could confirm it exists and maybe know a solution? xavp_params_explode("a=foo;c=hello,world;e=baar", "x"); $xavp(x=>a) is 'foo' but x=>c and x=>e are null Am I right to assume the issue being caused by the comma? Is there a way to allow a comma? -- 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 ______________________________________________________

3 4

registrar questions
by mm e 12 Jan '24

12 Jan '24

In addition to the RPC command "kamctl add username password", is there a module with a method to operate the "subscriber" database table?

2 4

Clarification on usrloc timer_interval param and contact-expires event route.
by James Lipski 12 Jan '24

12 Jan '24

Hello, Just need a bit of clarification on the usrloc parameter "timer_interval". Let's say this is configured with a value of 60 seconds; every 60 seconds, will this delete all expired contacts in the location table between runs -- as in between runs it will queue up a list of expired contacts, remove them all after the 60 seconds has elapsed, then execute the event route "event_route[usrloc:contact-expired]"? Or does the event route execute as soon as the contact expires, independent from "timer_interval" timer? Thank you.

2 1

Setting kamailio to use another DB
by SAMUEL MOYA TINOCO 11 Jan '24

11 Jan '24

Good evening everyone, I'm trying to configure my Kamailio to use another DB in case the one it commonly uses fails. Does Kamailio has any built in function to do this? Do I have to do it with an ha-proxy? Thank you in advance for your help Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DA431B.CC9B03A0] Soluciones inteligentes para la autonomía personal

4 4

How to know registration status
by mm e 10 Jan '24

10 Jan '24

After using the uac module to initiate registration to the remote end, is there a method to call back to notify the registration result? I want to get the registration result, such as 200 or 403

4 3

Re: force TLS RSA cipher suite (SOLVED)
by marek 10 Jan '24

10 Jan '24

good catch. thanks for the record ECC [root@sbc live]# openssl x509 -text -noout -in /etc/letsencrypt/live/<my domain>/fullchain.pem |grep -i "Public Key Algorithm" Public Key Algorithm: id-ecPublicKey certbot certonly --key-type rsa --cert-name <my domain> --manual --preferred-challenges dns RSA [root@sbc live]# openssl x509 -text -noout -in /etc/letsencrypt/live/<my domain>/fullchain.pem |grep -i "Public Key Algorithm" Public Key Algorithm: rsaEncryption but there was another problem modparam("tls", "cipher_list", "RSA") from https://kamailio.org/docs/modules/5.7.x/modules/tls.html#tls.p.cipher_list does nothing 0(2294) INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' valid option is in tls.cfg cipher_list = RSA 0(2328) INFO: tls [tls_domain.c:390]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='RSA' ServerHello Version 3.3 session_id[0]= cipherSuite TLS_RSA_WITH_AES_256_GCM_SHA384 Marek Cervenka Dne 2024-01-10 v 13:06 Lukas Tribus napsal(a): > > > On Wednesday 10 January 2024, marek via sr-users > <sr-users(a)lists.kamailio.org> wrote: > > hi, > > i'm trying force cipher list through options like > > modparam("tls", "cipher_list", "TLS_RSA_WITH_AES_256_CBC_SHA256") > > modparam("tls", "cipher_list", "RSA") > > > You are trying a RSA cipher. > > > ... > > > but > > ssldump -i enp2s0 port 5061 shows every time > > ServerHello > Version 3.3 > > > In TLS 1.2 > > > > session_id[0]= > cipherSuite TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384 > > > But the negotiated cipher suggest that you have an ECC certificate. > > In TLSv1.2 and older, ciphers available are dependent on the > certificate type (RSA vs ECC). > > In TLSv1.3 its all different again and the certificate doesn't matter > for cipher negotiation. > > Get a RSA (2048bit) certificate instead of a ECC one from Let's > Encrypt, then you should be able to use those ciphers. > > > > Lukas

