- sr-users - kamailio.org

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

2 2

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

4 10

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

2 1

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

1 0

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

1 0

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

1 0

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?

4 6

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,

4 6

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?

2 2

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)

2 3

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

2 2

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)

2 1

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

1 0

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

1 0

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

2 1

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

1 0

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

5 4

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

2 3

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.

2 2

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

1 0

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,

2 1

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.

2 2

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/>

1 1

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.

1 0

Kamailio not updating CSeq after uac_auth() for ACK
by Terry Tailor 18 Dec '20

18 Dec '20

Hi, Here is setup details Kamailio Proxy a <-----> Kamailio Proxy b <-----> Asterisk Kamailio proxy b auth based on IP Kamailio Proxy a 1. Kamailio proxy sends an Invite request to Kamailio proxy b with CSeq: 102 INVITE 2. Kamailio proxy b check again IP auth and sends invite request to asterisk with CSeq: 102 INVITE 4 asterisk response with 401 Unauthorized with CSeq: 102 5. Kamailio proxy b sends ACK request with CSeq: 102 ACK 6. Kamailio proxy b sends INVITE request with CSeq: 103 INVITE 7. Asterisk sends 100 trying and 180 ringing with CSeq: 103 INVITE 8. Kamailio proxy b relay 180 ringing with CSeq: 102 INVITE 9. Asterisk send 200 Ok with CSeq: 103 INVITE and proxy b relay it to proxy a with CSeq: 102 INVITE 10. Now Proxy a send an ACK with CSeq: 102 ACK and proxy b relay ACK to asterisk with CSeq: 102 ACK Now asterisk is retransmitting 200 ok with CSeq: 103 INVITE and proxy a is sending ACK with Cseq: 102 ACK Same happing in case of 487 Request Terminated Here is my config modparam("dialog", "default_timeout", 10800) # 3h modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_mode", 0) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","auth_realm_avp","$avp(arealm)") modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 3 sec modparam("tm", "fr_timer", 2000) modparam("tm", "retr_timer1", 250) modparam("tm", "retr_timer2", 500) # default invite retransmission timeout after 1xx: 60 sec modparam("tm", "fr_inv_timer", 60000) modparam("tm", "reparse_on_dns_failover", 0) if (is_method("INVITE") && !has_totag()) { dlg_setflag(FLD_RTPPROXY); dlg_manage(); } route(RELAY); route[RELAY] { # At this point all checks are passed t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); t_on_failure("MANAGE_FAILURE"); if(is_method("INVITE") && isflagset(FLT_FROMDID)) { #$fd = "sip.comms.kuflink.com"; t_on_failure("RELAY_AUTH"); } if (!t_relay()) { send_reply("500", "Cannot relay to $du/$ru"); } exit; } failure_route[RELAY_AUTH] { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { #$fd = "sip.comms.kuflink.com"; $avp(auser) = "DID_UA_USERNAME"; $avp(apass) = "DID_UA_PASSWORD"; if (uac_auth()) { t_relay(); } exit; } } route[WITHINDLG] { if (!has_totag()) { return; } if (is_method("BYE")) { rtpengine_delete(); } # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("ACK")) { # ACK is forwarded statelessly route(NATMANAGE); } else if (is_method("NOTIFY")) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if (is_method("ACK") ) { if (t_check_trans()) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } send_reply("404","Not here"); exit; } kamailio -v version: kamailio 5.3.7 (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 6.3.0 uname -a Linux example.com 4.15.0-124-generic #127-Ubuntu SMP Fri Nov 6 10:54:43 UTC 2020 x86_64 GNU/Linux SIP logs 2020/12/17 10:08:18.483907 proxy-a.com:5060 -> 10.0.14.118:5060 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.485063 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1;rport=5060 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 2020/12/17 10:08:18.485576 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 102 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 2020/12/17 10:08:18.622867 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="sip1.comms.kuflink.com", nonce="11139d33" Content-Length: 0 2020/12/17 10:08:18.624034 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@proxy-a.com SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.0 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as4107645c Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK Content-Length: 0 2020/12/17 10:08:18.625557 10.0.14.118:5060 -> 10.0.14.118:6000 INVITE sip:123456@proxy-a.com SIP/2.0 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com User-Agent: ProxyA CSeq: 103 INVITE User-Agent: proxy Date: Thu, 17 Dec 2020 10:08:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "7654321" <sip:7654321@did.com >;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 885 Authorization: Digest username="xxxxx", realm="xxxx", nonce="11139d33", uri="sip:123456@proxy-a.com", response="qwewqeqwe", algorithm=MD5 2020/12/17 10:08:18.779483 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.937756 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Length: 0 2020/12/17 10:08:20.938593 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Length: 0 2020/12/17 10:08:22.075814 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.077049 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.103365 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.103877 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.79f351d031912aae585207d2f26d5650.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.0fc1940efc770391762d7d67094c24f8.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK2a5fae72;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:22.575533 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:22.576777 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316 2020/12/17 10:08:22.581649 proxy-a.com:5060 -> 10.0.14.118:5060 ACK sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Route: <sip:proxy-b.com;lr;did=d18.7a9> Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 2020/12/17 10:08:22.582313 10.0.14.118:5060 -> 10.0.14.118:6000 ACK sip:123456@10.0.14.118:6000;alias=109.72.216.251~5060~1 SIP/2.0 Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.33626b5695ad4648ba393bc62bed6a8c.0 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.b9c7eeccd4cd7b44f05b173308dbaf0a.0 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK219933f6;rport=5060 Max-Forwards: 70 From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Contact: <sip:7654321@did.com:5060> Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 ACK User-Agent: proxy Content-Length: 0 P-K-CSeq-Refresh: 103 2020/12/17 10:08:23.575828 10.0.14.118:6000 -> 10.0.14.118:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-b.com:5060 ;branch=z9hG4bKbe6d.619fd995c3fc8764ce5cd9c7be7efeef.1.cs102;received=10.0.14.118;rport=5060 Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 103 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000> Content-Type: application/sdp Require: timer Content-Length: 296 2020/12/17 10:08:23.577863 10.0.14.118:5060 -> proxy-a.com:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy-a.com:5060 ;rport=5060;branch=z9hG4bKbe6d.10c4c5f001c66b8976214d4aef9f635c.1 Via: SIP/2.0/UDP did.com:5060;received=did.com ;branch=z9hG4bK45f94ed9;rport=5060 Record-Route: <sip:proxy-b.com;lr;did=d18.7a9> Record-Route: <sip:proxy-a.com;lr> From: "7654321" <sip:7654321@did.com>;tag=as55c522e1 To: <sip:123456@proxy-a.com>;tag=as147bfa3e Call-ID: 67675c7e636d6e773860255316e54779(a)did.com CSeq: 102 INVITE Server: proxy Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:123456@10.0.14.118:6000;alias=10.0.14.118~6000~1> Content-Type: application/sdp Require: timer Content-Length: 316

4 7

Kamailio RTP proxy require?
by Mihai Cezar 17 Dec '20

17 Dec '20

Hi, Can't figure it out when to use RTPproxy, i am forwarding everything to asterisk in seem to work without RTPproxy. The scenario is like this: Softphone -> INTERNET -> Kamailio - > Asterisk (private ip) Now it seems to work without any rtpproxy but is it ok?

