- sr-users - kamailio.org

Dialog Tracking with Inconsistent RR
by Charles Phillips 20 Jan '21

20 Jan '21

I am working to track concurrent channels with the dialog module, but I have run in to an issue. Due to the nature of my required configuration, I have a need to strip the record route headers from packets that are sent to the upstream proxy. This leaves no way for the dialog module to track the state of the call. I was hoping that DID_FALLBACK or DID_NONE modes would track by call id or some other constant, but it does not appear to. So, before I start down the path of using htable to store and reinsert the did when a packet returns, is there something I am missing with dialog? Is there some more efficient way to do this? - Charles

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Error: invalid json doc in command "kamctl ping" kamailio 5.4.3
by Julia Bo 20 Jan '21

20 Jan '21

Hello, Kamailio 5.4.3 has been installed from source. "kamctl ping sip:10.10.10.10:5060" is sent from the command line. Errors: Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_mod.c:1352]: jsonrpc_exec_ex(): invalid json doc [[{"jsonrpc": "2.0", "method": "tm.t_uac", "], "reply_name": "kamailio_receiver_85791", "id": 85791}]].", "From:sip:daemon@KAM-TEST Jan 12 15:59:24 KAM-TEST /usr/local/kam/kamailio[66587]: ERROR: jsonrpcs [jsonrpcs_fifo.c:461]: jsonrpc_fifo_server(): failed to execute the json document from fifo stream This command is working in 4.4. Has anyone tried using kamctl ping in 5.4? Is it a bug in 5.4.3 ? Thank you, Julia.

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Header Manipulation with returned REST data
by Mark Lawrence 19 Jan '21

19 Jan '21

Hi All, Sorry for the intrusion and asking directly, but there’s so much information out there that I would spend years looking for the answer myself and spending equal amount of time in a lab playing. I do like the last idea but time is a little short. I’m thinking of using Kamailio to overcome a few downfalls in the main SBC/PBX commercial product space that I’ve typically deployed and wanted to ask if it is possible to use as a stateful proxy and carry out the following header manipulation type scenario with a bit more dynamassium: Use the FROM and TO header to create two value pairs that would be sent via a REST request to an external application, which will return two corresponding values to (a) supply a different value to update the original FROM header and (b) a full PAI sting to add on to the outgoing message. If you have experience of this or if it’s a nonstarter I’d love to know. There’s so many modules and great stuff that you have in your toy chest it is amazing and kudos to all you developers being so inventive out there: it leaves me to think I’ve been deploying SIP wrong all these years. Thanks and keep safe and well, Mark

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Count the rows in a xavp from script
by Chaigneau, Nicolas 19 Jan '21

19 Jan '21

Hello, I'm looking for a way to retrieve the number of rows stored in a xavp. (Something better than manually iterating over the xavp) Basically, I'd like to call xavp_count from the script. Is that possible ? 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.

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Kamailio High Availability questions
by Andriano 17 Jan '21

17 Jan '21

Hello have some questions about Kamailio HA + loadbalancers (dispatcher modules) 1. is posible to make HA for RTP media? maybe RTPEngine +redis? is not possible to do this with RTProxy? where is some examples? 2. What do Kamailio db cluster module? it can replicate every DB query record like normal DB cluster? or better is to use native MySQL cluster mode? 3. How is working dispatcher modules in HA? how call is handled and htable in memory between loadbalancers endpoint nodes etc... IS it posible to do only with DMQ ? Just need understand how to do basic Kamailio setup with HA and load balancers in general, for example if will be 2 phisical servers, in every server is virtualisation and every server has 2 VPS. 1 VPS for frontend Kamailio loadbanancer, 1 VPS backend Kamailio app with accounts, routing logic, LCR, RTP media app etc... between load balancers i know maybe is possible to do floating ip, dns round robinetc... and if one VPs server down, call will be dropet in every situation? or call will not be droped and how to prevent it. Just need to know in general schema and architecture how to do this real HA in Kamailio and avoind sinlge point of failure and where to read and learn it ho to make HA in real. In the internet no many docs about it, many conference video how good it is, but no good examples with config, i did not founded, so will be very glad for any help in these newbie questions ;) with single kamailio is many answers, but is realy difficult to understand how to do voip HA with Kamailio without single point of failure. Thanks for any answers...

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http_client module not found
by Joli Martinez 15 Jan '21

15 Jan '21

Hello, I am running kamailio 5.3.8 on Debian 9. Kamailio is installed via apt package manager. I am trying to get apiban.org to work and it seems to be having a problem with the http_client module. I have searched for the kamailio modules and I am unable to find anything that resembles http. I have also installed the extra module and no luck. What would be the correct module to install? kamailio: ERROR: <core> [core/sr_module.c:500]: load_module(): could not find module <http_client> in </usr/lib/x86_64-linux-gnu/kamailio/modules/>

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Stubs for KEMI Javascript
by Nicki Otte 15 Jan '21

15 Jan '21

Hi everyone. I wanted all of you to know, that I have made JavaScript stubs, for autocompletion in IDE’s, for those of you that use KEMI JavaScript. Specifically tested on Jetbrains PHPStorm/Webstorm. The stubs are available for free at my GitHub page. The link is https://github.com/captainDuckay/Kamailio-javascript-stubs My main language isn't typescript or JavaScript, and this is one of my first public releases. With that said though, I am open for critics, or even guidance. I hope it's usable for you. // Nicki Bo Otte

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To header being malformatted.
by AL S 14 Jan '21

14 Jan '21

Hi, We noticed an issue with the way the Kamailio is handling certain messages being sent to them, specifically, the To header being relayed from Kamailio to the next hop is malformatted. Any assistance is greatly appreciated INVITE received by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+12053140346@72.31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 INVITE forwarded by Kamailio: Request: Invite Request URI: sip:2053140346;isub=33343@72.31.12.137:5060;transport=udp From: <sip:+13037819601@71.44.15.84;user=phone>;tag=gK00a85927 To: <sip:+120531403Oh.lc".3/+0]1..31.12.137>;tag=SDnpoa601-1c711123926 Call-ID: SDnpoa601-f1bc387aa5a486c1dd943d1b53ae8e6a-8q7og02 CSeq: 860668 INVITE Contact: <sip:+13037819601@107.14.122.29:5060> SDP IP: 0.0.0.0 SDP Port: 43126 Another example in BYE: BYE received by Kamailio: To: "WIRELESS CALLER" <sip:+18131111111@71.44.13.84>;tag=gK0a788ca6 BYE forwarded by Kamailio: To: "WIRELESS CALLER" <sip:+18131112222A.++.+,-94 Thanks. AL

