October 2020 - sr-users

[Serusers] Digest Authentication with Tenor DX
by Ashutosh Kumar 29 Apr '25

29 Apr '25

Hi, Does someone have any idea/experience with Quintum DX radius server's support for digest mode authentication. The Access-Request packets sent by SER are getting an error back. Guide. Regards, Ashutosh Kumar Chetu, Inc. Ph : 1(305) 402 6724 - Witin US Ph : 91 120 5323340 - Outside US Fax:1 (305) 832 5987 For more information, please visit http://www.chetu.com

2 1

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

4 10

Dialog and DMQ on restart
by Björn Klasen 25 Mar '24

25 Mar '24

Hi everybody, I'm just testing Kamailio 5.4.1 with dialog replication over DMQ. This seems to work very good. Dialogs are replicated without problems. When I'm restarting one node I would have expected, that all dialogs are synced again, just like in dmq_usrloc. But this does not happen. After a restart the nodes dialog-list is empty. Did I miss somethin? Is there a special parameter that I have to set? BR, Björn -- Björn Klasen, Specialist TNG Stadtnetz GmbH, Network Management (VoIP) Projensdorfer Straße 324 24106 Kiel Germany T +49 431/ 530530 F +49 431/ 7097-555 mailto: bklasen(a)tng.de http://www.tng.de Register: Amtsgericht Kiel HRB 6002 KI Executive board (Geschäftsführung): Dr.-Ing. Volkmar Hausberg, Sven Schade, Carsten Tolkmit, Dr. Sven Willert Tax-Id (Steuernr.): 2029047020, VAT-Id (USt-Id): DE225201428

4 9

Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

5 19

Re: [SR-Users] t_drop_replies not working with t_suspend in failure route
by Huber Andreas 03 Sep '21

03 Sep '21

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ 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

2 1

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

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

issues using evapi module
by David Escartin 26 Nov '20

26 Nov '20

Dear all We are trying to use the evapi module to send some data to an external application but I'm having problems getting the clients connected. I have the kamailio (version 5.3) running with a tcp socket 127.0.0.1:8228, and the evapi params are just modparam("evapi", "workers", 4) modparam("evapi", "netstring_format", 0) modparam("evapi", "bind_addr", "127.0.0.1:8448") modparam("evapi", "max_clients", 32) I tried a different number of workers and netstring_format 1 too. When I start the kamailio i added some debug to the code, and seems when doing the mod init of the evapi dispatcher 38(4779) DEBUG: <core> [core/sr_module.c:779]: init_mod_child(): idx 38 rank -2: evapi [EvAPI Dispatcher] it reaches to while(1) { ev_loop (loop, 0); } at evapi_run_dispatcher function. I guess if I connected to the tcp socket and sent some event, I would see the client accepted and the event route evapi:connection-new would be triggered. But i'm not able to do that. I tried to use the prime option, a tcp input client connection from logstash, so i could relay the data to the logstash using the evapi relay, but i only see the tcp socket being created but no client accepted. I also tried to connect with an erlang gen_tcp client, but it's the same i only see 47(4798) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 47(4798) DEBUG: <core> [core/tcp_main.c:1174]: tcpconn_new(): on port 54537, type 2, socket 105 47(4798) DEBUG: <core> [core/tcp_main.c:1497]: tcpconn_add(): hashes: 1117:1187:1505, 1 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 and if i try to send any data 47(4798) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xad0880, 105, -1, 0x0) fd_no=54 called 47(4798) DEBUG: <core> [core/tcp_main.c:4456]: handle_tcpconn_ev(): sending to child, events 1 47(4798) DEBUG: <core> [core/tcp_main.c:4129]: send2child(): selected tcp worker idx:0 proc:43 pid:4791 for activity on [tcp:127.0.0.1:8448], 0x7fc211712d58 43(4791) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7fc211712d58, fd=39 43(4791) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xb3c720, 39, 2, 0x7fc211712d58), fd_no=1 43(4791) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0xb3c720, 39, -1, 0x10) fd_no=2 called 43(4791) DEBUG: <core> [core/tcp_read.c:1671]: release_tcpconn(): releasing con 0x7fc211712d58, state 1, fd=39, id=1 ([127.0.0.1]:54537 -> [127.0.0.1]:8448) 43(4791) DEBUG: <core> [core/tcp_read.c:1672]: release_tcpconn(): extra_data (nil) 47(4798) DEBUG: <core> [core/tcp_main.c:3559]: handle_tcp_child(): reader response= 7fc211712d58, 1 from 0 47(4798) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0xad0880, 105, 2, 0x7fc211712d58), fd_no=53 47(4798) DEBUG: <core> [core/tcp_main.c:3686]: handle_tcp_child(): CONN_RELEASE 0x7fc211712d58 refcnt= 1 and when i try to send any data 38(10867) DEBUG: evapi [evapi_dispatch.c:610]: evapi_recv_notify(): received [0x7f17d23fc628] [{"test" : "1.1.1.1", "uuid" : "1-31629(a)3.3.3.3" , "pdd" : "4"}] (75) 38(10867) DEBUG: evapi [evapi_dispatch.c:316]: evapi_dispatch_notify(): the message was sent to 0 clients I don't know what i'm missing, or if i'm understanding the use of the module correctly could you please take a look? thanks a lot David -- [image: Logo] David Escartín Almudévar VoIP/Switch Engineer descartin(a)sonoc.io *SONOC* C/ Josefa Amar y Borbón, 10, 4ª · 50001 Zaragoza, España Tlf: +34 917019888 · www.sonoc.io