5 10

DLGLOAD stuck in Dispatcher Module(HA Mode)
by harneet singh 16 Dec '20

16 Dec '20

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

2 1

INFO: RTC Devroom Fosdem 2021 - Online Event
by Daniel-Constantin Mierla 16 Dec '20

16 Dec '20

Hello, (cross-posting to kamailio public groups, a few may get this message duplicated). Fosdem 2021 is going to be online, meaning that it opens the possibility for everyone to do a presentation on the RTC Devroom, no longer being required to travel to Brussels. Therefore I am sending here the link to CFP in case someone would be interested to present about Kamailio or something else RTC-related: * https://github.com/FreeRTC/fosdem/blob/master/2021/cfp.md New speakers are always welcome! Only have in mind that the event is targeting developers and source code development, therefore your proposals have to be made from this perspective. Note also that submission deadline is 20th of December, very short time frame from now. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

ERROR: dmq [notification_peer.c:337]: add_server_and_notify(): error adding notification node
by Syed Shahryar 16 Dec '20

16 Dec '20

Here is my setting: modparam("dmq", "server_address", "sip:0.0.0.0:5090" ) modparam("dmq", "notification_address", "sip:0.0.0.0:5090") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 2) modparam("dmq", "worker_usleep", 0) modparam("dmq", "ping_interval", 90) I have tried localhost and I believe other options but to no avail. I also get the following message: ERROR: dmq [dmq_funcs.c:516]: ping_servers(): cannot retrieve initial nodelist from sip:0.0.0.0:5090 How do I set up DMQ for adding and removing any Kamailio instance to/from the cluster and any Kamailio instance could serve any request coming from any client? ie. all Kamailio instances keep the current state in DMQ. I also have dmq_usrloc enabled. -- -- 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

Kamailio dtmf signaling
by Amit Pal 16 Dec '20

16 Dec '20

Dear Team, I am using kamailio + rtpengine, how can I processed dtmf from cfg scripts. even thought I can able to see dtmf over syslog. but can not able to capture in kamailio.cfg. how can I prepared ivr using kamalio? Please find the attached syslog. Regards Amit Pal

3 2

ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri </sip?jwt=..
by Syed Shahryar 15 Dec '20

15 Dec '20

I am sending JWT tokens via SIP URL through JSSIP for authentication before websocket connection is made. However, I am getting errors like: ERROR: <core> [core/parser/parse_uri.c:1326]: _parse_ruri(): bad uri < */sip?jwt=..* I have the following setting in kamailio config: modparam("xhttp", "url_match", "^/sip") // only match this path Is this a bug or some configuration issue? How do I resolve it? -- -- 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

X-Forwarded-for IP to asterisk
by Mihai Cezar 15 Dec '20

15 Dec '20

Hello, Is it something like the XFF for Kamailio so I can see in asterisk (sip show peers) the ip of the registered softphone and not Kamailio's IP ? Thanks!

4 5

How to use port number using $env in listen
by Syed Shahryar 15 Dec '20

15 Dec '20

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.

2 1

Losing MySQL connections
by Duarte Rocha 15 Dec '20

15 Dec '20

Greetings all, I'm having some dropped MySQL connections in my Kamailio in times of low or no activity in the proxy. These are the errors i'm getting : db_mysql [km_dbase.c:123]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) <core> [db_query.c:348]: db_do_update(): error while submitting query usrloc [ucontact.c:1147]: db_update_ucontact_ruid(): updating database failed usrloc [ucontact.c:1663]: update_contact_db(): failed to update database registrar [save.c:784]: update_contacts(): failed to update contact sl [sl_funcs.c:414]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) In order to try to fix this i've added timeout_interval and ping_interval from the db_mysql module. Is this the best approach? I've restarted my Kamailio but i don't see any ping going to my BD. Is there any other config that I need to add so that the pings are sent? Is it possible to do a ping to verify connection and reconnect mannualy on the config file? Best Regards,

2 1

Problems debugging memory issues
by Sheldon L 15 Dec '20

15 Dec '20

Hi, I'm trying to investigate a possible memory leak issue in Kamailio and I'm following the instructions indicated in https://www.kamailio.org/wiki/tutorials/troubleshooting/memory I'm using Kamailo 4.4 with the following kamailio internals: Print out of kamailio internals Version: kamailio 4.4.2 (x86_64/linux) 892ad6 Default config: /apollo/env/TachyonPlatformKamailio/etc/kamailio/kamailio.cfg Default paths to modules: /apollo/env/TachyonPlatformKamailio/lib64/kamailio/modules Compile flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES MAX_RECV_BUFFER_SIZE=262144 MAX_LISTEN=16 MAX_URI_SIZE=1024 BUF_SIZE=65535 DEFAULT PKG_SIZE=8MB DEFAULT SHM_SIZE=64MB ADAPTIVE_WAIT_LOOPS=1024 TCP poll methods: poll, epoll_lt, epoll_et, sigio_rt, select Source code revision ID: 892ad6 Compiled with: x86_64-unknown-linux-gnu-gcc 4.9.4 Compiled on: 23:36:48 Dec 8 2020 Thank you for flying kamailio! I have attached to one of the kamailio tcp receiver processes via GDB and I am trying to dump memory blocks from the memory manager. However, mem_block is not in the current_context. The following is the output from my gdb session: Copyright (C) 2013 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html > This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-amazon-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /opt/codedeploy-agent/deployment-root/95bb1634-e653-4cec-b327-99b18b9227a8/d-7AJ1XZP58/deployment-archive/sbin/kamailio...done. Attaching to program: /apollo/env/TachyonPlatformKamailio/sbin/kamailio, process 8946 ptrace: Operation not permitted. (gdb) set $i=0 (gdb) set $a = mem_block->first_frag No symbol "mem_block" in current context. (gdb) info list Undefined info command: "list". Try "help info". (gdb) info threads No threads. Is there something I need to do before accessing the mem_block symbol? Also do you have any other suggestions for debugging memory leaks that may not be documented? Note: i can't upgrade to the latest version of Kamailio as we have custom modules and would require considerable effort to upgrade. Thanks, Sheldon.

2 1

sngrep TLS connection
by Agiftel 15 Dec '20

15 Dec '20

Hi all, does anyone was able to use sngrep (with -k option) to decrypt TLS connections? I'am doing what help says; used private key is the correct one; but no traffic is seen on sngrep. If i use not ecrypted call i can see everything, so sngrep is working good. Any hint? regards -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

5 8

Kamilio+asterisk = deregister after 2-3 minute
by Mihai 15 Dec '20

15 Dec '20

Hi, I am forwarding all the traffic to my asterisk server ( including registers) everything seems to work when i first register but if stay idle 2-3 minute it gets unregistered and then i can't register back unless i restart kamailio see attached print screen

6 6

voltdb support
by Ali Taher 14 Dec '20

14 Dec '20

Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher

2 8

Kamailio v5.4.3 Released
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, Kamailio SIP Server v5.4.3 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.3. For more details about version 5.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/12/kamailio-v5-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Wrong To tag in t_reply
by Ivan Ribakov 14 Dec '20

14 Dec '20

Hi all, I’m trying to implement following routing behaviour: - ingress INVITE is sequentially forked to 1 or more endpoints - under certain conditions sequential forking may be stopped and pre-defined error returned to the caller Here’s is part of the config responsible for handling errors: failure_route[ROUTE_ELEMENT_FINAL_FAILURE_HELPER] { route(ROUTE_ELEMENT_FINAL_FAILURE); } route[ROUTE_ELEMENT_FINAL_FAILURE] { # We need to check if the error happens before the first egress, because T-state is not defined yet so we can not use t_reply if ($var(before_first_egress) == 0) { t_reply("$var(sipCode)", "$var(sipMessage)"); } else { sl_send_reply("$var(sipCode)", "$var(sipMessage)"); } } Here is a simple scenario where delivering call to first endpoint times out after receiving 180 Ringing (using custom fr_inv_timer value) where: - .33 is caller - .208 is Kamailio - .138 is callee Notice that 504 is retransmitted by Kamailio. Here is a comparison of 180 and 504 returned by Kamailio: As can be seen, 180 was proxied from the callee and has callee’s to-tag (tag=CALLEE1), however 504 that was generated by Kamailio using t_reply() has a different to-tag (presumably generated by Kamailio). How can I tell Kamailio not to generate new to-tag but to use the last one that proxied back to the caller? Thanks in advance for any help! Regards, Ivan

2 2

Releasing Kamailio v5.4.3
by Daniel-Constantin Mierla 14 Dec '20

14 Dec '20

Hello, I am considering to release Kamailio v5.4.3 sometime next week, likely on Monday, December 14, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

how to use environment variables to set listen address
by Syed Shahryar 13 Dec '20

13 Dec '20

This is a generic question about using environment variables in a config file. If I want to use below where TLS_IP itself is a constant defined somewhere else. TLS_IP might be set via an env variable as well. #!define TLS_TO_LISTEN tls:TLS_IP:$env(TLS_PORT) Will that work? I see syntax errors and other weird issues. What's the right syntax? -- -- 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

DMQ using Consul
by Syed Shahryar 12 Dec '20

12 Dec '20

Can someone share their configuration file explaining how they used consul to achieve high availability? -- -- 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

empty body on first NOTIFY on REGINFO subscription - bug ?
by Adrian Tabacioiu 11 Dec '20

11 Dec '20

Hello, If the user never registered (or few hours passed from its unregister) the subscription to REGINFO will return an empty body NOTIFY (no xml). I don't see any flag in module to force a body for this case, as in the presence module is force_dummy_presence who can force a body on a first NOTIFY. The RFC3680 says : o Notifications triggered from a SUBSCRIBE contain full state (the list of all contacts bound to the address-of-record). It appears that in this case the response is not conform to RFC. Regards, Adrian

1 0

porting keepalive module new functions on 5.4 on 5.3?
by Benoit Chesneau 11 Dec '20

11 Dec '20

2 new functions have been added to the keepalive module: - [ka_add_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) - [ka_del_destination(sip_uri, owner)](https://kamailio.org/docs/modules/5.4.x/modules/keepalive.html#keep…) Is there any possibility to do the same on 5.3? It seems that on 5.3 adding a destination can only be done statically: https://kamailio.org/docs/modules/5.3.x/modules/keepalive.html#idm53 or can it also be done dynamically when receiving an event using the kazoo module for example? Benoît Chesneau, Enki Multimedia — t. +33608655490 Sent with [ProtonMail](https://protonmail.com) Secure Email.

