- 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.

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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,

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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

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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.

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