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Source Port on TLS OPTIONS from Dispatcher
by Charles Phillips 14 Jan '21

14 Jan '21

Hello all. As they say in radio, “long time listener, first time caller” Anyway, I am having trouble getting past the following road block and any help would be greatly appreciated. Kamailio version is 5.4.3 When attempting to use dispatcher to send OPTIONS packets to several TLS destinations, the packets are leaving the Kamailio server on random ports. This is a problem because the servers I am sending the OPTIONS to (MS Teams) are enforcing rport so the responses are returned to a port that Kamailio is not listening on. I have tried to force the socket in the event route (relevant parts of snippet below) but it does not appear to help. I should also mention that I am not behind NAT and the TLS socket is specified in the dispatcher attrs. event_route[tm:local-request] { sip_trace(); $fs = “tls:**ip-address**:5061”; } I have used Kamailio as a TLS server for many projects, but this is my first time as a client. I am sure I am missing something. - Charles

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BYE and TCP
by Kjeld Flarup 13 Jan '21

13 Jan '21

Hello I have a Kamailio running behind NAT, which sends calls to a VOIP service provider. I have setup the Kamalio to listen on port 5071, and also setup a port forward in the router. Now the problem is that with TCP, 5071 is not used for the dialog, but a new port is chosen everytime. This means that when the mobile phone called hands up, I never sees the BYE, because BYE is a new dialog. To which port is the server supposed to send the BYE, and what field tells the server this. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

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handling rejected REFER messages
by Jaime Gajardo 13 Jan '21

13 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 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_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux

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handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environment. behind the nat we have 2 identical nodes using REFER for transfer call to another destiny. On the other NAt side we have our public carrier. When some node sends to kamailio the REFER message and kamailio reroutes this message to the carrier. The call is rejected due company policies. We want to know how we can convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 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_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

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handling rejected REFER messages
by Jaime Gajardo 12 Jan '21

12 Jan '21

Actually we have installed kamailio 5.4.2 and rtpengine like load balancer in a NAT environmet. behind the nat we have 2 identycal nodes using REFER for transfer call to another destiny. in the other NAt side we have our public carrier. When some node send to kamailio the REFER message and kamailio reroute this message to the carrier. The call is rejected due company policies. We want know how can we do for convert or modify the REFER message into INVITE for finally transfer the call. kamailio -v: version: kamailio 5.4.2 (x86_64/linux) 4fdc21 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_BLACKLIST, HAVE_RESOLV_RES 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: 4fdc21 compiled on 20:43:45 Dec 1 2020 with unknown Uname -a: Linux proxy 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 x86_64 x86_64 GNU/Linux rtpengine -v: Version: 9.2.0.0+0~mr9.2.0.0 git-master-aa5e0244

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Re: [SR-Users] Using websockets with kamailio 5.4
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Daniel - Google is your friend. I use it often. https://github.com/havfo/WEBRTC-to-SIP

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TSILO - transaction is not suspended
by Jeremy McNamara 11 Jan '21

11 Jan '21

Hi Folks - I am attempting to set up a mobile push configuration, by following the published example(s). If we have more than one endpoint registered, this configuration works as expected (via the ts_store() path below). If we do not have another registered endpoint the transaction seems to be suspended but then cannot be resumed (t_suspend) after REGISTER arrives. INFO: <script>: [R-LOG]: INVITE from IP:[24.XXX.XXX.XXX:62665] [sip:jeremy2@stage][<null>] NFO: <script>: [R-SUSPEND]: Suspended transaction [41359:415220139] INFO: <script>: [R-SUSPEND]: fU:[jeremy] rU:[jeremy2] tU:[jeremy2] INFO: <script>: [R-SUSPEND]: fU:[jeremy] => tU:[jeremy2] INFO: <script>: [R-SUSPEND]: vtp:[41359:415220139] INFO: <script>: [R-SENDPUSH]: Push sent [(55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78][com][hq01gt5utegi18ipkh1b] [Node-Push]: About to push: 55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78 INFO: <script>: R-REGISTER]: jeremy2 registered INFO: <script>: [R-REGISTER]: Contact: [<sip:ilyartjr@qjwhdafxlbtk.invalid ;transport=ws>;expires=10;app-id=stage;pn-voip-tok=55516cc061038f1283104abb39072a59107b7d8e13a1fd166b1704a315d07f78] INFO: <script>: [R-REGISTER]: App id: stage INFO: <script>: [R-REGISTER]: Expires: 10 INFO: <script>: [R-JOIN]: jeremy2 INFO: <script>: [R-JOIN]: Resuming trasaction [41359:415220139] jeremy2 (41359:415220139) WARNING: tm [t_suspend.c:197]: t_continue_helper(): transaction is not suspended [41359:415220139] WARNING: tmx [tmx_mod.c:847]: w_t_continue(): resuming the processing of transaction [41359:415220139] failed Relvant kamailio.cfg [Note: I did notice we had to switch things up a bit if calls come from a PSTN number via SBC] modparam("htable", "htable", "vtp=>size=10;autoexpire=300") ... if (!lookup("location")) { send_reply("100", "Suspending"); route(SUSPEND); // *testing this path here * } else { route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-LOCATION]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); } ... route[SUSPEND] { if (!t_suspend()) { xlog("L_ERROR", "[R-SUSPEND]: Failed to suspend transaction [$T(id_index):$T(id_label)]\n"); send_reply("501", "Unknown Destinaton"); exit; } xlog("L_INFO", "[R-SUSPEND]: Suspended transaction [$T(id_index):$T(id_label)]\n"); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] rU:[$rU] tU:[$tU]\n"); if (isbflagset(FLB_PSTN)) { $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => rU:[$rU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$rU)]\n"); } else { $sht(vtp=>join::$tU) = "" + $T(id_index) + ":" + $T(id_label); xlog("L_INFO", "[R-SUSPEND]: fU:[$fU] => tU:[$tU]\n"); xlog("L_INFO", "[R-SUSPEND]: vtp:[$sht(vtp=>join::$tU)]\n"); } } route[REGISTER] { if (!is_method("REGISTER")) { return; } if (isflagset(FLT_NATS)) { setbflag(FLB_NATB); } if (!save("location")) { sl_reply_error(); } route(JOIN); } # Append branches or resume the transaction route[JOIN] { xlog("L_INFO","[R-JOIN]: $tU\n"); $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) { xlog("L_INFO","[R-JOIN]: Appending transaction ($tu)\n"); ts_append("location", "$tu"); exit; } xlog("L_INFO","[R-JOIN]: No INVITE to resume\n"); exit; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); xlog("L_INFO", "[R-JOIN]: Resuming trasaction [$var(id_index):$var(id_label)] $tU ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "RESUME"); exit; } # Lookup and relay after resuming transaction route[RESUME] { xlog("L_INFO", "[R-RESUME]: ru: [$ru] tu: [$tu]\n"); lookup("location"); route(SETUP_ROUTES); route(RELAY); ts_store("$tu"); $sht(vtp=>stored::$tU) = 1; xlog("L_INFO", "[R-RESUME]: Stored transaction [$T(id_index):$T(id_label)] $fU => $tU\n"); exit; } What did I miss?