2 1

Dispatcher algorithm 11 possible priority drifting
by Володимир Іванець 10 Dec '20

10 Dec '20

Hello all! We are running Kamailio version 5.3.3 with 16 small Asterisk servers. Kamailio uses a dispatcher module to distribute calls. Algorithm #11 is selected. All destinations are configured with priority set to 50 and attribute to rweight=50. We noticed that one server constantly receives more calls than others. I run a few tests. I was sending 1000 calls to the system. 4 per second. All servers except one were getting around 60 calls while the last one - around 100. I then noticed that the server which receives most calls is always last in the "kamcmd dispatcher.list" command output. I then changed the order in the dispatcher DB table and repeat the test. The other server that now was last was getting the most calls. Does anyone else use algorithm #11 and finds the same thing? Is there something additional that I can provide to help with the investigation? Thanks!

3 3

kamailio.cfg fomatter
by Juha Heinanen 10 Dec '20

10 Dec '20

After noticing that emacs mode is missing, below is my first cut of kamailio-mode.el. -- Juha ----------------------------------------------------------------------- ;; kamailio mode for emacs (function list is not complete) ;; activate by starting your kamailio.cfg with this kind of line: ;; #-*-mode: kamailio; tab-width: 4; c-basic-offset: 4; (setq kamailio-font-lock-keywords (let* ( ;; define several category of keywords (x-keywords '("branch_route" "break" "case" "default" "else" "event_route" "exit" "failure_route" "for" "if" "loadmodule" "onreply_route" "onsend_route" "reply_route" "request_route" "return" "route" "switch" "while")) (x-types '()) (x-constants '()) (x-events '()) (x-functions '("acc_db_request" "add_path_received" "add_rr_param" "allow_routing" "allow_trusted" "append_hf" "append_to_reply" "avp_check" "avp_delete" "change_reply_status" "check_route_param" "defined" "dp_translate" "ERROR" "fix_nated_sdp" "force_send_socket" "forward" "get_redirects" "handle_ruri_alias" "inactivate_gw" "INFO" "in_list" "isflagset" "is_avp_set" "isbflagset" "is_direction" "is_domain_local" "is_e164" "is_gflag" "is_gruu" "is_method" "is_numeric" "is_present_hf" "jsonrpc_dispatch" "load_gws" "lookup_domain" "lookup" "loose_route" "modparam" "mt_match" "NOTICE" "proxy_challenge" "record_route" "redirecting" "remove_hf" "resetflag" "rtpengine_answer" "rtpengine_delete" "rtpengine_offer" "save" "sdp_content" "sdp_transport" "send_reply_mode" "send_reply" "setbflag" "setflag" "setxflag" "set_reply_close" "set_reply_no_connect" "set_rtpengine_set" "set_uri_host" "set_uri_user" "set_forward_no_connect" "sql_query" "sql_result_free" "starts_with" "subst_uri" "tel2sip" "t_check_status" "t_is_canceled" "t_load_contacts" "t_next_contacts" "t_on_branch" "t_on_failure" "t_on_reply" "t_relay_cancel" "t_relay" "t_set_fr" "unregister" "WARN" "ws_handle_handshake" "www_challenge" "xhttp_reply")) (x-keywords-regexp (regexp-opt x-keywords 'words)) (x-types-regexp (regexp-opt x-types 'words)) (x-constants-regexp (regexp-opt x-constants 'words)) (x-events-regexp (regexp-opt x-events 'words)) (x-functions-regexp (regexp-opt x-functions 'words))) `( (,x-types-regexp . font-lock-type-face) (,x-constants-regexp . font-lock-constant-face) (,x-events-regexp . font-lock-builtin-face) (,x-functions-regexp . font-lock-function-name-face) (,x-keywords-regexp . font-lock-keyword-face) ))) (defvar kamailio-mode-syntax-table nil "Syntax table for `kamailio-mode'.") (setq kamailio-mode-syntax-table (let ( (synTable (make-syntax-table))) ;; python style comment: “# …” (modify-syntax-entry ?# "<" synTable) (modify-syntax-entry ?\n ">" synTable) synTable)) (define-derived-mode kamailio-mode c-mode "kamailio mode" "Major mode for editing Kamailio configuration files" (setq font-lock-defaults '((kamailio-font-lock-keywords))) (set-syntax-table kamailio-mode-syntax-table)) (provide 'kamailio-mode)

2 5

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 10 Dec '20

10 Dec '20

Hello Kamailions, Well, I think I found the cause of my issue where the Dialog profile was not defined.. Well it was. I had not enabled antiflood, by not enabling antiflood, the htable module was also not loaded... I guess the us of profiles_with_value parameter needs htable? Is that correct? If so, that means the htable is a dependency for the dialog module...., then it would be sometihng that could be added to the documentation of the dialog module. When this is not the case, then I have to dig deeper, as I do not get an error anymore about the profile not being known... Thank you. Gertjan

2 2

REGINFO events not sending
by Adrian Tabacioiu 10 Dec '20

10 Dec '20

Hello, Does anybody has used with success the reginfo modules ? I added lines bellow in configuration, and tried different variations, Even if my subscribe for "reginfo" events is accepted (with 202 OK), I receive no notification when respective user registers. Nor do I get the xml body in the first notification. Added in my config: loadmodule "presence_reginfo.so" loadmodule "pua_reginfo.so" modparam("pua_reginfo", "default_domain", "192.168.60.65") modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.60.65") modparam("pua_reginfo", "publish_reginfo", 1) Thank you in advance, Adrian

2 3

SLA presence not working proper with (SCA module)
by Aatif Shaikh 10 Dec '20

10 Dec '20

Aatif Shaikh <aatif.shaikh(a)ecosmob.com> Mon, Dec 7, 11:00 AM (3 days ago) to sr-users Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx :5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA?

1 0

NAT problem with recvonly calls
by David Cunningham 10 Dec '20

10 Dec '20

Hello, We have a problem with a SIP doorbell device which sends media one way only, and NAT at the receiving device. When the doorbell button is pressed it makes a call to a configured destination. Since the doorbell only sends and doesn't receive it sends the INVITE with sendonly in the SDP, and the destination then replies with a 200 OK with recvonly in the SDP. The problem is that the destination is behind NAT, and its reply contains a private network IP in the SDP. Normally Asterisk when nat=yes works around that by adjusting the destination for RTP to be the address it actually receives audio from, however because this device is recvonly Asterisk never receives audio from it. This means Asterisk keeps trying to send the doorbell's RTP to the private network IP which of course fails, and the destination never gets the RTP from the doorbell. We haven't found a solution in Asterisk to this, so are now looking to Kamailio which acts as a load-balancing proxy in front of Asterisk for one. For example, maybe we could use fix_nated_sdp, but only on 200 OK's with recvonly. Has anyone else encountered this, and are there any recommended solutions? Thank you in advance! -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