2 5

Add Reason Header to kamailio generated CANCEL
by Alfonso Pinto 18 Nov '20

18 Nov '20

Hi everyone, I've got a specific case: when the inv_fr times out, I need to add a Reason header to the CANCEL generated by kamailio. I've tried to see if I could do it in the onsend_route, but that one is not triggered for the generated CANCEL. I also checked event_route[tm:local-request], but that one isn't triggered either for the generated CANCEL. Is there any way to do it? Or maybe to have any pointer about where to look in the code so I may try to trigger event_route[tm:local-request] for these generated CANCELs? Regards, Alfonso

6 17

tsilo - failure to add branches (-1)
by Jurijs Ivolga 05 Nov '20

05 Nov '20

Hi, I have several Kamailio proxies - loadbalancer which is used as TLS offload and Authorization server and behind registrar servers. In this case I can't make tsilo work on the registrar server. I'm always getting: "tm [t_append_branches.c:172]: t_append_branches(): failure to add branches (-1)" If I use just one Kamailio which is used as Authorization and Registrar server then tsilo works as expected. Looks like kamailio where I run ts_append tries to connect to UAC directly and not through Loadbalancer. Any ideas? How can I troubleshoot this? Thank you! Jurijs

3 9

htable + event_routes + mod-init + htable:expired
by Brandon Armstead 02 Nov '20

02 Nov '20

Running 5.5.0-dev3 modparam("htable", "htable", "ipban=>size=8;autoexpire=300;dmqreplicate=1;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("htable", "timer_procs", 4) modparam("htable", "timer_interval", 5) modparam("htable", "db_expires", 1) modparam("htable", "htable", "account=>size=4;dmqreplicate=1;") modparam("htable", "htable", "server=>size=4;autoexpire=15;") I'm attempting to retrieve an auth token in the init process and also refresh the token upon expiration in the htable:expired event. I'm seeing some behavior where every other execution of htable:expired the variable $sht(server=>auth::token) is set per kamcmd htable.dump server, however xinfo() reports that the token was retrieved in the variable $sht (server=>auth::token). Please see remaining event route config below: event_route[htable:mod-init] { # generate auth token into http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); } event_route[htable:expired:server] { # process expired htable, renew auth token xinfo("AUTH_TOKEN_EXPIRED, lets retrieve a new one"); http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::time) = $TS; $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); xinfo("AUTH_TOKEN_TIME, $sht(server=>auth::time)"); } I've tried both with locking and unlocking. Also one last thing worth mentioning is that on the alternation where kamcmd htable.dump server shows no auth token, $sht(server=>auth::time) is available, when the auth token is visible in kamcmd htable.dump server there is no sht(server=>auth::time) returned. Also just to be explicitly clear -- xinfo() always returns AUTH_TOKEN_RECEIVED correctly in both event routes. Perhaps I'm over looking something -- thank you for the help in advance. - Brandon