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Profile name as variable in set_dlg_profile
by Muhammad Sohaib 11 Jan '21

11 Jan '21

Is it possible to insert the current dialog into the profile by using the following way? modparam("dialog", "profiles_no_value", "mytest;xyz") $var(profilename)="xyz"; set_dlg_profile(" $var(profilename) "); Currently it is giving errors and not allowing to use variable instead of static profile name. Or is there any alternative to pass profile name as variable? --- Thanks,

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Unload UAC remote registry
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm on Kamailio 4.4.7 and I'm trying to do uac.reg_refresh over a row non existing on DB to try to unload, but data still persist on Kamailio. How to unload only one registry from UACs remote registrations when delete a record from DB without do a uac.reg_reload? UAC remote registrations is above 2k rows, I don't know if it's much or not. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

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Different responses for domain.dump
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there! I'm getting different responses from Kamailio 4.4.7 when domain.dump exec; From *kamcmd*: I'm getting two domains; .es and .net. Request to jsonrpc-s module: Response from jsonrpc-s module: And now, I'm getting only one domain. What's I'm doing wrong?

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uac.reg_refresh command
by Javier Valencia 11 Jan '21

11 Jan '21

Hello, I'm playing with RPC on my Kamailio 4.4.7, and I getting rare behaivor with *uac.reg_refresh* command. When I run it thru kamcmd like docs describes, this is without *l_uuid* parameter, I'm geting "*error: 500 - Invalid Parameters*" output. And when I run with *l_uuid* parameter, I get no output. But in both cases, Kamalio don't refresh data from DB. But I run this command thru HTTP (JSONRPC-S), both call modes are equivalent. With or without l_uuid parameter geting no output and refresh data from DB. It's a normal behaivor? Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

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list myself values
by Javier Valencia 11 Jan '21

11 Jan '21

Hi, a question from a Kamailio newbie. How can I list or show variable *myself* values? Regards, JV

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UAC module registrations interval
by Javier Valencia 11 Jan '21

11 Jan '21

Hi there, I'm using Kamailio 4.4.7 with UAC remote registrations module. In my database have all rows with expires column value's 3600, and my PBX (asterisk) min, max and default expires to 3600. But I can see how Kamailio it's sending REGISTER each 150segs. UAC module parameters values; modparam("uac", "reg_db_url", "mysql://XXX") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_contact_addr", "XXX") modparam("uac", "reg_timer_interval", 5) modparam("uac", "reg_retry_interval", 60) My uacreg table it's above 2k rows. Regards, JV -- Logo <http://www.voiper.es/> Javier Valencia | CTO Centro de Negocios Martín Buendía Camino de las Cañadas, nº 1C, Portal 1, 2ºG 29649 Mijas (Málaga) # 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759> # www.voiper.es <http://www.voiper.es/> ------------------------------------------------------------------------ AVISO LEGAL ----------- Este mensaje contiene información confidencial destinada para ser leída exclusivamente por el destinatario. Queda prohibida la reproducción, publicación, divulgación, total o parcial del mensaje así como el uso no autorizados por el emisor. En caso de recibir el mensaje por error, se ruega su comunicación al remitente lo antes posible. Por favor, indique inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo por Internet. Las opiniones, conclusiones y demás información incluida en este mensaje que no esté relacionada con asuntos profesionales de SMART RECARGAS, S.L, se entenderá que nunca se ha dado, ni está respaldado por el mismo. Responsable del Tratamiento de Datos ------------------------------------ SMART RECARGAS , SL Camino de las Cañadas, 1C PORTAL 1 2º G 29651 MIJAS COSTA (MALAGA)

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how does rtjson work?
by Syed Shahryar 11 Jan '21

11 Jan '21

My use case is such that I want to be able to send a SIP MESSAGE destined for a single username to multiple users. In order to be able to map that target username to multiple usernames, I tried using rtjson and evapi/http (to generate the routing json), but none of them worked. Is it the right approach? Is there any other way to programmatically generate multiple targets for SIP request? I am familiar with dynamic routing but I need more flexibility. Initially, I tried sending the request to a different username but that didn't work and got the following message in the http async result route. kamailio | 19(25) ERROR: nathelper [nathelper.c:1461]: sdp_1918(): Unable to parse sdp body kamailio | 19(25) NOTICE: <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi msg branch kamailio | 28(34) NOTICE: {2 6661 MESSAGE q6hikd6tumsr1dh35kvj} <core> [core/kemi.c:124]: sr_kemi_core_notice(): In evapi reply branch The message was received on the originally requested to user and also $tu still points to the original user instead of the new user even after executing the following code. KSR.rtjson.init_routes(response); KSR.rtjson.push_routes(); KSR.nathelper.fix_nated_contact(); In http response, I receive the following and don't see any rtjson errors when init_routes is executed on it. {"version":"1.0","routing":"parallel","routes":[{"headers":{"from":{"display":"test_1","uri":"sip: *test_1*@test.com:9908"},"to":{"display":"test_4","uri":"sip:*test_4*@ test.com:9908 "},"extra":""},"branch_flags":8,"fr_timer":5000,"fr_inv_timer":30000}]} Please note that the request was sent to *test_2*(a)test.com. I have tried other combinations of above but nothing works. I have already searched on Google and read relevant articles and blog posts. Thanks -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