1 0

Timeout on forwarding REGISTRATION
by dries＠degendt.com 10 Jan '24

10 Jan '24

Hi all, My Kamailio server has been throwing a lot of errors during the manipulation and forwarding of registrations to my Asterisk server (192.168.96.97) using uac_req_send(); and t_relay() [core/tcp_main.c:1402]: tcp_do_connect(): connect 192.168.96.97:5060: (99) Cannot assign requested address [core/tcp_main.c:1399]: tcp_do_connect(): connect 192.168.96.97:5060 failed Cannot assign requested address When looking deeper into the code it appears that the errors are thrown when "tcp_async" is enabled. I've enabled this option a few weeks ago to make sure clients can REGISTER quickly. The parameter "tcp_reuse_port" is also set to yes. I'm suspecting that on busy moments, the destination can't be reached because no TCP connections are available. Is there any of the following parameters that I can adjust to prevent this issue? tcp_children = 25 disable_tcp = no tcp_max_connections = 63536 tcp_connection_lifetime = 3605 tcp_accept_aliases = no tcp_async = yes tcp_connect_timeout = 10 tcp_conn_wq_max = 65536 tcp_crlf_ping = yes tcp_delayed_ack = yes tcp_fd_cache = yes tcp_keepalive = yes tcp_keepcnt = 3 tcp_keepidle = 30 tcp_keepintvl = 10 tcp_linger2 = 30 tcp_rd_buf_size = 80000 tcp_send_timeout = 10 tcp_wq_blk_size = 2100 tcp_wq_max = 10485760 tcp_reuse_port=yes /* tcp_connection_match=1 */ open_files_limit=63536 A quick tcpdump doesn't show outgoing packet during the error events. Best regards, Dries

1 0

force TLS RSA cipher suite
by marek 10 Jan '24

10 Jan '24

hi, i'm trying force cipher list through options like modparam("tls", "cipher_list", "TLS_RSA_WITH_AES_256_CBC_SHA256") modparam("tls", "cipher_list", "RSA") ... but ssldump -i enp2s0 port 5061 shows every time ServerHello Version 3.3 session_id[0]= cipherSuite TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384 i tried lower crypto policy to LEGACY. nothing changed https://access.redhat.com/documentation/en-us/red_hat_enterprise_linux/9/ht… certificates are from Lets Encrypt if that does matter OS Rocky9(RHEL9), kamailio 5.7.3 from official repo any ideas/tips? thanks Marek

