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37398 discussions

Dispatcher module and hash algorithm
by Luis Rojas G. 12 Jun '20

12 Jun '20

Hello, I noticed a couple of strange behaviors, using this version : kamailio-5.3.3-4.1.x86_64 I am using algorithm "0". Hash over Call-ID. The reason for that is that I am just re-sending the message, using send_udp(), to a destination in a group from file dispatcher.list. Kamailio is completely stateless, it does not need to remember anything. And using a hash over Call-ID I can be sure that any other message related to the same call will be sent to the same destination : retransmissions of the INVITE, CANCEL, ACK, BYE, etc. I general, it works, but I detected two problems. I have a group of 5 destinations, where every server is running at 80% of the nominal load. It's a 4 + 1 configuration, for redundancy. I know it's not 100% evenly distributed using hash over Call-ID, but in practice it's almost perfectly even. However, if one node is down, instead of having the four alive running at 100%, I get three running still at 80% capacity, and one is sent 160% of the load (which of course can't process). it seems, all traffic that was supposed to be sent to the failed node is transferred to the same unique destination. That implementation makes my solution for redundancy worthless. Why didn't the implementation use, for instance, a re-hash over the hash? That would allow to redistribute "evenly" over the remaining servers. I tried to solve this by repeating the entries in dispatcher.list, as I noticed that kamailio doesn't check if different lines contain duplicated uris. For instance, if I have two lines with "destination A", and one line with "destination B", like this: 0 sip:server_A 0 sip:server_A 0 sip:server_B I see that "destination A" receives twice the amount of SIP OPTIONS that "destination B" receives. And so, I used this dispatcher.list : 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 0 sip:server_1 0 sip:server_3 0 sip:server_5 0 sip:server_2 0 sip:server_4 0 sip:server_1 0 sip:server_4 0 sip:server_2 0 sip:server_5 0 sip:server_3 0 sip:server_1 0 sip:server_5 0 sip:server_4 0 sip:server_3 0 sip:server_2 (it's not a random order. It follows a sequence) I thought : "what a genius I am. This way, if any node fails, and kamailio select the "next" to send the traffic to, it will distribute evenly over the rest". It doesn't work. I detected something that looks like a bug. If one entry is repeated at least once, for instance : 0 sip:server_1 0 sip:server_1 0 sip:server_2 0 sip:server_3 0 sip:server_4 0 sip:server_5 if server_1 is down, kamailio still sends some of the INVITEs to it. Not all of them, but many. I can see the SIP OPTIONS been sent to it, and the response "ICMP Destination Unreachable", so Kamailio knows that server_down. However, still sends INVITE requests to that failed node. Look at this trace : You can see all the failed SIP OPTIONS. Still, kamailio sends traffic to that server. It does not send traffic to a failed destination if it's listed only once in dispatcher.list Any ideas? Thanks in advance, Luis -- Luis Rojas Software Architect Sixbell Los Leones 1200 Providencia Santiago, Chile Phone: (+56-2) 22001288 mailto:luis.rojas@sixbell.com http://www.sixbell.com

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RPC Command tm.clean
by Marat Gareev 12 Jun '20

12 Jun '20

Hello. I have a philosophical question about tm.clean <https://kamailio.org/docs/modules/5.1.x/modules/tm.html#tm.rpc.clean> command. Unfortunately, I don’t understand what this command is for, because in my view, expired transactions are automatically deleted. This is all I could find: pc command tm.clean to clean long time expired transactions (for troubleshooting purposes) [1] <https://www.kamailio.org/wiki/features/new-in-5.1.x> So, can you give an example when a transaction can remain in memory? Thanks!

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Queries regarding Load balancing setup
by Abdul Rasheed 12 Jun '20

12 Jun '20

Hi , I am a newbie who started working on Kamailio only few days ago. I am trying to setup kamailio in such a way that kamailio acts as a load balancer for 2 or more asterisk systems. i have following queries. 1. As i understand , all the SIP registration will be done with Kamailio and the Media of the calls will be routed to Asterisks in a round robin fashion. In such a case , is it required that all the extension numbers to be used are to be configured in all the asterisk systems included in the setup ? 2. If extension 1000 is in asterisk1 and extension 1001 is in asterisk2 , then is it possible to make internal calls between these extensions ? 3. Is it possible to have TLS based registrations to Kamailio 4. My plan is to build a system which can handle 10000s of extensions . So the system is supposed to be scaled up by adding more asterisk servers in order to accommodate more and more extensions as required . THanks and Regards Abdul Rasheed

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Prometheus module stat
by David Villasmil 12 Jun '20

12 Jun '20

Hello all, Anyone here uses the prometheus module? There’s a core stat that includes a hyphen (“-“) and Prometheus doesn’t support that, only underscores are allowed kamailio_usrloc_location-contacts I’m thinking maybe I can create a custom counter and set it to that stat but with the correct name, is that possible? -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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facing Call Connection and Disconnection issue in Kamailio SIP Server
by Rajkumar Samikkannu 12 Jun '20

12 Jun '20

Hi There, This is Raj. We have successfully installed Kamailio SIP Server by following this documentation link https://kamailio.org/docs/tutorials/5.3.x/kamailio-install-guide-git/#insta… We created some users and those users are in the Subscription Table. We registered(logged in) some users on our client side. We could get Acknowledgement while we registered(logged in). We couldn't get any acknowledgement while we tried to make a call or send a message. We got a "Request Timeout" Error in Client Side. If anyone knows about this issue, please guide us what we should do. Thanks in advance.