2 1

Lost MySQL connections - Error 500
by Duarte Rocha 11 Jan '21

11 Jan '21

Hi all, I've made a post last month regarding losing MySQL connections - https://lists.kamailio.org/pipermail/sr-users/2020-December/111389.html At the time I thought connections were dying as a consequence of low activity and traffic on the proxy. Meanwhile, I've migrated a great number of equipments to the proxy with Registers being refreshed every 10minutes and the problem still persists. In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. My SQL client on the Kamailio machine is mysql-community-client 5.6.50-2.el7. It writes and reads in a remote InnoDB database. This are the logs i get from Kamailio when the problem appears : Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lock wait timeout exceeded; try restarting transaction (1205) Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} <core> [db_query.c:348]: db_do_update(): error while submitting query Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1663]: update_contact_db(): failed to update database Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} registrar [save.c:784]: update_contacts(): failed to update contact Jan 7 09:43:27 sbc_bbt01_active /usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) Originally I had usrloc db_mode on mode 3 - DB-Only Scheme. In order to try to mitigate the issue I changed it to mode 1 - Write-Through scheme but even then I get the same log errors and an "500" error is still sent to the client. I chose this mode since, as far as I can understand it applies changes directly to DB but also uses cache. Please correct me if i'm wrong on that. Has this issue happened with anyone before? Is there a way to mitigate this issue? My only constraint is that I need the database to be always updated since I have an HA setup, and as such, I can't use cache only methods. Best Regards,

2 5

loose_route() - Local Proxy
by Duarte Rocha 08 Jan '21

08 Jan '21

Greetings, I'm using loose_route() from the RR module and i'm having troubles making it use the following exception from the code : "There is only one exception: If the request is out-of-dialog (no to-tag) and there is only one Route: header indicating the local proxy, then the Route: header is removed and the function returns FALSE." My example is a REGISTER without To-TAG which has a Route header with kamailio address. If i use proxy IP on Route header , loose_route() returns false as it should. However, if i use an hostname belonging to the proxy in the route, loose_route() returns true. I have hostnames and local ips defined in the "DOMAIN" table but it doesn't seem to be working. Which other places can local hostnames and ips be configured in order to be seen as local to loose_route() ? Best Regards

4 4

Contact header not removed properly Getting some junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my scenario call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org/> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060/>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.xx.xx.xxx 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.xx.xx.xxx 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

about Contact Header Junks
by Devang Dhandhalya 07 Jan '21

07 Jan '21

in my requirement call coming from sbc to kamailio proxy and we route to pbx server in that when kamailio proxy get the call at that time i am removing Contact header using remove_hf("Contact") and adding contact Header using append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i am getting modified Contact header but old contact header not removed properly some junks are remain in the signaling scenario : SBC ---------> Kamailio proxy ------> PBX before using remove_hf("Contact") INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 70 Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP> To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP> From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org ;transport=UDP>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp Proxy-Authorization: Digest username="user1",realm="domain.org <http://opensips.org> ",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org <sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5 82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5 User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 243 v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 10.212.xxx.xxx t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 97 101 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv after using remove-hf("Contact") function removing contact header but some junks are left and that eats next To header line and breaks Invite packet INVITE sip:user2@domain.org:;transport=UDP SIP/2.0 Record-Route: <sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA-> Via: SIP/2.0/UDP 172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0 Via: SIP/2.0/UDP 10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e Max-Forwards: 69 sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org <sip%3Adevang3032(a)opensips.org>> From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646 Call-ID: t9Ff-fCrin_VN1IO9Xf1mA.. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: Z 3.15.40006 rv2.8.20 Allow-Events: presence, kpml, talk Content-Length: 534 Contact: <sip:user2@172.xx.xx.xxx:5060 <http://sip:dppatel@172.16.16.163:5060>> v=0 o=Z 0 0 IN IP4 10.212.xxx.xxx s=Z c=IN IP4 172.xx.xx.xxx t=0 0 m=audio 11210 RTP/AVP 3 110 8 0 97 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:97 mode=30 a=fmtp:101 0-16 a=sendrecv a=rtcp:11211 a=ice-ufrag:lzjregPe a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host As you can see Contact header is gone but sip:1234@172.xx.xx.x:39931;transport=UDP part left before To header . and its breake invite packet . Any suggestion will be highly appreciated. -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