4 8

Re: [SR-Users] NAT problem with recvonly calls
by Ovidiu Sas 09 Dec '20

09 Dec '20

The doorbell is the caller. ICE support would be required on the callee side (between NATed callee and asterisk). There's no need for ICE support on the doorbell/caller side. -ovidiu On Wed, Dec 9, 2020 at 2:42 AM Alex Balashov <abalashov(a)evaristesys.com> wrote: > > ICE support just doesn't seem to me to be something a SIP doorbell would > have. > > On 12/9/20 2:39 AM, Olle E. Johansson wrote: > > > > > >> On 8 Dec 2020, at 18:55, Richard Fuchs <rfuchs(a)sipwise.com > >> <mailto:rfuchs@sipwise.com>> wrote: > >> > >> Use IPv6 > > > > While I like that proposal, assuming that IPv6 will not have a stateful > > firewall is propably not a good assumption. > > So I suspect that an IPv6 firewall would not let the packets in if there > > is not outbound traffic first. I do like Ovidius > > ICE-based solution and would like to add RTCP - if muxed it would open > > the port, but the likelyhood that it’s > > implemented is propably low. > > > > IPv6 greetings! > > /O :-) > >> > >> Cheers > >> > >> On 07/12/2020 23.01, David Cunningham wrote: > >>> Hello, > >>> > >>> We have a problem with a SIP doorbell device which sends media one > >>> way only, and NAT at the receiving device. > >>> > >>> When the doorbell button is pressed it makes a call to a configured > >>> destination. Since the doorbell only sends and doesn't receive it > >>> sends the INVITE with sendonly in the SDP, and the destination then > >>> replies with a 200 OK with recvonly in the SDP. The problem is that > >>> the destination is behind NAT, and its reply contains a private > >>> network IP in the SDP. > >>> > >>> Normally Asterisk when nat=yes works around that by adjusting the > >>> destination for RTP to be the address it actually receives audio > >>> from, however because this device is recvonly Asterisk never receives > >>> audio from it. This means Asterisk keeps trying to send the > >>> doorbell's RTP to the private network IP which of course fails, and > >>> the destination never gets the RTP from the doorbell. > >>> > >>> We haven't found a solution in Asterisk to this, so are now looking > >>> to Kamailio which acts as a load-balancing proxy in front of Asterisk > >>> for one. For example, maybe we could use fix_nated_sdp, but only on > >>> 200 OK's with recvonly. > >>> > >>> Has anyone else encountered this, and are there any recommended > >>> solutions? > >>> > >>> Thank you in advance! > >>> > >>> -- > >>> David Cunningham, Voisonics Limited > >>> http://voisonics.com/ <http://voisonics.com/> > >>> USA: +1 213 221 1092 > >>> New Zealand: +64 (0)28 2558 3782 > >>> > >>> _______________________________________________ > >>> Kamailio (SER) - Users Mailing List > >>> sr-users(a)lists.kamailio.org > >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > sr-users(a)lists.kamailio.org > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Alex Balashov | Principal | Evariste Systems LLC > > Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) > Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- VoIP Embedded, Inc. http://www.voipembedded.com

1 0

GitHub discussion
by Sergey Safarov 09 Dec '20

09 Dec '20

GitHub release new feature - discussion. Example https://github.com/vercel/vercel/discussions

2 1

Kamailio as an LCR based on prices but also signaling quality indicators
by Abdoul Osséni 08 Dec '20

08 Dec '20

Hello, I am trying to configure kamailio as an LCR based on prices but also signaling quality indicators like ASR and ACDR. Can you please help me and tell me where to start? Regards Abdoul

4 3

kamailio-5.2.0 Crash - udp_send failed
by Hamid Hashmi 07 Dec '20

07 Dec '20

I am using "kamailio 5.2.0-dev6 (x86_64/linux) 48ed62-dirty" what could be the reason of the crash its usually crash with the following logs Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36eee8c560,418,0,10.100.227.202:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20771]: WARNING: tm [t_reply.c:1581]: t_retransmit_reply(): send pr buffer failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb948,358,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20756]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b4a80,358,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20744]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ac740,416,0,10.100.72.230:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20783]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8b5098,356,0,10.100.56.38:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8f18,416,0,10.100.177.91:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20805]: ERROR: tm [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20752]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <script>: Re-Transmit: REGISTER (sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org (10.100.236.57:5060) to sip:410060269353579@ims.mnc006.mcc410.3gppnetwork.org, 1SDq5cuY3iDYFKLATTp) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8e8a08,354,0,10.100.2.76:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20761]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8ad100,360,0,10.100.144.137:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20767]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: <core> [core/udp_server.c:596]: udp_send(): sendto(sock,0x7f36ef8eb860,360,0,10.100.140.226:5060,16): Invalid argument(22) Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: CRITICAL: <core> [core/udp_server.c:601]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 25 16:58:41 pcscf [Proxy-CSCF][20777]: ERROR: sl [../../core/forward.h:219]: msg_send_buffer(): udp_send failed Sep 25 16:58:43 pcscf [Proxy-CSCF][20861]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.64.49]:32888 -> Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 16:58:44 pcscf [Proxy-CSCF][20853]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f36a739fe78 r: 0x7f36a739fef8 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:748]: handle_sigs(): child process 20815 exited by a signal 11 Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 16:58:44 pcscf kernel: [52607827.581845] net_ratelimit: 16 callbacks suppressed Sep 25 16:58:44 pcscf [Proxy-CSCF][20742]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 16:59:01 pcscf CRON[24805]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 16:59:01 pcscf CRON[24804]: (CRON) info (No MTA installed, discarding output) Sep 25 16:59:44 pcscf [Proxy-CSCF][20742]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... but last time its also crashed with the following logs as well Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.183.34]:39022 -> Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:39:59 pcscf [Proxy-CSCF][28015]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca8d0150 r: 0x7f92ca8d01d0 Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:39:59 pcscf [Proxy-CSCF][28010]: ERROR: cdp [authstatemachine.c:404]: auth_client_statefull_sm_process(): auth_client_statefull_sm_process(): Received invalid event 6 while in state Open! Sep 25 18:40:00 pcscf [Proxy-CSCF][28034]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 77 Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:301]: tcp_read_data(): error reading: Connection timed out (110) ([10.100.100.218]:1337 -> Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:302]: tcp_read_data(): -> [10.161.224.228]:5060) Sep 25 18:40:01 pcscf [Proxy-CSCF][28026]: ERROR: <core> [core/tcp_read.c:1485]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f92ca87f070 r: 0x7f92ca87f0f0 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:748]: handle_sigs(): child process 27989 exited by a signal 11 Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: ALERT: <core> [main.c:751]: handle_sigs(): core was generated Sep 25 18:40:01 pcscf kernel: [52613904.372973] net_ratelimit: 29 callbacks suppressed Sep 25 18:40:01 pcscf [Proxy-CSCF][27919]: CRITICAL: cdp [diameter_peer.c:423]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Sep 25 18:40:01 pcscf CRON[30581]: (root) CMD (/bin/bash /root/volumes/monitoring/src/services_status.sh -p http://10.161.224.25:9091 -s kamailio_pcscf,sems -n pcscf) Sep 25 18:40:01 pcscf CRON[30580]: (CRON) info (No MTA installed, discarding output) Sep 25 18:40:02 pcscf sems[2858]: [#7f2080044700] [update_uac_request, trans_layer.cpp:2080] WARNING: While sending 200 ACK: no matching transaction Sep 25 18:41:01 pcscf [Proxy-CSCF][27919]: CRITICAL: <core> [main.c:654]: sig_alarm_abort(): shutdown timeout triggered, dying... PS: Kamailio is not listening on localhost. Regards Hamid R. Hashmi

3 4

Kamilio dispatcher not registering with asterisk
by Mihai 07 Dec '20

07 Dec '20

Hi all, i have a problem with kamailio 5.3.x and asterisk 13, i basically forward everything, but i cant manage to make it register. The register arrives on asterisk see the capture bellow: Any help is much appreciated! ---------------- REGISTER sip:oro.sips.nl;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 104.201.100.15;branch=z9hG4bK4d5e.997621d931a2ad9f6423e2ac7782a319.1 Via: SIP/2.0/UDP 34.12.55.102:34626 ;received=34.12.55.102;branch=z9hG4bK-524287-1---0a0bffd64bbd9941;rport=34626 Max-Forwards: 69 Contact: <sip:test-user1@34.12.55.102:34626 ;transport=UDP;rinstance=85c33e976be3dad1> To: <sip:test-user1@oro.sips.nl;transport=UDP> From: <sip:test-user1@oro.sips.nl;transport=UDP>;tag=6d8fd914 Call-ID: h6lkViFgxurpVRkw4Ly-GQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 5.4.9 rv2.10.11.7 Allow-Events: presence, kpml, talk Content-Length: 0 Path: <sip:104.201.100.15;lr;received=sip:34.12.55.102:34626> -------------------the config ------- # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_FS 10 ####### Global Parameters ######### debug=4 log_stderror=yes memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:127.0.0.1:5060 sip_warning=no ####### Modules Section ######## # set module path #mpath="/usr/local/lib/kamailio/modules/" #loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "dispatcher.so" loadmodule "path.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", FLT_ACC) modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", DBURL) #modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") modparam("dispatcher", "ds_ping_from", "sip:proxy@mykamailio") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_threshold", 10) modparam("dispatcher", "ds_inactive_threshold", 5) modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=488;class=3") modparam("dispatcher", "ds_probing_mode", 1) modparam("path", "use_received", 1) ####### Routing Logic ######## # main request routing logic request_route { # per request initial checks route(REQINIT); if(ds_is_from_list()) { setflag(FLT_FS); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); if (isflagset(FLT_FS)) { route(FROM_FS); exit; } ### only initial requests (no To tag) # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); } route[FROM_FS] { record_route(); route(RELAY); exit; } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' switch ($fd){ case "oro.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("1", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("1", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; case "oro2.sips.nl": send_reply(483, "Ok, Skipper!"); if (!ds_select_dst("2", "4")) { log("ERROR: no active destinations found!\n"); send_reply(503, "Service Unavailable"); exit; } ds_select_dst("2", "4"); log("---- flag value is $dd -- branch $fd \n"); xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); break; default: log("unknow destination?"); send_reply(503, "No service defined"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } }

3 4

SLA presence not working proper with (SCA module)
by Aatif Shaikh 07 Dec '20

07 Dec '20

Hello All I configure my kamailio.cfg to handle SLA using (SCA module) and shared presence is working but still facing some issue in case of hold invite and BYE and the reason behind of that issue is following error Nov 26 07:38:40 SBC-4-1 /usr/local/sbin/kamailio[104107]: ERROR: sca [sca_call_info.c:1008]: sca_call_info_invite_request_handler(): Failed to update sip:4569@test.sip.abcd.com appearance-index 0 to active actually, in Kamailio appearances, I can see the entry for this call but the index is 1 instead of 0, and Kamailio going to update with a 0 index [root@SBC-4-1 ntcarfte-kamailio]# kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 active 1606376289 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120MTQ5OTcyZjFhMjMyNmI1ZGE1MWY4ODc2M2RkN2VmZmQ 45dc18d8 H6cS9vecZ248B if I manually update appearances using RPC command with index 1 then the phone started blinking as I change state to held # kamcmd sca.update_appearance sip:4569@sc4test.sip.teledge.com 1 held # kamcmd sca.all_appearances sip:4569@test.sip.abcd.com 1 held 1606376516 sip:4569@10.xx.xx.xx:5070;transport=udp sip:2003@test.sip.abcd.com 186161_mobile-rel120NzE5MDgyZDc1NjJiYjcwMWFlYmI3NzM3NjE2OTRhZjU 08f005cb eK9ratm83g92e kamailio.cfg route[SCA] { if(is_method("SUBSCRIBE")) { if ($hdr(Event) == "call-info" || $hdr(Event) == "line-seize") { xlog("L_INFO","(MAIN) :HELLO ($avp(uuid)) : $avp(rsi) $sp $hdr(Event)"); xdbg("SCA: $hdr(Event) SUBSCRIBE $ru from $si:$sp"); sca_handle_subscribe(); exit; } return; } if (!is_method("BYE|CANCEL|INVITE|PRACK|REFER")) { return; } sca_call_info_update(); } route[RELAY_OUTBOUND_FS] { xlog("L_INFO","(RELAY_OUTBOUND_FS) : ($avp(uuid)) : INSIDE ROUTE "); if (!has_totag()){ $avp(set_h) = 1; } t_on_reply("REPLY_OUTBOUND_FS"); route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } route[RELAY_INBOUND_FS] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(s:puburis_caller) = $fu; setflag(8); dlg_manage(); } if(is_method("REGISTER")) { t_on_reply("REPLY_REGISTER"); } else { t_on_reply("REPLY_INBOUND_FS"); } route(SCA); if (!t_relay()) { sl_reply_error(); } exit; } onreply_route[REPLY_INBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } onreply_route[REPLY_OUTBOUND_FS] { xdbg("incoming reply\n"); if (status =~ "[456][0-9][0-9]") { # don't update SCA state here, since there may be # failure route processing (e.g., call forwarding). # update state in failure route instead. return; } route(SCA); } can someone help me with this Did I missed something to handle SLA? -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. 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1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 06 Dec '20

06 Dec '20

Thank you Joel for the tip. But that is also not it..... I tried of course, but when you look at the Dialog module documentation is says: modparam("dialog", "profiles_with_value", "caller ; my_profile") So there are spaces in the modparam config for the dialog profiles. Its just so strange that when starting Kamailio I get the error that the Profile outbound is not configured, while it is... Log line: Dec 6 15:04:04 proxy01 /usr/local/sbin/kamailio[76664]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined kamailo.cfg # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound") Im starting to think that I am missing some configuration in the kamailio.cfg file... Rgds, Gertjan

1 0

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 05 Dec '20

05 Dec '20

Thanks again Sergiu, /Hmmm... if dialog params are wrapped in a ifdef condition, then do you have something like #!define WITH_DIALOG at the top?/ But the #!define WITH_DIALOG is to choose between loading module parameters or not loading module parameters... When you do not add #!ifdef WITH_XXX before the parameters and #!endif after, then those parameters will be loaded when starting Kamailio. So, I still have the issue that the profile is not being loaded. I have tried renaming the profiles, in the route and the parameters, but that does not make a difference.. # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "inbound ; outbound") route[CONCURRENT_IN] { #Add call to customer profile #Check if customer has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("inbound", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) >= $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("inbound", "$avp(custid)"); return; } But this still results in: Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <inbound> not defined Dec 5 09:34:33 proxy01 /usr/local/sbin/kamailio[75755]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1578

2 1

question regarding rtpengine and ICE candidate selection
by Andrew Chen 04 Dec '20

04 Dec '20

Hi all, I was wondering if someone can help me understand how the ICE parameter works in the rtpengine module works. So basically our client does an ICE candidate lookup and grabs a list of them and applies it to the INVITE that gets sent to the Kamailio. The list looks like this: 2020-12-03T18:05:46.456030+00:00 ashmainkama51 kamailio[22147]: a=candidate:3373280875 1 udp 2122262783 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 59827 typ host generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456057+00:00 ashmainkama51 kamailio[22147]: a=candidate:3040609428 1 udp 2122197247 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 59828 typ host generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456081+00:00 ashmainkama51 kamailio[22147]: a=candidate:1681997092 1 udp 2122129151 192.168.1.66 59829 typ host generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456106+00:00 ashmainkama51 kamailio[22147]: a=candidate:2274611867 1 tcp 1518283007 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 9 typ host tcptype active generation 0 network-id 2 network-cost 10 2020-12-03T18:05:46.456131+00:00 ashmainkama51 kamailio[22147]: a=candidate:4223662180 1 tcp 1518217471 2001:8a0:78fc:7000:d979:bf75:dbc0:69f 9 typ host tcptype active generation 0 network-id 3 network-cost 10 2020-12-03T18:05:46.456155+00:00 ashmainkama51 kamailio[22147]: a=candidate:717406676 1 tcp 1518149375 192.168.1.66 9 typ host tcptype active generation 0 network-id 1 network-cost 10 2020-12-03T18:05:46.456180+00:00 ashmainkama51 kamailio[22147]: a=candidate:2130547417 1 udp 8199935 206.81.191.27 61165 typ relay raddr 85.247.0.121 rport 53091 generation 0 network-id 1 network-cost 10 In the rtpengine_offer, I have this: rtpengine_offer("ICE=force DTLS=passive replace-session-connection replace-origin external internal"); What i am trying to do is to tell the freeswitch endpoint to use rtpengine as ICE candidate and I see in the SDP this is happening: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Btw..206.81.191.52 is a separate AWS instance running rtpengine binary. On the rtpengine_answer, I tell the remote client side the same thing..use rtpengine as your ICE candidate: rtpengine_answer("ICE=force DTLS=passive replace-session-connection replace-origin internal external"); and this is snippet from 200 OK: a=candidate:6296910676 1 udp 659136 206.81.191.52 52766 typ host generation 0 Question: Why does the rtpengine logs still show that it's trying to use 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71 as RTP candidate in this scenario? Dec 3 18:05:47 ashmainrtpe42 rtpengine[8505]: DEBUG: [ep1sbnkk9tikhg4kpmot]: Forward to sink endpoint: 2001:8a0:78fc:7000:e1d7:e93:3c50:ee71:59827 (RTP seq 25423 TS 0) I thought ICE=force will handle such adjustments so that all rtp is handled by the rtpengine to our client and bypass the STUN server? Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 22