2 1

Re: uac_req_send + evroute + crash
by Péter Dr. Barabás 10 Jan '24

10 Jan '24

Hello, Any idea about the crash below? I found a ticket regarding to this, I made some comments and backtraces there also: https://github.com/kamailio/kamailio/issues/3635 Now this crash is very frequently, 3-4 times every day. There is a timer mechnism, where kamailio refreshes registrations towards asterisk automically to ensure continuous availability for mobile users. The crash exists 3-5 secs after this refresh process, but not always. Sometimes this process runs fine only CRITICAL logs appears (mentioned below) after it, but sometimes kamailio crashes during writing these CRITICAL logs. Peter From: Dr. Barabás Péter via sr-users <sr-users(a)lists.kamailio.org> Date: Wednesday, 2023. November 15. 17:03 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com> Subject: [SR-Users] Re: uac_req_send + evroute + crash Hello Daniel, I’m using kamailio 5.7.2 currently and this misterious crash occurs again. Now core files are generated also, here is the trace: (gdb) bt #0 __GI_raise (sig=sig@entry=6) at ../sysdeps/unix/sysv/linux/raise.c:50 #1 0x00007fae9a2d0859 in __GI_abort () at abort.c:79 #2 0x00005614ad79afd8 in qm_debug_check_frag (qm=0x7fae78e77000, f=0x7fae79805028, file=0x7fae77c8c134 "uac: uac_send.c", line=860, efile=0x5614ad936f99 "core/mem/q_malloc.c", eline=511) at core/mem/q_malloc.c:129 #3 0x00005614ad79f7c1 in qm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:511 #4 0x00005614ad7ab410 in qm_shm_free (qmp=0x7fae78e77000, p=0x7fae79805060, file=0x7fae77c8c134 "uac: uac_send.c", func=0x7fae77c8cf10 <__func__.15157> "uac_send_tm_callback", line=860, mname=0x7fae77c8c014 "uac") at core/mem/q_malloc.c:1350 #5 0x00007fae77c3686b in uac_send_tm_callback (t=0x7fae797dcf40, type=131072, ps=0x7fff8bd990d0) at uac_send.c:860 #6 0x00007fae783abc27 in run_trans_callbacks_internal (cb_lst=0x7fae797dcfe8, type=131072, trans=0x7fae797dcf40, params=0x7fff8bd990d0) at t_hooks.c:241 #7 0x00007fae783abd1e in run_trans_callbacks (type=131072, trans=0x7fae797dcf40, req=0x0, rpl=0x0, code=0) at t_hooks.c:261 #8 0x00007fae782bd8cb in free_cell_helper (dead_cell=0x7fae797dcf40, silent=0, fname=0x7fae783e9f62 "timer.c", fline=653) at h_table.c:165 #9 0x00007fae7837649c in wait_handler (ti=724701450, wait_tl=0x7fae797dcff8, data=0x7fae797dcf40) at timer.c:653 #10 0x00005614ad50be68 in timer_list_expire (t=724701450, h=0x7fae78ef5d18, slow_l=0x7fae78ef95e0, slow_mark=33641) at core/timer.c:857 #11 0x00005614ad50c3a7 in timer_handler () at core/timer.c:922 #12 0x00005614ad50c8d5 in timer_main () at core/timer.c:961 #13 0x00005614ad45fbf3 in main_loop () at main.c:1833 #14 0x00005614ad46ae93 in main (argc=14, argv=0x7fff8bd99b48) at main.c:3086 I think it occurs only if $uac_req(evroute) is set to 1 and I define the event_route[uac:reply] route. My uac_reply route is easy: if ($uac_req(evcode)==200) { xlog("L_INFO", "Registration for user $fU has been refreshed successfully."); $sht(vtp=>asterisk_restarted) = $null; route(SAVE_REG); } else { xlog("L_INFO", "Registration failed: $uac_req(evcode)"); } the SAVE_REG route is the following: route[SAVE_REG] { xlog("L_INFO", "Saving pbx registration...:$fU"); $var(maxExpiry) = MAX_PBX_REG_EXPIRY; route(GET_CLIENT_ID); sql_pvquery("ca", "select count(*) from pbxusers where user = '$fU'", "$var(pbxusersCnt)"); if ($var(pbxusersCnt)>0) { if ($var(clientId) == 0) { sql_query("ca", "update pbxusers set expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } else { sql_query("ca", "update pbxusers set client_id = '$var(clientId)', expiry = TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), online=1 where user = '$fU'"); } } else { sql_query("ca", "insert into pbxusers (user, expiry, client_id, online) values ('$fU', TIMESTAMPADD(SECOND, $var(maxExpiry), SYSDATE()), '$var(clientId)', 0)"); } } And GET_CLIENT_ID route is: route[GET_CLIENT_ID] { $var(clientId) = 0; if (is_present_hf("X-AT-ClientId")) { $var(clientId) = $hdr(X-AT-ClientId); } } Any idea, is it a bug or I do something wrong in config files? Peter From: Dr. Barabás Péter via sr-users <sr-users(a)lists.kamailio.org> Date: Sunday, 2023. October 8. 18:49 To: miconda(a)gmail.com <miconda(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com> Subject: [SR-Users] Re: uac_req_send + evroute + crash Hello Daniel, I do not see any “failed to send request with authentication”. The next CRITICAL log appears after calling uac_req_send(): CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring Now we use kamailio 5.7.2 and no crash has come, but the critical logs above exist. I saw a ticket in github: https://github.com/kamailio/kamailio/issues/3522, may the crash be similar or the same to that? I will make a try next week with downgrading to 5.6.2 and try to reproduce the crash. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Friday, 2023. October 6. 13:55 To: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com>, Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, do you also get error log messages that include "failed to send request with authentication"? Which CRITICAL log appears when you call uac_req_send()? You pasted a couple of them in the initial email? Cheers, Daniel On 05.10.23 17:32, Dr. Barabás Péter wrote: Hello, I used kamailio version 5.6.2. I refreshed to 5.7.2 today. As I remember it was in kamailio long time ago therefore I could skip evroute route. But this CRITICAL log appears always when I call uac_req_send(). I call it with settings: $uac_req(auser) = $var(username); $uac_req(apasswd) = $var(password); Where username and password are retrived from web service before call. In event_route I got first 401 after 200. Peter From: Daniel-Constantin Mierla <miconda(a)gmail.com><mailto:miconda@gmail.com> Date: Thursday, 2023. October 5. 17:25 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Cc: Dr. Barabás Péter <dr.peter.barabas(a)gmail.com><mailto:dr.peter.barabas@gmail.com> Subject: Re: [SR-Users] uac_req_send + evroute + crash Hello, On 05.10.23 16:34, Dr. Barabás Péter via sr-users wrote: Hi All, I use kamailio In front of Asterisk and kamailio needs to refresh registrations periodically towards Asterisk to ensure the availability of users from Asterisk side. I use uac module and call uac_req_send() for sending REGISTER requests. I set $uac_req(evroute)=1 The event_route[uac:reply] is called fine, but in kamailio logs I see the next lines: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) - ignoring After some time, kamailio has crashed. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f114995ada8), called from uac: uac_send.c: uac_send_tm_callback(860), first free uac: uac_send.c: uac_resend_tm_callback(732) – ignoring CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 34 ALERT: <core> [main.c:774]: handle_sigs(): child process 1407950 exited by a signal 6 what version of Kamailio are you using? Is Asterisk challenging for authentication? Does it happen every time or seldom? Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy and Development Services Kamailio Advanced Training - Online - Nov 14-16, 2023 -- asipto.com