2 1

Dispatcher SNI Questions
by Tim Chubb 06 Jan '21

06 Jan '21

Hi All Is it possible to explicitly define the tls cert to be used per destination by dispatcher? I'm attempting to integrate with a service that requires domain name presented to match that of the cert. Where the use of dispatcher is probably not as intended anyway as it targets the same set of destinations multiple times (primary and secondary endpoints for the service im integrating with), with multiple distinct set id's (basically a set per customer for multi tenancy) which expects a sip ping per account to verify that connection is alive and healthy. #Example Dispatcher list #Account 1 1 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061 1 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061; #Account 2 2 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; 2 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061; Currently I'm running a solution based around certs with multiple SAN (subject alternative name) defined, but this is a pain to administrate, and not as scalable as I would like. I want to be able to define multiple client:any tls profiles and explicitly send via that profile. Its easy enough to do that in config using the tls xavps to define a server name and/or id. But for the sip options ping by dispatcher to work you need to specify the server name or id before the tm:local-request event route fire's. I've resorted to hacking away at dispatcher to see what would happen if I set the tls server name or id before the OPTIONS message is sent, and it works great for the first tls client profile matched, but any others you define re-use the initial tcp connection so reuses the first connections tls config, thus presenting the wrong cert (I'm probably wrong but logs show only one tls complete_init() related to dispatcher pings and the second set's pings skips all the tls init logging so suggests some thing is cached/reused, and logging on the service indicates cert name mismatch with set1s cert being offered) So I've hit a bit of a dead end, is there a way to force a new tcp connection per server name/id I fear outbound SNI might be a bit of a can of worms, for it to work nicely with dispatcher as is, would be an extension of the proto:host:port format to include a name or id, maybe something like: tls(name=sub.domain.com):10.0.0.5:5061 But that looks like it would have a lot of wide reaching knock-on effects, so setting it as an attribute like ping_from works fine as I'm currently doing, if I could force a new connection from dispatch.c.... The other thing that's occurred to me is that it might be better to stop trying to adapt a small percentage of dispatcher's functionality and trying to hammer a round peg into a square hole and instead try and create a new module similar to dispatcher in so much that it sends pings (that's all I really need in this case) and use dispatcher as intended with a single set of non-repeated destinations for routing. This would also allow much simpler integration with MS teams (if added ability to set contact address on the background pings) as well as the service im trying to integrate, and a more natural data set to be used as in my example you can see the only difference is the customers domain between each set. How I would imagine it to work would be to create a tls connection per customer domain, then iterate through the connections per destination in pseudo json the data would look like this: { Destinations:[ [1,["a.cloudservice.com","b.cloudservice.com"]], [2,["sip.pstnhub.microsoft.com", "sip2.pstnhub.microsoft.com", "sip3.pstnhub.microsoft.com"] ], Sources:[ {uri:"sip.account1.com",destinationMapping:[1]}, {uri:"sip.account2.com",destinationMapping:[1,2]} ] } And would just run in the background sending pings with SNI support, and if based on dispatcher should make it easy enough adapt things like the destination up and down event routes, and would allow me to use Kamailio much like I would use nginx for web stuff as a reverse proxy and handle TLS offload on my edge gateway proxy. Any thoughts?, im frustratingly close to having something that will do if I could get a new connection to be created, but I fear my use case might be sufficiently different from dispatchers to make a new module a sensible approach.. Regards Tim.

2 1

Pre-loaded routes verification
by Duarte Rocha 04 Jan '21

04 Jan '21

Greetings, I'm doing a migration from a opensips system to kamailio one and i'm trying to replicate its functionalities. The system is a Registrar with a pstn gateway. I noticed that on the opensips version, loose_route() is called on initial requests and it seems to be used to protect against preloaded routes. However, on the default Kamailio configuration file loose_route() is only called on requests in-dialog and the verification mentioned above is not used. Why doesn't Kamailio includes this verification? Is there are any security concerns not using it? Best Regards,

2 1

using kamailio 5.4 with the websocket or xmlhttprequest
by Daniel Hermann N'don 04 Jan '21

04 Jan '21

Hello everyone, I'm a beginner in kamailio technology, and I would like to know how to communicate with kamailio from the web.

3 3

Unable to connect sip with Kamailio
by Rohit Kumar 04 Jan '21

04 Jan '21

Hello everyone, I'm beginner in Kamailio, I want to connect my asterisk Server with Kamailio. Can you please help me in this context. I shall be very thankful to you. Regards, Rohit Kumar On Thu, Dec 31, 2020, 16:31 <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 2. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Antony Stone) > 3. Re: using kamailio 5.4 with the websocket or xmlhttprequest > (Daniel Hermann N'don) > 4. Using a specific log level for "script" ? (Chaigneau, Nicolas) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 30 Dec 2020 14:15:19 +0000 > From: Daniel Hermann N'don <danielhermann.ndon(a)outlook.com> > To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> > Subject: [SR-Users] using kamailio 5.4 with the websocket or > xmlhttprequest > Message-ID: > < > MWHPR05MB291167705F86195717CA7E3181D70(a)MWHPR05MB2911.namprd05.prod.outlook.com > > > > Content-Type: text/plain; charset="iso-8859-1" > > Hello everyone, I'm a beginner in kamailio technology, and I would like to > know how to communicate with kamailio from the web. >

2 1

Using a specific log level for "script" ?
by Chaigneau, Nicolas 04 Jan '21

04 Jan '21

Hello, Is it possible to set a specific log level for the Kamailio script ? I've seen that the < debugger > module allows to do that with modules or < core > : https://kamailio.org/docs/modules/5.4.x/modules/debugger.html#dbg.p.mod_lev… So I've tried to do : modparam("debugger", "mod_level_mode", 1) modparam("debugger", "mod_hash_size", 5) modparam("debugger", "mod_level", "my_module=3") // this works modparam("debugger", "mod_level", "core=3") // also works modparam("debugger", "mod_level", "script=3") // nope... But it doesn't seem to work for < script >... :/ 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.

2 4

[Presence] publ_cache=2 usage
by Denys Pozniak 30 Dec '20

30 Dec '20

Hello! Could somebody explain how the parameters *publ_cache=2* and *subs_db_mode* depend on each other? Do I understand correctly that only the following modes of *subs_db_mode* are allowed 0,1,2? *subs_db_mode:0 - This disables database completely. Only memory will be used. Subscriptions will not survive restart. Use this value if you need a really fast presence module and subscription persistence is not necessary or is provided by other means.1 - Write-Through scheme. Subscriptions are updated synchronously in database and in memory(used for read operations). Use this scheme if speed is not top priority, but it's important that no subscriptions will be lost during crash or reboot or if you have an external application that reads the state of the subscriptions from database and they need to be updated synchronously.2 - Write-Back scheme. This is a combination of previous two schemes. All changes are made to memory and database synchronization is done in the timer. The timer deletes all expired contacts and flushes all modified or new subscriptions to database. Use this scheme if you encounter high-load peaks and want them to process as fast as possible. Latency of this mode is much lower than latency of mode 1, but slightly higher than latency of mode 0. To control the interval at which data is flushed to database, set the db_update_period parameter.3 - DB-Only scheme. No memory cache is kept, all operations being directly performed with the database. The timer deletes all expired subscriptions from database. The mode is useful if you configure more servers sharing the same DB without any replication at SIP level. The mode may be slower due the high number of DB operation.* -- BR, Denys Pozniak