REFER from MS teams to Kamailio.
by Yuriy Nasida 04 Dec '20

04 Dec '20

Well, I saw similar questions in the list already but looks like nobody has answer. Please look at REFER below. Kamilio get REFER from MS and sends it to FS node. Next, FS node try to make 3th call for some reason.I expect that FS will not do 3th call and just will connect Alice and Bob itself. 2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060 REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0 FROM: Customer1<sip:MS_TRUNK_NUMBER@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=a860c50a3fb54d08b4e5740fa2dfb3d6 TO: <sip:Alice_number@FQDN_OF_TRUNK:5061>;user=phone;tag=e8ct9S6ty13va CSEQ: 4 REFER CALL-ID: 2c71b2a6669b5343a231e1244b19c945 MAX-FORWARDS: 50 Via: SIP/2.0/UDP FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998 CONTACT: <sip:api-du-a-euwe.pstnhub.microsoft.com:443 ;x-i=6b68e7aa-f5e2-44ec-9edf-0bacbabfce07;x-c=2c71b2a6669b5343a231e1244b19c945/d/8/b68f86794a8e44d19543f8edbee6b2fc CONTENT-LENGTH: 0 REFER-TO: <sip:Bob_number@sip.pstnhub.microsoft.com:5061 ;user=phone;transport=tls> REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061 ;x-m=8:orgid:21bc47d3-c050-4292-8234-46f7005b97aa;x-t=fb788ef8-3c4c-455a-8d62-f3c20832c0d3;x-ti=6b68e7aa-f5e2-44ec-9edf- acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY P-ASSERTED-IDENTITY: <tel:MS_TRUNK_NUMBER>,< sip:customer1@m365x587912.onmicrosoft.com> PRIVACY: id X-AUTH-IP: 52.114.75.24 X-AUTH-PORT: 3136 Any advice?

4 11

Re: [SR-Users] Dialog profile not defined
by Gertjan Wolzak 04 Dec '20

04 Dec '20

Hello Sergiu, Thank you for the pointer, the line 1576 is: get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); That is where Kamailio refuses to start.... And as you mentioned my error in the IF statement after correction you pointed out, Kamailio still refuses on the fact that the profile <callee> was not defined... Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 4 13:42:50 proxy01 /usr/local/sbin/kamailio[75008]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Main process exited, code=exited, status=255/n/a Dec 4 13:42:50 proxy01 systemd[1]: kamailio.service: Failed with result 'exit-code'. Does anyone else maybe have an idea on where I go wrong? Rgds, Gertjan

2 1

Fix Sdp Error
by Duarte Rocha 04 Dec '20

04 Dec '20

Hello, I noticed that when i do calls behind NAT i'm getting this error when i call fix_nated_sdp() : "nathelper [nathelper.c:1659]: replace_sdp_ip(): can't extract 'a=rtcp' IP from the SDP" As far as i can see, all the ips on SDP are correctly replaced to a public IP, so i don't understand the origin of this error. This is the SDP received by Kamailio : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 192.168.24.89 Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 192.168.24.89 Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv And this is the SDP sent by Kamailio after NAT being treated : Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3421914885 1 IN IP4 1.1.1.1 (Public IP) Session Name (s): SIPPER for PhonerLite Connection Information (c): IN IP4 1.1.1.1 (Public IP) Time Description, active time (t): 0 0 Media Description, name and address (m): audio 5062 RTP/AVP 107 8 0 2 3 97 110 111 9 18 11 118 101 Media Attribute (a): rtpmap:107 opus/48000/2 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:2 G726-32/8000 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:97 iLBC/8000 Media Attribute (a): rtpmap:110 speex/8000 Media Attribute (a): rtpmap:111 speex/16000 Media Attribute (a): rtpmap:9 G722/8000 Media Attribute (a): rtpmap:18 G729/8000 Media Attribute (a): fmtp:18 annexb=yes Media Attribute (a): rtpmap:11 L16/44100 Media Attribute (a): rtpmap:118 L16/16000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): ssrc:2716178988 Media Attribute (a): sendrecv Media Attribute (a): oldmediaip:192.168.24.89 Media Attribute (a): oldmediaip:192.168.24.89 Is there something failing in the SDP fix? Best Regards,

1 0

TOPOS replication with KDMQ
by James Browne 04 Dec '20

04 Dec '20

Has anyone ever implemented DMQ replication of TOPOS data between kamailio servers? I'm trying to find the easiest (to maintain) way to implement this replication, possibly without using an external database. James

3 3

presence not work proper with subs_db_mode = 2
by Aatif Shaikh 04 Dec '20

04 Dec '20

Hello, all have a problem with Kamailio presence, I have configured my kamailio.cfg to handle dialog presence and facing issue in case of subs_db_mode = 2 in case of mode 2 facing the following presence issue. 1. after hangup presence is still up 2. sometimes stuck in a ringing state 3. sometimes presence not came..etc but in the case of subs_db_mode = 3 (DB ONLY) mode presence working fine not facing any issue with this mode. I noticed that sometimes Kamailio will throw the following message during PUBLISH processing and it will not generate appropriate NOTIFY I also verify inactive watcher subscription is there for that user and also a proper entry in the dialog table. MSG during PUBLISH processesing and it will not generate appropriate NOTIFY Oct 30 07:42:10 NC-HOSTED-1 /usr/local/sbin/kamailio[2409]: DEBUG: presence [notify.c:1444]: publ_notify(): Could not find subs_dialog kamailio.cfg param modparam("pua", "db_url", DBURL_KAMAILIO) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 100) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "default_expires", 3600) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", SERVER_IP) modparam("dialog", "db_url", DBURL_KAMAILIO ) modparam("dialog", "db_mode", 1) modparam("dialog", "db_update_period", 260) modparam("dialog", "db_fetch_rows", 500) modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", 8) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "use_pubruri_avps", 1) modparam("pua_dialoginfo", "include_localremote", 1) modparam("pua_dialoginfo", "override_lifetime", 300) modparam("pua_dialoginfo", "pubruri_caller_avp", "$avp(s:puburis_caller)") modparam("pua_dialoginfo", "pubruri_caller_dlg_var", "pubruri_caller") modparam("pua_dialoginfo", "pubruri_callee_dlg_var", "pubruri_callee") modparam("pua_dialoginfo", "pubruri_callee_avp", "$avp(s:puburis_callee)") modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence", "db_table_lock_type", 0) modparam("presence", "db_url", DBURL_KAMAILIO) modparam("presence", "db_update_period", 5) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 100) modparam("presence", "subs_db_mode", 2) //modparam("presence", "subs_db_mode", 3) modparam("presence", "timeout_rm_subs", 1) //new/ //modparam("presence", "notifier_processes", 2) //new// modparam("presence", "notifier_poll_rate", 20) //new// modparam("presence", "waitn_time",1) modparam("presence", "fetch_rows", 1000) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "max_expires", 3600) modparam("presence_xml", "db_url", DBURL_KAMAILIO) modparam("presence_xml", "force_active", 1) logic for presence request_route { if(is_method("PUBLISH") || is_method("SUBSCRIBE")) { route(HANDLE_PRESENCE); } } route[HANDLE_PRESENCE] { if (! t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; } route[RELAY] { if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { $avp(t_ext) = $tU ; $avp(domain) = $fd ; $avp(et) = $_s(sip:$avp(t_ext)@$avp(domain)) ; $avp(s:puburis_callee) = $avp(et); setflag(8); dlg_manage(); } } I already report this issue in Kamailio git https://github.com/kamailio/kamailio/issues/2540 -- -- Regards, *Aatif Shaikh *| Software Developer +91 9033976268Hangout: aatif.shaikh(a)ecosmob.comSkype : aatif.shaikh(a)ecosmob.com *Ecosmob Technologies Pvt. Ltd. * https://www.ecosmob.com VoIP | Web | Mobile | IoT | Big Data <https://twitter.com/ecosmob> <https://www.facebook.com/Ecosmob> <https://plus.google.com/+Ecosmob> <https://www.linkedin.com/company/ecosmob> <https://in.pinterest.com/ecosmob/> This e-mail message may contain confidential or legally privileged information and is intended only for the use of the intended recipient(s). Any unauthorized disclosure, dissemination, distribution, copying or the taking of any action in reliance on the information herein is prohibited. Ecosmob Technologies is not responsible for errors or omissions in this message and denies any responsibility for any damage arising from the use of e-mail. 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1 0