2 3

Releasing Kamailio v5.3.7
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, I am planning to release Kamailio v5.3.7 sometime next week, likely on Monday or Tuesday, this being the usual prior notice for heads up to identify issues not reported yet on bug tracker as well as the commits you are aware of that were not backported yet. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Confusion about TCP worker ports
by Noah Mehl 30 Oct '20

30 Oct '20

Hey all, I’m a little stuck on an implementation of a set of dispatchers via TCP. There are some oddities about the behavior of the TCP source port of the Kamailio tcp worker/s, and maybe this is expected, but it doesn’t seem valid. For instance, I have a dispatcher: "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": “sip:2.2.2.2:5060;transport=tcp", "FLAGS": "AP", "PRIORITY": 5 } }] } }] But when Kamailio sends an OPTIONS keep alive, the source port for the worker is 33940, and not 5060 (which is the TCP listen port), as received by Freeswitch: recv 447 bytes from tcp/[1.1.1.1]:33940 at 18:58:24.958720: ------------------------------------------------------------------------ OPTIONS sip:2.2.2.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bK1525.80a9e442000000000000000000000000.0 To: <sip:2.2.2.2:5060;transport=tcp> From: <sip:inbound-kamailio-01>;tag=3c52ba62ee4c4621b9660728159919d3-cda8066f CSeq: 10 OPTIONS Call-ID: 3aa18693487268dc-2790(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.2 (x86_64/linux)) ------------------------------------------------------------------------ Also, I get weird debug messages when this tcp worker is spun up (specifically about Resource temporarily unavailable): 11(2790) DEBUG: dispatcher [dispatch.c:3340]: ds_ping_result_helper(): probe all, mode DS_PROBE_ALL 11(2790) DEBUG: dispatcher [dispatch.c:3383]: ds_ping_set(): probing set #1, URI sip:2.2.2.2:5060;transport=tcp 11(2790) DEBUG: dispatcher [dispatch.c:3414]: ds_ping_set(): Default ping_from: sip:inbound-kamailio-01 11(2790) DEBUG: dispatcher [dispatch.c:3424]: ds_ping_set(): Default outbound proxy: 11(2790) DEBUG: tm [uac.c:450]: t_uac_prepare(): next_hop=<sip:2.2.2.2:5060;transport=tcp> 11(2790) DEBUG: tm [uac.c:158]: dlg2hash(): hashid 21073 11(2790) DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one 11(2790) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 2.2.2.2 11(2790) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5060, type 2, socket -1 11(2790) DEBUG: <core> [core/tcp_main.c:1494]: tcpconn_add(): hashes: 2712:0:0, 1 11(2790) DEBUG: <core> [core/tcp_main.c:2886]: tcpconn_1st_send(): pending write on new connection 0x7f24e64c1e18 sock 8 (-1/447 bytes written) (err: 11 - Resource temporarily unavailable) 11(2790) DEBUG: tm [uac.c:678]: send_prepared_request_impl(): uac: 0x7f24e65285a8 branch: 0 to 2.2.2.2:5060 11(2790) DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring 27(2806) DEBUG: <core> [core/tcp_main.c:3792]: handle_ser_child(): read response= 7f24e64c1e18, 5, fd 46 from 11 (2790) 27(2806) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0x56490f0f8060, 46, 2, 0x7f24e64c1e18), fd_no=37 27(2806) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0x56490f0f8060, 46, 0x1, 0xffffffff) fd_no=38 called 27(2806) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x56490f0f8060, 46, -1, 0x0) fd_no=38 called 27(2806) DEBUG: <core> [core/tcp_main.c:4457]: handle_tcpconn_ev(): sending to child, events 1 27(2806) DEBUG: <core> [core/tcp_main.c:4127]: send2child(): selected tcp worker idx:0 proc:19 pid:2798 for activity on [tcp:1.1.1.1:5060], 0x7f24e64c1e18 19(2798) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7f24e64c1e18, fd=8 