1 0

Presence - issue with NOTIFY "terminated" state
by Zhan Bazarov 29 Dec '20

29 Dec '20

Hello everyone! I faced some unfortunately issue with presence module. Let me describe setup scheme a bit: I'm publishing dialog states via pua_dialoginfo from one kamailio server(for calls proccessing itself) to another one kamailio(standalone kamailio just for handling presence,blf,mwi etc) So, about issue, let's say we have device A which has outgoing subscription to extension which belongs to device B and device C(both devices has same extension but different usernames, webrtc+sip phone, effectively its the same Phone, I'm using pubruri_avps to change entities from usernames to extension) . And, in some cases,for examle, when thirtparty is making call to this extension(B,C devices) and making CANCEL after couple seconds - I see the PUBLISH from callserver to presence server with state "terminated', I see 200ok response.. but presence server isn't sending NOTIFY to device A about state "terminated", everything is good with "trying" and "early" states... That problem is happening 1 of 50 tries.. I've tried a lot of things to fix that - still are not succeed. p.s. everything is good from logs/sip point of view... And also, when I disable the webrtc device - I'm not able to reproduce that issue anymore. Looks like something wrong with case, when you have multiple different username behind the same presentity. Maybe someone had same issue and may help me. Thanks! -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Kamailio as a presence watcher
by Jukka Heikkilä 29 Dec '20

29 Dec '20

Hello, I'm working with IMS network and VoLTE related services. My problem is that I would like to use Kamailio as a presence watcher but I haven't managed to make it happen yet. The approach would be that once Kamailio gets 3rd party registration from S-CSCF the Kamailio would send SUBSCRIBE back to S-CSCF to subscribe the presence of subscription. Once a subscription place or get calls the status of subscription would be updated to Kamailio and to the database where it can be fetched. I have checked presence and pua modules, but haven't found a way to make it work as expected. Is there way to do that or is this totally incompatible approach to do that? Best Regards, Jukka Heikkilä jutsco(a)gmail.com

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How to implement a self-sustained Presence server
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration? That is I want the Presence module to be fed status updates as Kamailio is routing calls and handling registrations. So when a user registers, his phone is ringing, he is on the phone, etc.., the Presence module gets notified and can update its subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

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Kamailio suddenly stops
by Javier Valencia 28 Dec '20

28 Dec '20

Hi there! When I start my webrtc kamailio suddenly stops. I can see this on syslog: Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c2ab8), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c3138), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring ... omit vervose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f55756c1f38), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_free(323) - ignoring ...omit verbose ... Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557569f250), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_realloc(299) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f557565cf60), called from tls: tls_init.c: ser_free(323), first free tls: tls_init.c: ser_malloc(293) - ignoring Dec 28 16:08:53 webrtc /usr/sbin/kamailio[803]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized OS info: root@webrtc:/etc/kamailio# lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 10 (buster) Release: 10 Codename: buster root@webrtc:/etc/kamailio# free -m total used free shared buff/cache available Mem: 16041 134 15551 108 356 15524 Swap: 974 0 974 Kamailio version: version: kamailio 5.4.3 (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_BLACKLIST, 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 8.3.0 Kamailio params: SHM_MEMORY=512 PKG_MEMORY=32 SSL info: root@webrtc:/etc/kamailio# apt show openssl Package: openssl Version: 1.1.1d-0+deb10u4 Priority: optional Section: utils Maintainer: Debian OpenSSL Team <pkg-openssl-devel(a)lists.alioth.debian.org> Installed-Size: 1.495 kB Depends: libc6 (>= 2.15), libssl1.1 (>= 1.1.1) Suggests: ca-certificates Homepage: https://www.openssl.org/ Download-Size: 843 kB APT-Manual-Installed: no APT-Sources: http://security.debian.org/debian-security buster/updates/main amd64 Packages Description: juego de herramientas Secure Sockets Layer - utilidad criptográfica Este paquete forma parte de la implementación del proyecto OpenSSL de los protocolos criptográficos SSL y TLS para la comunicación segura a través de Internet. . Contiene el binario de propósito general para la línea de órdenes /usr/bin/openssl, útil para operaciones criptográficas como: * creación de parámetros clave RSA, DH y DSA; * creación de certificados X.509, CSR y CRL; * calcular resúmenes de mensajes; * cifrado y descifrado con cifras; * probar clientes y servidores SSL/TLS; * gestión de correo firmado o cifrado S/MIME. Should I use another version of openssl, or maybe use gnutls? Regards, Javier Valencia

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Self-sustained Presence server.
by Michel Pelletier 28 Dec '20

28 Dec '20

Hi, What is the recommended practice for implementing a self-sustained Presence server from my current Kamailio configuration. That is I want the Presence module to be fed status updates from the same Kamailio that is routing the calls. So when a user registers, his phone is ringing, he is busy, on the phone, etc.., the Presence module gets notified and can update subscribers accordingly. Must I add snippets of codes to all of these events and generate local Publish messages, or is there a better way? I'm using the latest version 5.4.x. Regards, Michel Pelletier

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Read $fd from mysql?
by Mihai Cezar 27 Dec '20

27 Dec '20

Hello, I am redirecting the agents/reps to their asterisk server based on the hostname which they connect, and, i wonder if it's possible to read from db instead of the file cause it's getting big ;) # Dispatch requests route[DISPATCH] { switch ($fd){ case "oro.streamlinepbx.nl": if (!ds_select_dst("1", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; case "oro3.streamlinepbx.nl": if (!ds_select_dst("2", "4")) { send_reply(503, "Service Unavailable $fd"); exit; } break; ............ default: log("unknow destination?"); send_reply(503, "No service defined"); xlog("--- SCRIPT: going to <$ru> via <$du> ...Exiting"); exit; } xlog("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } Thanks,

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Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 27 Dec '20

27 Dec '20

A year that was very tough for the entire world is getting close to its end, no doubt that was the most challenging in our life so far for many of us, however, with a distributed, friendly and collaborative community, Kamailio project continued to develop consistently, bringing out a new major release. I trust it helped people and companies world wide to to connect with the dear ones and build a better way of living. The season holidays are ahead, time to rest a bit, spend time with family and friends, being online or face to face. As usual, I take this chance to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2020 another great year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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app_lua
by rob.van.den.bulk＠gmail.com 26 Dec '20

26 Dec '20

Hallo, I try to install the module app_lua in Kamailio 5.3. ls: cannot access '/usr/lib/liblua*': No such file or directory ls: cannot access '/usr/lib/liblua*': No such file or directory CC (gcc) [M app_lua.so] app_lua_mod.o In file included from app_lua_mod.c:32: app_lua_api.h:25:10: fatal error: lua.h: No such file or directory #include <lua.h> ^~~~~~~ compilation terminated. make[2]: *** [../../Makefile.rules:100: app_lua_mod.o] Error 1 make[1]: *** [Makefile:511: modules] Error 1 make[1]: Leaving directory '/usr/local/src/kamailio-5.3/kamailio/src' make: *** [Makefile:34: all] Error 2 Any help appreciated Greeting Rob van den Bulk

