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37 participants
37510 discussions

http_async_client and sl_send_reply()
by Masud Muborakshohi 02 May '24

02 May '24

Hello! I have a problem when sending a response sl_send_reply("415", "The other party does not support video calling!"); in route[HTTP_REPLY]. When I make a request using http_async_query() to send a push notification to a mobile device, I need to process the response from the push server and respond with a SIP message to the client. Before sending the request I do $http_req(suspend) = 0; What am I doing wrong? Why are $fU and $ru equal to null ? #processing HTTP responses route[HTTP_REPLY] { if ($http_ok && $http_rs == 200) { xlog("L_INFO", "route[HTTP_REPLY]: status $http_rs\n"); xlog("L_INFO", "route[HTTP_REPLY]: body $http_rb\n"); } else if($http_rs == 415 || $http_rs == "415") { xlog("L_ERR", "route[HTTP_REPLY]: caller: $fU phone: $ru error: $http_err)\n"); send_reply("415", "The other party does not support video calling!"); } else { xlog("L_ALERT", "route[HTTP_REPLY]: error $http_err)\n"); } #exit; }

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No ACK from Kamailio
by Arseniy Moskvich 02 May '24

02 May '24

Hi everyone. We are new to Kamailio and using Kamailio 5.8 with FreeSwitch. We have the issue with the incoming call from our provider. They are dropping after 30 seconds. The client doesn't receive ACK from Kamailio. But FreeSwitch sends ACK to Kamailio. The outbound call is working fine. At the moment flow is our provider -> FreeSwitch -> Kamailio -> our web client. Could someone give us some directions please? Please find attached Kemi.py. Thank you so much in advance

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Kamailio works but voice is not present during the calls!
by christian.marinelli＠hotmail.it 02 May '24

02 May '24

Hi all, i have a problem with my Kamailio SIP Server. I set up a Kamailio SIP server in a virtual machine on a private network and i connect to this with a WireGuard VPN. The problem is that i can connect to the SIP server throught different clients and users and i can call the other users, the devices ring and i can answer to the calls but unfortunatly there is no audio during the call :( I can't understand why, it seems all si OK, a year ago i set up another SIP server with the same configuration and all works correctly with it. Can anyone help me to understand why? I can copy and paste here all the config files if you need it. Thank you so much in advance Christian

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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)

1 1

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

1 0

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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