19(2798) DEBUG: <core> [core/tcp_read.c:1548]: tcp_read_req(): content-length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): SIP Reply (status): 19(2798) DEBUG: <core> [core/parser/msg_parser.c:622]: parse_msg(): version: <SIP/2.0> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:624]: parse_msg(): status: <200> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:626]: parse_msg(): reason: <OK> 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1525.80a9e442000000000000000000000000.0>; state=6 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <33940>; state=16 19(2798) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 19(2798) DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 19(2798) DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1mB9HryQ8ZBFc 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 19(2798) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [59]; uri=[sip:2.2.2.2:5060;transport=tcp] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (39)[<sip:2.2.2.2:5060;transport=tcp>], to tag (13)[1mB9HryQ8ZBFc] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS> 19(2798) DEBUG: <core> [core/receive.c:319]: receive_msg(): --- received sip message - reply - call-id: [3aa18693487268dc-2790(a)1.1.1.1] - cseq: [10 OPTIONS] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Are these SIP messages expected to come from other ports than the listen port (5060 in this case)? Also, if the worker source port is not 5060, shouldn’t the SIP message get updated with the correct port? In the case of OPTIONS, Freeswitch is replying correctly to the source port: 33940. However, in the case of an in dialog BYE, Freeswitch is NOT replying to the source port of the Kamailio messages, but only to port 5060. Here is an example (relayed from web sockets to freeswitch over TCP) INVITE (as received from Freeswitch): recv 1481 bytes from tcp/[1.1.1.1]:33940 at 16:56:47.920698: ------------------------------------------------------------------------ INVITE sip:991012@sip.domain <sip:991012@sip.domain>.com SIP/2.0 Record-Route: <sip:1.1.1.1;transport=tcp;r2=on;lr;nat=yes> Record-Route: <sip:1.1.1.1:5061;transport=tls;r2=on;lr;nat=yes> Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bKd408.3f53e940ccb20c1033df4b3a8ebd8a34.0;i=1 Via: SIP/2.0/TLS 172.22.199.110:55304;received=5.5.5.5;rport=39518;branch=z9hG4bKPj5Css6JomCt9Cli2cCINbXi4FbPM5wewG;alias Max-Forwards: 69 From: "Noah Mehl" <sip:5135555555@inbound-jail>;tag=s3i3y2tiOCgnUId5TD4Vp0UChf9GyEy9 To: <sip:991012@inbound-jail> Contact: <sip:74895612@172.22.199.110:54887;transport=tls;alias=5.5.5.5~39518~3> Call-ID: 5aoRBMBHahxqSLzrIpFnlfRz.UcGsmfq CSeq: 27271 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink Pro 4.6.0 (MacOSX) Content-Type: application/sdp Content-Length: 528 v=0 o=- 3812979407 3812979407 IN IP4 5.5.5.5 s=Blink Pro 4.6.0 (MacOSX) t=0 0 m=audio 50016 RTP/SAVP 113 0 101 c=IN IP4 5.5.5.5 a=rtcp:50017 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UhHq6hth9HqALmiJ3AEeoGkixObBzkLURG60wJKT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:VKYaSCVwgvXCPaRvudTrgLORhWmOA7wyDJVeGjcu a=sendrecv a=oldmediaip:172.22.199.110 a=oldmediaip:172.22.199.110 ------------------------------------------------------------------------ This doesn’t seem valid, as the top Via doesn’t have a port specified? For reference, just rebuilt form the 5.4 branch today: commit 62dff5b8b157236cae7defe64291a6e4a8ae27b5 (upstream/5.4) Author: Kamailio Dev <kamailio.dev(a)kamailio.org> Date: Wed Oct 28 20:16:28 2020 +0100 modules: readme files regenerated - modules ... [skip ci] Thanks! ~Noah