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Fwd: [CFP] FOSDEM 2021, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 23 Dec '20

23 Dec '20

FOSDEM - Real Time Communications devroom CfP ============================================= NOTE: we have extended the deadline but we will give preference to people who submitted earlier when we set the time for each talk. Please submit your proposal ASAP so that you can get your preferred time slot. Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2021 takes place 6-7 February 2021 and for the very first time, it will be online. This document contains information about: - Real-Time Communications developer room (devroom) - speaking opportunities - volunteering New rules for the online edition -------------------------------- This year FOSDEM will be fully online instead of being held in Brussels, here are the most important things to know about this (quite significant) change: - The reference time will be Brussels local time (CET) - Talks will be pre-recorded in advance, and streamed during the event - The Q/A session will be live - A facility will be provided for people watching to chat between themselves - A facility will be provided for people watching to submit questions Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers who can help manage the scheduling and live Q&A sessions.** The devroom is only on Saturday, 6th of February 2021. To discuss the devroom, volunteer or ask questions, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Key dates - 20th Dec: Submission deadline (extended to 8 January) - 24th Dec: Announcement of selected talks - 15th Jan: Presentations upload deadline - 6th & 7th Feb: Conference dates (online) - 6th Feb: RTC devroom date (online) ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 20th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM21/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. This year this is more true than ever, being able to record your presentation offline without an audience in front can greatly help build up one's confidence! ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. ### Recording help The devroom organization is able to provide help with recording your session. The recording would be performed at a scheduled time with one of us, so you won't be alone giving your presentation. Minimal edits will be possible, but the ideal plan is to record it in one shot. Thanks Dan Jenkins for providing us with the means to do this! Volunteers needed ----------------- To make the devroom run successfully, we are looking for volunteers. This year many things be done for the first time, so all the help we can get is more than welcome. Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

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MID-Registrar Example
by moisesynfante 23 Dec '20

23 Dec '20

Hello, I'm a newbie when it comes to Kamailio and I would like to know if there's a working example of a Mid-Registrar implementation. I read on other posts that the UAC module can be used to achieve this but, honestly, I don't understand how. If any of you could provide some path I could follow I would be very thankful. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

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Re: [SR-Users] How to use port number using $env in listen
by Syed Shahryar 22 Dec '20

22 Dec '20

Any updates on this? On Tue, Dec 15, 2020 at 8:13 PM Syed Shahryar <syedshahryar(a)gmail.com> wrote: > I am new to Kamailio so may be doing something wrong. I looked at > https://www.kamailio.org/wiki/cookbooks/5.4.x/core. > > I tried the following combinations but getting errors in all cases: > > #!subst "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdef "/EVAPI_PORT/$env(EVAPI_PORT)/" > #!substdefs "/EVAPI_PORT/$env(EVAPI_PORT)/" > > *Errors:* > 0(6) ERROR: <core> [core/pvapi.c:924]: pv_parse_spec2(): error searching > pvar "env" > 0(6) ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char > [E/69] in [$env(EVAPI_PORT)/] at [5 (5)] > 0(6) ERROR: <core> [core/re.c:170]: parse_repl(): bad specifier in > replace part /$env(EVAPI_PORT)/ > 0(6) ERROR: <core> [core/ppcfg.c:69]: pp_subst_add(): bad subst > expression: /EVAPI_PORT/$env(EVAPI_PORT)/ > > At a later step, I would like to do: > #!define EVAPI_LISTEN "0.0.0.0:EVAPI_PORT" > > And use EVAP_LISTEN in mod_param for the evapi module. > > And something similar to above for TLS settings. > > > On Tue, Dec 15, 2020 at 12:41 PM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> as written on the tracker, look at using #!subst or #!substdef (see core >> cookbook), they should be able to get the substitution value from >> environment using $env(NAME). >> >> The master branch introduced #!defenv as a (simpler) alternative. >> >> Cheers, >> Daniel >> On 12.12.20 05:33, Syed Shahryar wrote: >> >> I tried many combinations of the following but I keep getting syntax >> errors: >> >> listen=tls:TLS_IP:$env(TLS_PORT) advertise PUBLIC_IP:$env(TLS_PORT) >> >> >> TLS_IP and PUBLIC_IP are constants defined at the top using #!define. >> And, TLS_PORT is supposed to be the env variable. >> >> >> -- >> -- >> This message contains confidential information and is intended only for >> the individual named. If you are not the named addressee, you should not >> disseminate, distribute or copy this email. Please notify the sender >> immediately by email if you have received this email by mistake and delete >> this email from your system. Email transmission cannot be guaranteed to be >> secure or error-free, as information could be intercepted, corrupted, lost, >> destroyed, arrive late or incomplete, or contain viruses. The sender, >> therefore, does not accept liability for any errors or omissions in the >> contents of this message which arise as a result of email transmission. If >> verification is required, please request a hard-copy version. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Funding: https://www.paypal.me/dcmierla >> >> > > -- > -- > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee, you should not > disseminate, distribute or copy this email. Please notify the sender > immediately by email if you have received this email by mistake and delete > this email from your system. Email transmission cannot be guaranteed to be > secure or error-free, as information could be intercepted, corrupted, lost, > destroyed, arrive late or incomplete, or contain viruses. The sender, > therefore, does not accept liability for any errors or omissions in the > contents of this message which arise as a result of email transmission. If > verification is required, please request a hard-copy version. > -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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ndb_redis with keydb master/master
by David VILLAUME 22 Dec '20

22 Dec '20

Hello, I consider using keydb-server in an active/action setup without sentinel. Do you know if there is a way to declare both hosts in ndb_redis with the same server name (target use : topos) ? Regards, David

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Re: [SR-Users] DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 22 Dec '20