2 1

Crash of Kamailio when using module http_async_client - function http_async_query, if module "tm.so" is not loaded (Kamailio 5.7.3)
by Chaigneau, Nicolas 10 Jan '24

10 Jan '24

Hello, When using http_async_query function, provided by module http_async_client: https://www.kamailio.org/docs/modules/devel/modules/http_async_client.html#… If module "tm.so" is not loaded, then immediately upon calling function http_async_query, Kamailio crashes: 9(3807541) CRITICAL: <core> [core/pass_fd.c:281]: receive_fd(): EOF on 8 0(3807527) ALERT: <core> [main.c:791]: handle_sigs(): child process 3807533 exited by a signal 11 0(3807527) ALERT: <core> [main.c:795]: handle_sigs(): core was generated 0(3807527) INFO: <core> [main.c:818]: handle_sigs(): terminating due to SIGCHLD Of course, I now know that I must load the module "tm.so". But I think it would be better to catch the issue when Kamailio starts, rather than crashing later on when trying to call http_async_query. What do you think ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

3 3

How to use uac.reg_add command?
by mm e 06 Jan '24

06 Jan '24

./kamcmd uac.reg_add 789798 789798 mydomain.com 789798 myteldomain.com . 789798 123456 . sip:sip6.telco.nl 360 0 10 mydomain.com:5060 udp:mydomain.com:5060 error: 400 - Invalid Parameters Can you provide me with a case, I don't know how to enter the correct format.

2 1

DB invalid connection parameter
by Sergiu Pojoga 04 Jan '24

04 Jan '24

Hi, Facing the following error after bumping Kam from 5.3 => 5.4 => 5.5 => 5.6, following instructions on each step about DB structure and module changes. These errors show up since. Any clue what they mean? A couple other modules are using the same DB MySQL connector. The data in the DB also seems to get updated properly. Jan 03 15:08:30 sip kamailio[20881]: ERROR: <core> [db.c:489]: db_use_table(): invalid connection parameter *Jan 03 15:08:30 sip kamailio[20881]: ERROR: usrloc [ucontact.c:1170]: db_update_ucontact_ruid(): sql use_table failedJan 03 15:08:30 sip kamailio[20881]: ERROR: usrloc [urecord.c:438]: wb_timer(): updating contact in db failed (aor: user(a)domain.com <user(a)domain.com>)* version: kamailio 5.6.5 (x86_64/linux) 48d48f compiled on 02:32:48 Jan 2 2024 with gcc 4.9.2 DB structure: location | CREATE TABLE `location` ( `id` int(10) unsigned NOT NULL AUTO_INCREMENT, `ruid` varchar(64) NOT NULL DEFAULT '', `username` varchar(64) NOT NULL DEFAULT '', `domain` varchar(64) DEFAULT NULL, `contact` varchar(512) NOT NULL DEFAULT '', `received` varchar(128) DEFAULT NULL, `path` varchar(512) DEFAULT NULL, `expires` datetime NOT NULL DEFAULT '2030-05-28 21:32:15', `q` float(10,2) NOT NULL DEFAULT 1.00, `callid` varchar(255) NOT NULL DEFAULT 'Default-Call-ID', `cseq` int(11) NOT NULL DEFAULT 1, `last_modified` datetime NOT NULL DEFAULT '2000-01-01 00:00:01', `flags` int(11) NOT NULL DEFAULT 0, `cflags` int(11) NOT NULL DEFAULT 0, `user_agent` varchar(255) NOT NULL DEFAULT '', `socket` varchar(64) DEFAULT NULL, `methods` int(11) DEFAULT NULL, `instance` varchar(255) DEFAULT NULL, `reg_id` int(11) NOT NULL DEFAULT 0, `server_id` int(11) NOT NULL DEFAULT 0, `connection_id` int(11) NOT NULL DEFAULT 0, `keepalive` int(11) NOT NULL DEFAULT 0, `partition` int(11) NOT NULL DEFAULT 0, PRIMARY KEY (`id`), UNIQUE KEY `ruid_idx` (`ruid`), KEY `account_contact_idx` (`username`,`domain`,`contact`), KEY `expires_idx` (`expires`), KEY `connection_idx` (`server_id`,`connection_id`), *KEY `tcpcon_idx` (`connection_id`)* ) ENGINE=InnoDB AUTO_INCREMENT=287989 DEFAULT CHARSET=latin1 > select * from version where table_name='location'; table_name: location table_version: 9 Thanks.