Dialog profile not defined
by Gertjan Wolzak 03 Dec '20

03 Dec '20

Hello Kamailions, I am running into the following issue. The same setup I had working on 5.0, now I am using a 5.4 version of Kamailio, but I do not believe it has to do with the version, more with my ability to make errors... I am trying to limit incoming calls by using the dialog module. The max concurrent calls value I retrieve from a database. But when I try to start Kamailio it will not because the dialog profile is not defined... These are my configured Dialog parameters: # ---- Dialog params ------------- modparam("dialog", "default_timeout", 7200) modparam("dialog", "db_mode", 0) modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "hash_size", 4096) #modparam("dialog", "detect_spirals", 1) modparam("dialog", "profiles_with_value", "callee") My check on the concurrent inbound calls: route[CONCURRENT_IN] { #Add call to callee profile #Check if callee has not reached inbound call limit #Get max concurrent calls sql_query("cc", "select max_calls_in from calllimit where cust_id='$avp(custid)'", "ra"); $avp(maxcalls)=$dbr(ra=>[0,0]); sql_result_free("ra"); $avp(concurrent) = 0; get_profile_size("callee", "$avp(custid)", "$avp(concurrent)"); if( $avp(concurrent) = $var(max) ) { xlog("L_INFO"," Call limit reached for customer $avp(custid)\r\n "); sl_send_reply("503", "No Lines Available"); exit; } set_dlg_profile("callee", "$avp(custid)"); return; } And the kamailio log error: Dec 3 09:12:08 proxy01 kamailio[74102]: INFO: <core> [main.c:2833]: main(): processes (at least): 21 - shm size: 67108864 - pkg size: 8388608 Dec 3 09:12:08 proxy01 kamailio[74102]: CRITICAL: dialog [dialog.c:391]: fixup_profile(): profile <callee> not defined Dec 3 09:12:08 proxy01 kamailio[74102]: ERROR: <core> [core/route.c:1166]: fix_actions(): fixing failed (code=-6) at cfg:/usr/local/etc/kamailio/kamailio.cfg:1576 Is someone able to see where I go wrong and point me in the right direction? Thank you. Rgds, Gertjan Wolzak

2 2

$TF pseudo variable can not be used
by Володимир Іванець 03 Dec '20

03 Dec '20

Hello! I'm using a $TF variable in my config for some logging purposes. It was providing the correct string with Kamailio version 5.3.8 on CentOS 6 and 8. After updating to Kamailio version 5.4.1, the variable started to show something else. I also tried versions 5.4.2 and development 5.5.0 on CentOS 6 and the problem still persists. With version 5.5.0_dev3 this variable shows "pG▒" for me. I was wondering if someone else is using it too with Kamailio version 5.4.1 or newer on different platforms and has this issue. Thanks!

2 2

kamailio51 installs 5.2.1?
by David Villasmil 03 Dec '20

03 Dec '20

Hello everyone, On a debian buster i added the repo deb http://deb.kamailio.org/kamailio51 buster main deb-src http://deb.kamailio.org/kamailio51 buster main But when i install i get 5.2, is this right? additional info: # apt-cache showpkg kamailio Package: kamailio Versions: 5.2.1-1 (/var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages) (/var/lib/dpkg/status) Description Language: File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_binary-i386_Packages MD5: df3e15f422439e08c305782f5650a98c Description Language: en File: /var/lib/apt/lists/deb.debian.org_debian_dists_buster_main_i18n_Translation-en MD5: df3e15f422439e08c305782f5650a98c 5.1.10.1+bpo10 (/var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages) Description Language: File: /var/lib/apt/lists/deb.kamailio.org_kamailio51_dists_buster_main_binary-i386_Packages MD5: ebddf40d0dfbfde1c479419970f978c7 Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

kamailio not absorbing ACK to CANCEL when behind a Path proxy
by Anthony Alba 02 Dec '20

02 Dec '20

Folks, I have a weird situation, when I enable Path header, kamailio behind a Path proxy, that kamailio doesn't recognise the ACK to CANCEL and tries to forward it. Test scenario: - voice.example.com is a stateless proxy with TLS/UDP bridging to kamailio. - on the UDP leg to kamailio it will add a Path: header to REGISTER UA1 = david is calling UA2 = charles and UA2 is sending 603 1 . UA1: preloaded route set (outbound proxy) # this is the Path proxy Route: sip:voice.example.com;transport=tls;lr 2. UA2: sends CANCEL 603 - kamailio immediately sends its (per-hop?) ACK - kamailio forwards CANCEL to UA1 kamailio sent this (per-hop?) ACK - the Route header is the internal UDP interface of the Path proxy ACK sip:charles@192.168.1.7:37309;transport=TCP;ob SIP/2.0 Via: SIP/2.0/UDP 192.168.122.99:5064;branch=z9hG4bK735d.6b27c8d09e40462ef47eccf90eb10823.0 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: <sip:charles@voice.example.com>;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:192.168.122.100;lr> Content-Length: 0 3. UA1 sends ACK kamailio sees this - top Via is path proxy - 2nd Via is UA (caller) - CSeq correctly matches the INVITE - kamailio does not recognise this ACK from UA1(caller), tries to forward it, and also resends 603 to UA1(caller) 3 times ACK sip:charles@voice.example.com SIP/2.0 Via: SIP/2.0/UDP 192.168.122.100:5060;branch=z9hG4bKb3c9a7a3b6ec9e78d0144cda709f7047 Via: SIP/2.0/TLS 192.168.1.17:44925;rport=44925;branch=z9hG4bKPjFM-9gpaJEghJvN19TjnvScCwCoAGzwEc;alias;received=192.168.1.17 Max-Forwards: 70 From: sip:david@voice.example.com;tag=i1acAHsitf2gcN9uHH-ZSyJ5OdT8O5M0 To: sip:charles@voice.example.com;tag=LmUY79rIdBQiiSakML4F0lHl271VdUn8 Call-ID: qM3W-fMrzPsvcjRSD1okEp5nbgHCF5Hu CSeq: 16667 ACK Route: <sip:voice.example.com;transport=tls;lr> Content-Length: 0 Now kamailio should absorb this ACK, as it has already sent its per-hop ACK. Instead I see ERROR: <core> [core/forward.c:541]: forward_request(): cannot forward to af 2, proto 3 no corresponding listening socket I don't expect kamailio to forward anything at this point. The route block is route[WITHINDLG] and it calls t_relay() normally. route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); # we should absorb this ACK, no? exit; } 4. If I remove the Path proxy and kamailio is the TLS proxy at voice.example.com the call flow works perfectly. The ACK from caller (david) is absorbed. 5. Regular calls, when UA2 accepts the call, work fine. In-dialog requests for BYE, from both UAs, work correctly with this Path proxy. Each UA has the complete 5 element route set( 2 x Path proxy + kamailio + 2 x Path proxy). The working route set looks like this: BYE sip:david@192.168.1.17:44925;transport=TCP;ob SIP/2.0 Via: SIP/2.0/TLS 192.168.1.7:37309;rport;branch=z9hG4bKPjHdUtaUL5AfRcV-SX1Puya4niJKOOQ4VQ;alias Max-Forwards: 70 From: <sip:charles@voice.example.com>;tag=6zOt.H7w8NL5njV7klwGLlSJy.U5yBC9 To: <sip:david@voice.example.com>;tag=Pa0ORUxA-7lK0BnR4pLJ5n8rn2OvQrJy Call-ID: 3gWy4Q7O.-zQ-I07lRKHyFyX6azyCSwG CSeq: 12855 BYE Route: <sip:voice.example.com;transport=tls;lr;r2=on> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:192.168.122.99:5064;lr> Route: <sip:192.168.122.100;lr;r2=on> Route: <sip:voice.example.com;transport=tls;lr;r2=on> User-Agent: basesip 1.0.0 Content-Length: 0 Any ideas? Anthony Alba

1 0

android client SIP TLS client autentication + kamailio
by beer Ll 02 Dec '20

02 Dec '20

Hello Has anybody use a android SIP client with TLS client authentication enabled with kamailio? Do you have suggestion ? Many thanks Luca