6 19

Double port in TO header
by David Villasmil 30 Oct '20

30 Oct '20

Hello guys, This happened to me a long time ago but i totally forgot the reason. I using a simple dispatcher, and when relaying out to the destination, kamailio is adding an additional :5060 port to the To header: To: <sip:910609738@172.31.6.1:5080:5060> The Destination is actually 172.16.6.1:5080, i don't know why Kamailio is adding that :5060 The dispatcher is if (!ds_select_dst($var(dispatcher_outbound), "4")) { xlog("[DISPATCH]: ds_select_dst FAILED!\n"); send_reply("404", "No destination"); exit; } else { xlog("[DISPATCH]: ds_select_dst was succesful\n"); } xlog("L_DBG", "[DISPATCH]: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Any ideas? I totally forgot... and it's late, i guess... Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 8

Re: [SR-Users] Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 30 Oct '20

30 Oct '20

Perfect, that's what I was looking for. :) Thanks! > This is what I have always done: > > # export PATH=$PATH:/usr/pgsql-${MAJOR_VER}/bin > # cd src/modules/db_postgres > # make install > > The `pg_config` utility in /usr/pgsql-${MAJOR_VER}/bin ensures that the proper header include path is supplied. > > -- Alex 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.

1 0

Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 29 Oct '20

29 Oct '20

Hello, I'd like to build Kamailio module db_postgres, with the PostgreSQL headers and libs in a non-standard location. I'm trying to figure out how to do that, if it's possible, without having to edit db_postgres/Makefile. I see in the Makefile that it's trying to find the files relative to $(LOCALBASE): # use standard know paths # libpq-fe.h locations DEFS +=-I$(LOCALBASE)/include -I$(LOCALBASE)/pgsql/include \ -I$(SYSBASE)/include/pgsql -I$(SYSBASE)/include/postgresql \ -I$(SYSBASE)/include/postgresql/8.0 LIBS +=-L$(LOCALBASE)/lib -L$(LOCALBASE)/pgsql/lib \ -L$(LOCALBASE)/lib/pgsql -lpq However, LOCALBASE (which defaults to /usr/local) is used for many things other than PostgreSQL, so changing that is not a great idea, I think... Any idea ? 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

www.kamailio.org maintenance on 29th October 2020, 10:00 CET
by Henning Westerholt 29 Oct '20

29 Oct '20

Hello, there will be a short planned maintenance on www.kamailio.org<http://www.kamailio.org> on 29th October 2020 at 10:00 CET. Relevant services including main homepage and documentation wiki will be unavailable for about 10-15 minutes. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio unresponsive with Dialog+DMQ
by Patrick Wakano 28 Oct '20