2 2

webRTC Test Client?
by Andy Newlands 03 Jan '24

03 Jan '24

Hi, I have a Freeswitch installation, fronted by a Kamailio proxy I would like to configure Kamailio to bridge SIP between webRTC clients and Freeswitch. The first issue I hit is how to set up a test webRTC client to make calls into Kamailio/FS. Can anyone point me to a simple webRTC client I can use to make test webRTC/SIP calls into my setup? Thanks BTW: I'm not JS experienced (I work with Python, C#, C/C++). Kind regards, Andy

6 5

Kamailio - Initiating invite on REFER
by Muhammad Danish Moosa 02 Jan '24

02 Jan '24

Hi champs, In this flow like endpoint -> B2BUA 1-> kamailio -> B2BUA 2, there is a requirement that REFER should be handled by Kamailio and it should manage the transfer , initiating a new invite based on REFER. Any pointer will be helpful -- Danish

3 2

Connect to a remote ENUM(E.164 Number Mapping) server
by elhar.mohamed＠gmail.com 25 Dec '23

25 Dec '23

Hello, I'm using the Kamailio version 5.7.x, I didn't find in the module "enum.so" any option that allow to connect to a remote ENUM server. Could you please help me ! Thanks in advance

2 3

Merry Christmas and Happy Holidays 2023-2024!
by Daniel-Constantin Mierla 25 Dec '23

25 Dec '23

Kamailio project continued to evolve steadily, a new major release was out during this year, next one planned in a few months, relying on an amazing community of developers and users. We hope it kept helping people and companies world wide to connect with the dear ones and build a better way of living. The season holidays are ahead, time to relax, spend time with family and friends! I take again this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2023 another fantastic year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services Kamailio World Conference - April 18-19, 2024, Berlin

2 1

kemi python3s - module 'KSR' has no attribute 'nathelper'
by marek 22 Dec '23

22 Dec '23

hi, trying kamailio 5.7.3 with KEMI pythons3s sample from https://github.com/kamailio/kamailio/blob/master/misc/examples/kemi/kamaili… trying simple register with sipsak sipsak -U --sip-uri "sip:cervajs@kamailio.example.com" -u cervajs -a top_secret 4(28805) ERROR: PY8 {REGISTER}: app_python3s [apy3s_kemi.c:141]: apy3s_exec_func(): error exception occurred 4(28805) ERROR: PY8 {REGISTER}: app_python3s [apy3s_exception.c:167]: apy3s_handle_exception(): apy3s_exec_func: ksr_request_route((null)): Unhandled exception in the Python code: Traceback (most recent call last): File "/etc/kamailio/kamailio.py", line 31, in ksr_request_route if ksr_route_natdetect()==-255 : File "/etc/kamailio/kamailio.py", line 238, in ksr_route_natdetect if KSR.nathelper.nat_uac_test(19)>0 : AttributeError: module 'KSR' has no attribute 'nathelper' i see nat_uac_test in doc https://kamailio.org/docs/tutorials/5.7.x/kamailio-kemi-framework/modules/#… any tips/ideas? happy christmas Marek

1 1

Errors with TLS
by SAMUEL MOYA TINOCO 21 Dec '23

21 Dec '23

Hi everyone, All my environment was working properly. But a few days ago I've started to have problems with my websocket connectios. Everytime I try to register my websocket Kamailio log this Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1330]: tls_h_read_f(): protocol level error Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_util.h:50]: tls_err_ret(): TLS accept:error:0A000416:SSL routines::sslv3 alert certificate unknown (sni: unknown) Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1334]: tls_h_read_f(): src addr: IP_webclient:61423 Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: tls [tls_server.c:1337]: tls_h_read_f(): dst addr: IP_Kamailio:8443 Dec 20 09:02:10 prod-kamailio /usr/sbin/kamailio[599]: ERROR: <core> [core/tcp_read.c:1476]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7fbc423053b0 r: 0x7fbc423054d8 (-1) My tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = yes private_key = /etc/kamailio/cert/clave_SSL.key certificate = /etc/kamailio/cert/certificado.pem Kamailio version: root@prod-kamailio:/etc/kamailio# kamailio -v version: kamailio 5.6.3 (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, 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 12.2.0 I don't know why it stopped working. And I'm not sure what to do or what to check. Has anyone faced this problem? Any help would be appreciated Thanks in advance Samuel Moya Tinoco Departamento de Sistemas y Redes Móvil: (+34) 606985997 smoya(a)vivelibre.es<mailto:smoya@vivelibre.es> [cid:image001.png@01DA33E6.D2D833D0] Soluciones inteligentes para la autonomía personal