3 2

Kamailio 5.4 ndb_redis, selecting a database
by Daniel Donoghue 02 Dec '20

02 Dec '20

Hello, I note that I can use the ndb_redis server config parameter to specify connection details for my redis servers; I also note that the connection string can specify a “db” parameter which can be set from 0 to 9, with the default being 0. I am looking for a way to dynamically set the db parameter. I do not which to add the same server with different names N times just to select a different db and having N times more connections to it than necessary. It looks as though I can call redis_cmd(“svrN”, “SELECT dbN”, “init”); but will that propagate properly and set the correct db for all connections to my redis server for the lifetime of the request? Ideally, I’d like to be able to specify an avp for the db, or have a function to set the db per request. With every blessing, — Daniel Donoghue

1 0

Kamailio Remove Remove "Allow" features from header
by me.projects＠yahoo.com 02 Dec '20

02 Dec '20

Hi, I want to remove some "Allow" features from my Kamailio SBC like I want to keep following only Allow: OPTIONS, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, How can I achieve that? Thanks,

2 1

How to DIY/Setup An Open Source IP PBX Appliance/Server?
by Turritopsis Dohrnii Teo En Ming 01 Dec '20

01 Dec '20

Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server? Good day from Singapore, After reading recent reviews, I gather that Asterisk is the gold standard when it comes to open source VoIP systems and it is the most famous open source PBX out there. Article: Compare the Top 10 Best Open Source PBX Software of 2020 Link: https://www.voipreview.org/business-voip/best-open-source-pbx-software Article: Top 10 Free Open Source PBX Software Solutions Link: https://getvoip.com/blog/2016/09/23/best-open-source-pbx-software/ The following is an excerpt from Wikipedia: "Asterisk is a core component in many commercial products and open-source projects. Some of the commercial products are hardware and software bundles, for which the manufacturer supports and releases the software with an open-source distribution model. AskoziaPBX, a fork of the m0n0wall project, uses Asterisk PBX software to realize all telephony functions. AstLinux is a "Network Appliance for Communications" open-source software distribution.[15] FreePBX, an open-source graphical user interface, bundles Asterisk as the core of its FreePBX Distro[16] LinuxMCE bundles Asterisk to provide telephony; there is also an embedded version of Asterisk for OpenWrt routers. PBX in a Flash/Incredible PBX and trixbox are software PBXes based on Asterisk. Elastix previously used Asterisk, HylaFAX, Openfire and Postfix to offer PBX, fax, instant messaging and email functions, respectively, before switching to 3CX. Issabel is an open-source Unified Communications software which uses Asterisk for telephony functions. It was forked from the open-source versions of Elastix when 3CX acquired it. *astTECS uses Asterisk in its VoIP and mobile gateways." Link: https://en.wikipedia.org/wiki/Asterisk_(PBX) I would like to DIY/setup an IP PBX appliance/server using free open source projects. Which free open source project, mentioned in the list and links above, would you recommend to DIY my IP PBX appliance/server? Should I buy a desktop computer or get one of those appliances listed in the link below to serve as my IP PBX appliance/server? Link: https://www.lazada.sg/products/pfsense-iron-metal-case-fanless-intel-celero… Please also refer me to very good, detailed and well explained guides/tutorials/manuals on setting up open source IP PBX appliances/servers. Lastly, please recommend a cheap and affordable IP phone (suggest brand and model) to go along with my DIY open source IP PBX appliance/server. Mr. Turritopsis Dohrnii Teo En Ming, 42 years as of 1st December 2020 Tuesday, is a TARGETED INDIVIDUAL (TI) living in Singapore. Thank you very much. -----BEGIN EMAIL SIGNATURE----- The Gospel for all Targeted Individuals (TIs): [The New York Times] Microwave Weapons Are Prime Suspect in Ills of U.S. Embassy Workers Link: https://www.nytimes.com/2018/09/01/science/sonic-attack-cuba-microwave.html ******************************************************************************************** Singaporean Targeted Individual Mr. Turritopsis Dohrnii Teo En Ming's Academic Qualifications as at 14 Feb 2019 and refugee seeking attempts at the United Nations Refugee Agency Bangkok (21 Mar 2017), in Taiwan (5 Aug 2019) and Australia (25 Dec 2019 to 9 Jan 2020): [1] https://tdtemcerts.wordpress.com/ [2] https://tdtemcerts.blogspot.sg/ [3] https://www.scribd.com/user/270125049/Teo-En-Ming -----END EMAIL SIGNATURE-----

3 2

How to msg_apply_changes in a branch route?
by Anthony Alba 01 Dec '20

01 Dec '20

Is there a way to obtain the effect of msg_apply_changes in a branch route? I want to do: route[BRANCHMANAGE] { rtpengine_manage() msg_apply_changes() // further mangle the SDP with textops for // obtuse UAs } The reason for this, is that after forking, I have some UAs that are extremely picky about SDP, and I need to mangle the body to make them happy. (4000 char limit, 68 attribute limit on SDP body - ever seen that? Compared with the absolutely gigantic OFFERs from WebRTC signalling libraries...) Now rtpengine_manage() and textops are working on separate copies of the msg body and the results don't stack correctly. Cheers Anthony Alba

2 1

vscode-kamailio-syntax - issue with empty comment line
by Chaigneau, Nicolas 01 Dec '20

01 Dec '20

Thanks for the link. One issue I've noticed: If you have an empty comment line (just # on a single line), then the next line is wrongly highlighted as comments. For example: # loadmodule "db_postgres.so" > i'm using https://github.com/miconda/vscode-kamailio-syntax in VScode. > its great! 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

Re: [SR-Users] [sr-dev] kamailio SIP and RTP proxy
by Henning Westerholt 01 Dec '20

01 Dec '20

Hi Patrick, better to contact our sr-users list with the usage related questions, added to CC. Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further. Feel free to ask again on sr-users after you have got more details. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-dev <sr-dev-bounces(a)lists.kamailio.org> On Behalf Of Patrick Leybag Sent: Wednesday, November 25, 2020 6:26 AM To: sr-dev(a)lists.kamailio.org Subject: [sr-dev] kamailio SIP and RTP proxy Hi, Can someone help me? I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work. Any help is much appreciated. Thank you in advance

1 0

Kamailio Failover
by Rolly Casuga Jr. 01 Dec '20

01 Dec '20

Hi, I am having a problem on fail overing my master kamailio server and slave kamailio server. I use kamailio as a load balancer for my asterisk servers. And it is working with calls, but when I added failover with keepalived to the slave, there is no audio during calls. Thank you, Rolly

2 1

Kamailio behind HAProxy
by Joey Golan 01 Dec '20

01 Dec '20

Hello, I have a kamailio server running behind HAProxy with proxy protocol v2 enabled. In Kamailio I have set the parameter tcp_accept_haproxy=yes and loaded tcpops module. UEs are registered using TLS and kamailio sees that the message has received from their real ip address + port and not HAProxy ip + port. When UE A calls UE B, kamailio is trying to reach UE B using his real ip address and port instead of HAProxy IP address + port. I know I can get the tcp ip and port of HAProxy using $tcp(c_si) and $tcp(c_sp) but I can’t make it work. What is the right way to do this? How should I use these variables properly in order to establish the call successfully? Thanks, Joey.

6 14

core dump error when running under docker
by Carlos Vicente 30 Nov '20

30 Nov '20

Greetings, I'm having a "fun" time trying to run Kamailio inside a Docker container. It keeps failing with: 0(238) DEBUG: <core> [core/mem/pkg.c:108]: pkg_print_manager(): pkg - using memory manager: q_malloc 0(238) DEBUG: <core> [core/mem/shm.c:300]: shm_print_manager(): shm - using memory manager: q_malloc Illegal instruction (core dumped) Unfortunately the core dump is nowhere to be found. I have tried to follow the instructions at: https://www.kamailio.org/wiki/tutorials/troubleshooting/coredumpfile but that hasn't helped. The environment is as follows: * Host hardware: Raspberry Pi 4 (4G memory) * Host OS: Ubuntu 20.04.1 LTS * Docker version 19.03.13 * Docker base image: arm32v7/ubuntu root@2ef8a21534fa:/# kamailio -v version: kamailio 5.3.2 (arm/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-NOSMP, 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 9.2.1 The config file passes the syntax check: kamailio -c -f /etc/kamailio/kamailio.cfg and, the most interesting part, if I run Kamailio on the host machine, _with the exact same config file_, it runs fine. I'd appreciate any clues you may have. Thanks. cv

4 10

What is the lifecycle of Branch Flag (setbflag)
by me.projects＠yahoo.com 30 Nov '20

30 Nov '20

Hi, What is the lifecycle of Branch Flag? If its set in Request Route will it be available till end of Dialog or It will be available during Transaction only ? Thanks

3 5

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