28 Oct '20

Hello list, Hope all are doing well! We are running load tests in our Kamailio server, that is just making inbound and outbound calls and eventually (there is no identified pattern) Kamailio freezes and of course all calls start to fail. It does not crash, it just stops responding and it has to be killed -9. When this happens, SIP messages are not processed, dmq keepalive fails (so the other node reports as down), dialog KA are not sent, but Registrations from UAC seem to still go out (logs from local_route are seen). We don't have a high amount of cps, it is max 3 or 4 per sec, and it gets around 1900 active calls. We are now using Kamailio 5.2.8 installed from the repo on a CentOS7 server. Dialog has KA active and DMQ (with 2 workers) is being used on an active-active instance. >From investigation using GDB as pasted below, I can see UDP workers are stuck on a lock either on a callback from t_relay... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e95ffb0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_UPDATE, dlg=0x7ffb362ea3c8, needlock=1, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a61f28e in dlg_on_send (t=0x7ffb36c98120, type=16, param=0x7fff2e9601e0) at dlg_handlers.c:739 #7 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36c98198, type=16, trans=0x7ffb36c98120, params=0x7fff2e9601e0) at t_hooks.c:260 #8 0x00007ffb2ef286d0 in run_trans_callbacks (type=16, trans=0x7ffb36c98120, req=0x7ffb742f27e0, rpl=0x0, code=-1) at t_hooks.c:287 #9 0x00007ffb2ef38ac1 in prepare_new_uac (t=0x7ffb36c98120, i_req=0x7ffb742f27e0, branch=0, uri=0x7fff2e9603e0, path=0x7fff2e9603c0, next_hop=0x7ffb742f2a58, fsocket=0x7ffb73e3e968, snd_flags=..., fproto=0, flags=2, instance=0x7fff2e9603b0, ruid=0x7fff2e9603a0, location_ua=0x7fff2e960390) at t_fwd.c:381 #10 0x00007ffb2ef3d02d in add_uac (t=0x7ffb36c98120, request=0x7ffb742f27e0, uri=0x7ffb742f2a58, next_hop=0x7ffb742f2a58, path=0x7ffb742f2e20, proxy=0x0, fsocket=0x7ffb73e3e968, snd_flags=..., proto=0, flags=2, instance=0x7ffb742f2e30, ruid=0x7ffb742f2e48, location_ua=0x7ffb742f2e58) at t_fwd.c:811 #11 0x00007ffb2ef4535a in t_forward_nonack (t=0x7ffb36c98120, p_msg=0x7ffb742f27e0, proxy=0x0, proto=0) at t_fwd.c:1699 #12 0x00007ffb2ef20505 in t_relay_to (p_msg=0x7ffb742f27e0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:334 or loose_route... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e9629d0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_STATE, dlg=0x7ffb363e0c10, needlock=0, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a62b3bf in dlg_onroute (req=0x7ffb742f11d0, route_params=0x7fff2e962ce0, param=0x0) at dlg_handlers.c:1538 #7 0x00007ffb2e7db203 in run_rr_callbacks (req=0x7ffb742f11d0, rr_param=0x7fff2e962d80) at rr_cb.c:96 #8 0x00007ffb2e7eb2f9 in after_loose (_m=0x7ffb742f11d0, preloaded=0) at loose.c:945 #9 0x00007ffb2e7eb990 in loose_route (_m=0x7ffb742f11d0) at loose.c:979 or t_check_trans: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2a5ea9c6 in futex_get (lock=0x7ffb35e78804) at ../../core/futexlock.h:108 #2 0x00007ffb2a5f1c46 in dlg_lookup_mode (h_entry=1609, h_id=59882, lmode=0) at dlg_hash.c:709 #3 0x00007ffb2a5f27aa in dlg_get_by_iuid (diuid=0x7ffb36326bd0) at dlg_hash.c:777 #4 0x00007ffb2a61ba1d in dlg_onreply (t=0x7ffb36952988, type=2, param=0x7fff2e963bf0) at dlg_handlers.c:437 #5 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36952a00, type=2, trans=0x7ffb36952988, params=0x7fff2e963bf0) at t_hooks .c:260 #6 0x00007ffb2ef286d0 in run_trans_callbacks (type=2, trans=0x7ffb36952988, req=0x7ffb3675c360, rpl=0x7ffb742f1930, code=200) at t_hooks.c:28 7 #7 0x00007ffb2ee7037f in t_reply_matching (p_msg=0x7ffb742f1930, p_branch=0x7fff2e963ebc) at t_lookup.c:997 #8 0x00007ffb2ee725e4 in t_check_msg (p_msg=0x7ffb742f1930, param_branch=0x7fff2e963ebc) at t_lookup.c:1101 #9 0x00007ffb2eee44c7 in t_check_trans (msg=0x7ffb742f1930) at tm.c:2351 And the DMQ workers are here: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1d6c81 in futex_get (lock=0x7ffb35217c34) at ../../core/futexlock.h:108 #2 0x00007ffb2b1d7c3a in worker_loop (id=1) at worker.c:86 #3 0x00007ffb2b1d5d35 in child_init (rank=0) at dmq.c:300 Currently I will not be able to upgrade to latest 5.4 version to try to reproduce the error and since 5.2.8 has already reached end-of-life, maybe is there anything I can do on the configuration to avoid such condition? Any ideas are welcome! Kind regards, Patrick Wakano