22 Dec '20

Apologies for not having included the sr-users in the previous email. Anyhow, the solution suggested by Social Boh indeed worked, as a side effect of which we would not run into the issue described. *NOTE:* There was a peculiar configuration in keepalived with keeping both the nodes as BACKUP and setting the 'nopreempt' flag only in the actual MASTER(one with higher 'priority') side to make the keepalived thing to work properly. However, I was wondering if someone could confirm if this issue has been fixed inside the source code itself as I am using an older release v5.3.2 Regards, Harneet Singh On Thu, Dec 17, 2020 at 9:52 PM harneet singh <hbilling(a)gmail.com> wrote: > Thanks Social Boh, > > I will investigate this possibility and if it does work in our case, that > should circumvent the issue. > In addition, is there a fix known to the SR-Users which could be available > inside the Dispatcher module to cause syncing just like the syncing is > there in the Dialog module via DMQ. I wanted to check this as I am using > v5.3.2 and unaware if new releases could have fixed this issue? > > Regards, > Harneet Singh > > On Wed, Dec 16, 2020 at 10:46 PM Social Boh <social(a)bohboh.info> wrote: > >> Maybe a option is leave the secondary like primary when primary go up >> newly. >> >> Example: Primary go down, Secondary is the new Primary. Primary came back >> up but Secondary still is the VIP and still process the calls. >> >> Only if Secondary go down, primary is the new VIP. >> >> The keepalived parameter for this behaviour is: >> >> *nopreempt* >> >> --- >> I'm SoCIaL, MayBe >> >> El 16/12/2020 a las 11:57 a. m., harneet singh escribió: >> >> Hi All, >> >> I am using Kamailio in HA mode with Keepalived providing the VIP(Virtual >> IP) functionality, and have run into a rather peculiar issue. >> >> *Setup:* >> >> Caller ------------ KamailioVIP(*Primary *and *Secondary* >> Kamailio)------- Callee >> >> HA provided by Keepalived >> DMQ is used between the Primary and Secondary Kamailio for dialog sync >> >> *Issue Seen:* >> >> *Suppose the Primary Kamailio has been brought down* and the Secondary >> one is actually active and tied to the VIP. >> >> When a call is fired from the Caller, it traverses through the Secondary >> Kamailio and lands onto the callee. The dialogs are updated properly. At >> this point, the Primary Kamailio is brought up, and dialog state is synced >> due to the DMQ module. >> The Keepalived will now attach the VIP to the Primary Instance. If the >> caller hangs up the phone at this point, the BYE sip message traverses >> through the Primary Kamailio instance to the callee and the call is >> cleared, but there are two issues here. >> >> 1. The Primary Kamailio throws an error in *ds_load_remove() *saying >> that it cannot find the load for the specific call id. This is apparently >> due to the fact that the dispatcher data is not synced between the two >> modules but dialog data is. So dialog wise, things are fine. Can this be >> fixed somehow? >> >> 2. The above is still as grave a problem as the dialogs are cleared. >> BUT, if we check the '*kamcmd dispatcher.list*' on the *SECONDARY* >> kamailio even well after the call is cleared, the *DLGLOAD* shows 1. >> Since we are using the *Call Load based distribution* for the >> dispatching, this is effectively one call stuck on the dispatcher, which >> leads to resource leak. >> >> Is this a known issue, and if so, do we have a way to circumvent this? >> >> Regards, >> Harneet Singh >> -- >> "Once you eliminate the impossible, whatever remains, no matter how >> improbable, must be the truth" - Sir Arthur Conan Doyle >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > -- > "Once you eliminate the impossible, whatever remains, no matter how > improbable, must be the truth" - Sir Arthur Conan Doyle > -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

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Kamailio v5.3.8 released
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, Kamailio SIP Server v5.3.8 stable has been released. This is a maintenance release of the previous stable branch (5.3), that includes fixes since the release of v5.3.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are recommended to be upgraded to v5.3.8. For more details about version 5.3.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/12/kamailio-v5-3-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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MSTeams Kamailio with PBX Hold Refer issue
by me.projects＠yahoo.com 21 Dec '20

21 Dec '20

Hi, I have been trying to integrate MSTeams with PBX Through Kamailio. Calls are working now. Hold has Issues. MSTeams uses REFER to manage this all with NOTIFY. I am sending t_send_reply("202", "Accepted"); when receive Refer now I need to send NOTIFY I think I can use t_uac_send() NOTIFY I am unable to send t_uac_send properly on when recertifications refer from MSTeams. Any one with the experience on this ? Thanks,

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Trying to use $sel(contact.uri.params)
by Chaigneau, Nicolas 21 Dec '20

21 Dec '20

Hello, I'd like to use a < select > as described here: http://www.kamailio.org/wiki/cookbooks/5.4.x/selects#contactparams_s I'm trying to extract a parameter from the Contact URI. What I've tried to do: xlog("L_INFO", "contact.uri: $sel(contact.uri)\n"); # OK xlog("L_INFO", "contact.uri.params: $sel(contact.uri.params)\n"); # null ?? This does not seem to work... see output here : 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri: sip:+33145678901;tgrp=YC031;trunk-context=YC90@10.31.22.2:5099 3(10316) INFO: {1 INVITE r8wEHsAgxNIS5OJL} <script>: contact.uri.params: <null> What am I doing wrong ? Maybe this is the wrong approach for what I'm trying to do ? 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.

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Releasing Kamailio 5.3.8
by Henning Westerholt 21 Dec '20

21 Dec '20

Hello, I am planning to release next week (likely on Monday 21st December) the next minor version out of branch 5.3, version 5.3.8. Backports of fixed issues from git master will be integrated as usual. If you are aware of new issues which were not yet reported, let us know (by creating an issue on the tracker). Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

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send MESSAGE to other sessions of same user
by Syed Shahryar 21 Dec '20

21 Dec '20

What's the best way of sending a message to all other sessions (and the receiver) of the same sender? For example: If user1(a)example.com is logged in to Kamailio from multiple devices and from one of the devices user1 sends a SIP message to user2(a)example.com. I want the message forwarded not only to user2(a)example.com but also to all other user1(a)example.com sessions. What's the easiest way to achieve this? -- -- This message contains confidential information and is intended only for the individual named. If you are not the named addressee, you should not disseminate, distribute or copy this email. Please notify the sender immediately by email if you have received this email by mistake and delete this email from your system. Email transmission cannot be guaranteed to be secure or error-free, as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender, therefore, does not accept liability for any errors or omissions in the contents of this message which arise as a result of email transmission. If verification is required, please request a hard-copy version.

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