2 2

How to hang up the call in xhttp routing
by mm e 21 Dec '23

21 Dec '23

event_route[xhttp:request] { if ($hu =~ "hangupCall"){ $var(body) = $rb; xhttp_reply("200", "OK", "",""); jansson_get_field($var(body), "callid", "$var(callid)"); jansson_get_field($var(body), "cseq", "$var(cseq)"); if (t_cancel_callid("$var(callid)", "$var(cseq)", "22")) { xlog("L_NOTICE", "t_cancel_callid============"); } exit; } xhttp_reply("404", "Not Found", "",""); exit; } According to this route, the answered call cannot be hung up。How should I hang up the answered call in xhttp routing, similar to freeswitch's uuid_kill。

1 0

Kamailio - Ms Teams direct Routing - dispatcher issues | FLAGS: IP
by faisal.jamil.khan＠gmail.com 19 Dec '23

19 Dec '23

Hi Team, I am trying to setup Kamailio with MS Teams Direct Routing and after resolving TLS issues now I am getting this at my dispatcher flags. Possible suggestions required. kamcmd dispatcher.list | egrep "URI|FLAGS" URI: sip:sip.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP URI: sip:sip2.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP URI: sip:sip3.pstnhub.microsoft.com:5061;transport=tls FLAGS: IP Dec 14 20:23:27 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKd9f6.c2e7d2f4000000000000000000000000.0 To: <sip:sip.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-c7b52d5e CSeq: 10 OPTIONS Call-ID: 23b2dc436850e949-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux)) Dec 14 20:23:28 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip2.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKccf6.53fea6f0000000000000000000000000.0 To: <sip:sip2.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-7a126c3a CSeq: 10 OPTIONS Call-ID: 23b2dc436850e94a-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux)) Dec 14 20:23:28 abcsbc.com /usr/sbin/kamailio[9479]: INFO: <script>: Sent out tm request: OPTIONS sip:sip3.pstnhub.microsoft.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS abcsbc.com:5061;branch=z9hG4bKdcf6.5a4f89e6000000000000000000000000.0 To: <sip:sip3.pstnhub.microsoft.com:5061;transport=tls> From: <sip:abcsbc.com>;tag=64ff6b492a7d9ab14de1f0b7c15c9c17-992aa610 CSeq: 10 OPTIONS Call-ID: 23b2dc436850e94b-9479(a)0.0.0.0 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.6.5 (x86_64/linux))

3 33

Info: Next Kamailio World Conference - April 18-19, 2024, Berlin, Germany
by Daniel-Constantin Mierla 18 Dec '23

18 Dec '23

Hello, a short announcement to inform that the dates for next Kamailio World Conference have been set to April 18-19, 2024, the event to take place again in Berlin, Germany. It will be a different venue than the past editions, that one not being available. A pre-conference day is considered on April 17, 2024, but it is not yet known in what form and where. The conference is scheduled a bit earlier than the usual in the year because Germany hosts the Euro 2024 football championship during June-July 2024, making May very busy and harder to find adequate conferencing space in Berlin. The website and more details will be published soon. 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

Regretfully we were unable to process the URI
by Gavin Baumanis 18 Dec '23

18 Dec '23

Hi Everyone, Very new to SIP And to Kamailio Configuration / administration. I have a working AWS instance with Kamailio installed on it - and it works fine. We are creating a whole new environment and I have cloned the existing AWS EC2 instances. I have edited all the configs to ensure that they are using the correct DNS names and / or IP addresses. Otherwise the "configs" are identical. While attempting to call a SIP number, I get the following message, returned to the soft phone I am using (microSIP) *Regretfully we were unable to process the URI (479/SL)* I can also see the same error message in the console of the Kamailio server. I am hoping that this might mean something to someone enough - to suggest some troubleshooting steps, please. Thanks;

3 3

Encoded Call-ID - Topoh module
by Vanderlei Torres Batistela 15 Dec '23

15 Dec '23

Hello all, How can I get the encoded Call-Id generated by TOPOH module to store in the CDR ? Regards, Vanderlei