3 6

Kamailio v5.4.2 Released
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, Kamailio SIP Server v5.4.2 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. 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 v5.4.1. For more details about version 5.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/10/kamailio-v5-4-2-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

Releasing Kamailio v5.4.2
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, I am considering to release Kamailio v5.4.2 sometime next week, likely on Tuesday, October 27, 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

KEMI and Native mixing
by Marcin Kowalczyk 26 Oct '20

26 Oct '20

Hi, Is it possible to mix both KEMI (lua) and Native configs in one kamailio instance? So some blocks are called from native script and some others from KEMI (lua)? Marcin

6 5

Kamailio-5.x and Asterisk16.X Integration Walkthrough Videos or Wikis
by Lorem Ipsum 26 Oct '20

26 Oct '20

Hi there, After reading about Kamailio, i like this project but i am having trouble finding the wikis or step by step guide to successfully integrate asterisk with kamailio for simple use case where phones register to the asterisk pbx via kamailio and make/receive calls to ITSP, there is no proper documentation on 2 products integration, really appreciated your guidance and if somebody could provide me the step by step guide. Looking forward to the guides.

2 1

I want to write function in group module but i could not
by ahmed moghazy 25 Oct '20

25 Oct '20

I have 2 function first If(is_user_in(“Request-URI”, ”ld”)){ ..... } So i dont know what to do , second If(get_user_group(“Request-URI”, ”$avp(i:10)”){ ... } I put users in 2 groups but they could call each others but i dont want that . Any help please , Thank you.

1 0

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 22 Oct '20

22 Oct '20

Hi Duarte, I'm not sure if your situation is more complicated than mine. I have two dispatcher nodes in the HA setup, with real IPs say IPa, IPb. They share a floating/virtual IP, let's call it VIP. Both nodes listen to the VIP (not its real IP), and use the VIP as the dispatcher's socket. In this case the master node which has the VIP can always send OPTIONS pings fine. My problem was with the backup node, which doesn't have the VIP so fails to send OPTIONS pings. Joel Serrano gave a good suggestion: both nodes start up with polling disabled, and only when a node turns master it enables the polling. This has solved my problem. Cheers, Yufei

1 0

Incoming DID calls routing to subscriber
by Jose N Figueroa 22 Oct '20

22 Oct '20

Hello everybody, I'm trying to route incoming calls to kamailio with specific subscriber but I can't make it working. The complete scenario is: SoftSwith -> kamailio 5.3 -> zoiper What i'm trying to do is to receive a call into the zoiper when I dial specific DID which is already coming into the kamailio. It goes until the routing from kamilio to the subscriber account but it keeps in INVITE command, no response from the subscriber. Any help would be great! This is what I've been trying but no luck yet: if($rdir(name)=="downstream") { xlog("DID request was incoming call, called number: $rU caller_id: $fU and source_ip $si \n"); $sht(rtpProxy=>mip_$ci)=1; # We should take the called_number as username in table # Also validate if the number has active that DID, balance if needed, etc. # route(validate_customer); # I know, hardcoded for testing sql_xquery("ca", "select * from location where username = 1212 ", "ra"); $avp(location_addr) = $xavp(ra=>received); xlog("DID did_incoming location is $xavp(ra=>received) \n"); # Validate SIP OK #setflag(SIP_IP_OK); #$rU=$avp(dnis); #$rd=$avp(identity); #$rp=$avp(port); # uac_replace_to($xavp(ra=>contact)); uac_replace_to("sip:1212@voip.x.net:5060"); # $ru = $var(duri); $ru=$xavp(ra=>contact); $ru="sip:1212@voip.x.net:5060"; route(RELAY); # t_on_failure("dst_failure"); # t_on_reply("dst_reply"); if(!t_relay()) { sl_reply_error(); exit; } exit; } -- Regards, Jose N Figueroa