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Which mode to capture "written to network" traffic with homer and topos
by Karsten Horsmann 12 Jun '20

12 Jun '20

Hi Mailinglist, some of my kamailio-version is 5.3.4 with topos. During the last discussion about sip_trace_mode("t") i found new behaivor. With this tracingmode i see the "internal" before topos writing. Here my questions: Can i change it to "whats on the wire" (topos rewritten mode)? I didnt checked all modes. And if so, can i switch that on-the-fly with $var for example? For topos debugging its fine with sip_trace_mode("t") - not for "hey carrier, you made some mistakes". #!ifdef WITH_HOMER #Siptrace modparam("siptrace", "force_send_sock", "MY_LOCAL_SENDSOCK") # local interface from where to send the duplicated traffic modparam("siptrace", "duplicate_uri", "MY_HOMER_CAPTURE") modparam("siptrace", "hep_mode_on", 1) # enable/disable Homer encapsulate mode modparam("siptrace", "hep_version", 3) # HEPv3 the timestamp and capture agent ID will be included in the HEP header modparam("siptrace", "trace_to_database", 0) # trace to database modparam("siptrace", "trace_on", 1) # enable/disable trace modparam("siptrace", "trace_mode", 0) # default 0, if 1 then you dont need call siptrace flag or sip_trace() modparam("siptrace", "hep_capture_id", 8) # capture agent id Limitation: 32-bit for HEPv3. #!endif #!ifdef WITH_HOMER # Set the tracing mode: message, transaction or dialog. Need 5.3.x sip_trace_mode("t"); #!endif *** captured on the topos kamailio: 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Record-Route: <sip:172.24.32.28;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Record-Route: <sip:100.80.0.20;transport=tcp;r2=on;lr=on;ftag=1ea69f36;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAO3VzZXI9cGhvbmU-;vst=AAAAAAAAAAAAAAAAAAAAAAAAAABLXVQATFVCVUheUUgcXAIBHnBvcnQ9dGNwO3VzZXI9cGhvbmU-;proxy_media=yes;nat=yes> Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Via: SIP/2.0/TCP AA.AAA.AAA.AAA:5060;rport=5060;branch=z9hG4bKeca7a7f789c9f647560353e3cf1cd592.ba7930dc Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Contact: <sip:TFFRA002@AA.AAA.AAA.AAA:5060;transport=tcp> Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk --------------------------------------- *** captured on the kamailio behind 172.24.52.3 2020-06-11 08:35:11 +0200 : 172.24.32.28:5060 -> 172.24.52.3:5060 INVITE sip:+49XXXXXXXXXXXX@172.24.52.3:5060 SIP/2.0 Via: SIP/2.0/UDP 172.24.32.28;branch=z9hG4bK34b.1b6e5fbd9cf7af26bb168d1469096714.0;i=74 Max-Forwards: 52 To: <sip:+49XXXXXXXXXXXX@some.provider.example> From: <sip:+49YYYYYYYYYYY@some.provider.example>;tag=1ea69f36 Call-ID: 3af87db089119841(a)AA.AAA.AAA.AAA Supported: histinfo,replaces CSeq: 647169 INVITE Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, REFER, REGISTER, UPDATE P-Asserted-Identity: <sip:+49BBBBBBBBBB@some.provider.example;user=phone> Content-Type: application/sdp Content-Disposition: session Content-Length: 298 X-Group-SBC: 2006 X-Routing-SBC: un2tr-cgn-trunk Contact: <sip:btpsh-5ee1105a-1903c-1@172.24.32.28> -- Cheers *Karsten Horsmann*

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Re: [SR-Users] [EXT] RE: kamailio 522 S-CSCF failing to start
by Martin W Woscek 12 Jun '20

12 Jun '20

What version of MySQL should we be at for kamailio 5.2.2? Thanks, Martin ________________________________ From: Henning Westerholt <hw(a)skalatan.de> Sent: Monday, June 8, 2020 4:49:54 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Cc: Martin W Woscek <mwoscek(a)mitre.org> Subject: [EXT] RE: kamailio 522 S-CSCF failing to start Hello, the error message below is from the DB startup version check. So do you have sometimes DB errors on this virtual machine? Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Martin W Woscek Sent: Monday, June 8, 2020 2:46 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] kamailio 522 S-CSCF failing to start Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

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How to change contact in SIP INVITE
by Trần Đức Thành 09 Jun '20

09 Jun '20

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Reconnect call
by Gaurav Bmotra 09 Jun '20

09 Jun '20

hi how we can implement , that if call got disconnected due to low network and will reconnect .. pl help me -- *Regards:* Gaurav Kumar

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Does DMQ auto-discover cluster members + how?
by George Diamantopoulos 09 Jun '20

09 Jun '20

Hello all (again), When running dmq.list_nodes on one of my production kamailio servers, I get several entries corresponding to staging instances of kamailio. These staging instances are located on separate broadcast domains (different VLANs), and none of them are configured in the notification_address parameter for the dmq module. How do they end up there, and how do I get rid of them? Thanks! George

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