2 1

{uri.duri} not working
by hanxiaofang＠x-y-t.cn 13 Dec '23

13 Dec '23

Hi Kamailio experts, OS: Debian11 Kamailio ver: 5.7.3 {uri.duri} doesn't seem to work The following script can be reproduced: ``` debug=2 log_stderror=yes # mpath="/usr/local/lib/kamailio/modules/" loadmodule "tm.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "ctl.so" loadmodule "htable.so" modparam("tm", "failure_reply_mode", 3) modparam("tm", "fr_timer", 30000) modparam("tm", "fr_inv_timer", 120000) request_route { xinfo("*** $sel(msg)\n"); sl_send_reply("404", "Not here"); exit; } event_route[htable:mod-init] { $var(ouri) = "sip:alice@server.com:5060;nat=yes;transport=tcp;line=xyz"; $var(duri) = $(var(ouri){uri.duri}); # => "sip:server.com:5060;transport=tcp" xinfo("ouri = $var(ouri)\n"); xinfo("duri = $var(duri)\n"); if ($var(duri) == "sip:server.com:5060;transport=tcp" ) { xinfo("uri.duri work well\n"); } else { xerr("something is wrong\n"); } } ```

2 4

Proxy Inactivity and Kamailio Log Disruption Issue.
by satyaprakash ch 12 Dec '23

12 Dec '23

HI, Scenario: We possess 2 proxies and 2 registrar servers equipped with dmq functionality. Unexpectedly, one of the Proxy IP addresses was marked as inactive (FLAG: IP) in the dispatcher list, causing the proxy to cease responding. Simultaneously, Kamailio logs stopped recording any activity. Upon restarting the Kamailio service, the problem was resolved, and the proxy resumed normal operation, with the Flag changing back to AP (ACTIVE). The sudden occurrence of this issue is not clear, and due to the cessation of logs during the incident, we lack information for the investigation. Any suggestions on how to address this issue would be appreciated. And is this scenario happened to any one in your experience?

2 3

usrloc ka_mode - Flooding OPTIONS at same time?
by Benoit Panizzon 12 Dec '23

12 Dec '23

Hi We have reached 20000 registered CPE and start facing issues. Am I observing correctly that when ka_mode 4 is enabled, OPTIONS are send simultaneously to all registered CPE and not spread over the interval? They could be the cause very high pps peaks on network equipment? If so, is there a way to spread the ka options over time so they don't cause such high peaks? Or even better, is there a way to enable them for only some customers via a variable? 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 ______________________________________________________

4 5

is_method_f(): cannot parse cseq header in xhttp:request event route?
by Benoît Panizzon 12 Dec '23

12 Dec '23

Hi all I use event_route[xhttp:request] on Kamailio 5.5 Every time a xhttp request is processed I get this log entry: ERROR: textops [textops.c:3541]: is_method_f(): cannot parse cseq header Almost identical config on Kamailio 5.6 does not show this error. Is that something that got fixed in 5.6? I see a change in the documentation, While 5.5 does not state $rm can be accessed, 5.6 states $rm contains the HTTP method. Or do I somehow leak the http request to somewhere I call is_method() on it after terminating the event route? How should the event_route[xhttp:request] properly be terminated? With 'return' or with 'exit'? -- 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 ______________________________________________________

1 0

Upgrade Minor versions
by Gavin Baumanis 11 Dec '23

11 Dec '23

Hi there, I have taken over a project that uses Kamailio. I am aware of what we use it for - I am just not sure on the how... apart from what I have read in the docs. We're currently running 5.4.2 and I'd like to upgrade to (at least) the latest 5.4 offering - as that seems as if it would be the easiest and safest option. So I am hoping I might get some direction, please - on the way forward for an upgrade. The documentation seems OK if you're upgrading minor versions... but I couldn;t find anything about upgrading patch versions. Thanks!

2 2

Updated: Online devel meeting on matrix - 15:00 UTC, Dec 5, 2023
by Daniel-Constantin Mierla 05 Dec '23

05 Dec '23

Hello, we should consider an online devel meeting sometime soon to summarize what was done at (and still needs to be done after) devel meeting in Dusseldorf and plan a bit the targets for next major release (5.8 or 6.0)? If considered useful, I propose Dec 5 at 15:00UTC (16:00 Berlin/Paris/Madrid/Rome), but we can also look for other dates as well. Topics to be discussed can be added at: - https://github.com/kamailio/kamailio-wiki/blob/main/docs/devel/irc-meetings… Pull requests can be made by users without git access. Cheers, Daniel PS: Previous announcement indicated the start time 14:30 UTC, this message changes it to 15:00 UTC - 30min later due to a constraint in availability of one of the Kamailio developers. -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 3

stats.get_statistucs dialog: counter over/underrun?
by Benoit Panizzon 05 Dec '23

05 Dec '23

Hi List Kamailio 5.5 # kamcmd stats.get_statistics dialog: dialog:active_dialogs = 17 dialog:early_dialogs = 18446744073709551615 dialog:expired_dialogs = 0 dialog:failed_dialogs = 31198 dialog:processed_dialogs = 107519 Where could this early_dialog value come from? 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 ______________________________________________________

1 0

Setting up Kamailio as SIP Proxy
by Saul Ibarra 04 Dec '23

04 Dec '23

Hi, I am needing some help setting up kamailio as a sip proxy between a provider(trunk) and a test pbx The trunk requires username&password and i would like to add that information to the request within the Kamailio proxy, so that the PBX can authenticate directly by IP (without username&password) through the Kamailio proxy. As a programmer with a limited background in SIP (currently learning), I am keen on understanding the intricacies of this setup at the SIP level. Could you please direct me to resources or documentation that covers scenarios similar to mine? Any help or guidance you could provide would be greatly appreciated. Thanks

4 8

tsilo dependency on usrloc
by Alex Balashov 03 Dec '23

03 Dec '23

Hi, I wanted to revisit the topic of tsilo dependence on the location service. All three ts_append*() functions have the following quality: - ts_append(): "performing a contact lookup on the table specified by the domain parameter." - ts_append_by_contact(): "the contact lookup is performed" - ts_append_to(): "performing a contacts lookup on the table specified by the domain parameter" Why is this extensive coupling to usrloc necessary? This makes it impossible to use `tsilo` in case of providing a push-notification add-on that front-ends an upstream registrar, requiring a kind of local shadow registrar or mid-registrar. What would make more sense is a generic mechanism that allows one to "drip" new contacts into an existing transaction, whether suspended or active. Kamailio has a mechanism to add more branches to an existing transaction, but the scope of that mechanism is only from *inside* the vantage point of the transaction in question. The key parlour trick of `tsilo` is that it permits dripping new branches into a *different* transaction. ts_append_to() almost does the trick, providing a target index and label, but it just insists on doing a registrar lookup to source the contacts. What is really wanted and needed for the downstream PN gateway use-case is a means of extracting contacts from incoming registrations (or other sources, potentially) without storing them in any fashion locally, without using or even loading usrloc, and just throwing them over the fence into a different transaction. Is this somehow possible by means other than tsilo? Am I overlooking something? Cheers, -- Alex -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800

2 3

Unable to use db_alias_lookup to call multiple extensions
by Merul Patel, PhD 01 Dec '23

01 Dec '23

Hi, Kamailio newby here. Having some trouble with using alias_db to forward an incoming call to two extensions. Thanks in advance for any help. I've tried using the Ubuntu Jammy ARM64 packages, as well as compiling the source for version 5.7.3. In both cases, I've added three extensions, and calls directly between them work fine with the vanilla kamailio.cfg. However, I then wanted to add an alias so that when called it would dial two extensions, but only the first extension is ever called. My kamailio.cfg only differs in a few places: $ diff kamailio.cfg kamailio.cfg.original 3,6d2 < #!define WITH_AUTH 1 < #!define WITH_DEBUG 1 < #!define WITH_ALIASDB 1 < # 268,270d263 < loadmodule "db_sqlite.so" < #!define DBURL "sqlite:///usr/local/kamailio-devel/etc/kamailio/kamailio.sqlite" < 367c360 < modparam("tm", "failure_reply_mode", 0) --- > modparam("tm", "failure_reply_mode", 3) 450d442 < modparam("alias_db", "append_branches", 1) I can see from the log that when dialling the desired alias (5000) an alias_db query is performed: Dec 01 20:00:23 merrivale /usr/local/kamailio-devel/sbin/kamailio[3264]: DEBUG: {1 2 INVITE 651794328853009-230332414159388(a)192.168.17.145} alias_db [alookup.c:169]: alias_db_query(): new URI [0] is [sip:6002@merrivale.localdomain] Dec 01 20:00:23 merrivale /usr/local/kamailio-devel/sbin/kamailio[3264]: DEBUG: {1 2 INVITE 651794328853009-230332414159388(a)192.168.17.145} alias_db [alookup.c:169]: alias_db_query(): new URI [1] is [sip:6001@merrivale.localdomain] But only the first result is called. I've changed the order of the aliases in the sqlite db and just the first result is called. What am I missing? Merul

1 0

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