3 3

Load balance on HA scenario
by Duarte Rocha 21 Oct '20

21 Oct '20

Greetings, I have two machines with Kamailio in a HA setup with replicated DB. For simplicity let's say each machine has one HA IP and that IP can jump to the other machine in case something happens (kamailio stopping, etc). I'm using Dispatcher with load balance configuration. I have Dispatcher configured so that each peer has one instance for socket with HA IP 1 and HA IP 2. In order for this to work correctly on the load balance scenario I must disable via RPC command the peer which has the socket that doesn't belong to the machine. I also must do it every time the IP jumps back and forth, which adds complexety to my system. Does Dispatcher has any sort of help on this? It could not include peers with sockets IPs that don't belong to the machine in the destination set, for example. Is this possible? I could also work with failover support but i would rather avoid having so many failovers. Best Regards,A

3 8

kamailio RTP Engine in kubernetes pods
by Pavithra Mohanraja 21 Oct '20

21 Oct '20

Hi, I have installed rtp engine inside pcscf pod and tried to make a call. caller was unable to reach the pod environment with 2223 port exposed for rtp. Kindly anyone help, if you have already done with pod based deployment of ims. Thanks, Pavithra

4 7

Kamailio + rtpengine
by Amit Pal 21 Oct '20

21 Oct '20

Hi, I have using kamailio with rtpengine, I have started rtpengine session (rtpengine_offer() /rtpengine_answer() )only when call getting hold and then play media file and on un hold detach rtpengine session(rtpengine_delete()). during this activity I have received the log "No supported output codec found in SDP" from rtpengine. And getting silence on other leg. Above activity log are attach in mail. if we do same step again (i.e. hold/unhold) in same call then second leg listening the file perfectly and rtpengine working fine vice-versa. I want to know , what else missing to configure here. Pls guide me to get out this. Regards Amit Pal

2 1

How to read "duid" from dispatcher.list in kamamilio.cfg
by Jani Biju Babjan 21 Oct '20

21 Oct '20

Hi, I have a dispatcher.list file like the following: 1 sip:127.0.0.1:5060 0 0 duid=abcd;maxload=5 1 sip:127.0.0.2:5060 0 0 duid=efgh;maxload=5 After calling the ds_select_dst("1", "4") function(ie, round-robin selection), I want to read the value of the "duid" attribute of the selected destination and then pass it to a database. How can I read this value in the configuration file? Jani Biju Babjan

2 2

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 21 Oct '20

21 Oct '20

Hi Joel, Thanks for the tip! Having the pings disabled and enable it on HA status being master sounds like a good idea :) Cheers, Yufei

1 0

ERROR: <core> [core/socket_info.c:2225]: parse_protohostport():
by Andy Kama 20 Oct '20

20 Oct '20

I am seeing this error in the logs i am using kamailio 5.4.1 any idea what the cause is? ERROR: <core> [core/socket_info.c:2225]: parse_protohostport(): bad protocol in sip:myp:5060

2 4

Re: [SR-Users] Load balance on HA scenario
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Duarte, Not sure if I understand it correctly but are you trying to set up an HA pair of dispatchers? If so I am doing it differently in that, I use the Virtual/floating IP (VIP) as socket. This way the master node (which has got the VIP) can send OPTIONs pings out fine but the backup node won't be able to send. Functionally it is fine, but the backup node keeps logging errors because it doesn't have the VIP: ERROR: <core> [core/udp_server.c:599]: udp_send(): sendto(sock, buf: 0x7f0e6b54e2e0, len: 372, 0, dst: (192.168.213.143:5060), tolen: 16) - err: Network is unreachable (101) ERROR: tm [../../core/forward.h:231]: msg_send_buffer(): udp_send failed When FO happens, the backup node becomes master and takes the VIP, and it functions correctly. But I haven't worked out how to avoid the ERROR loggings on the backup node. Maybe someone else has any better ideas/suggestions? Cheers, Yufei

2 1

No retry on INVITE
by Allan Kristensen 20 Oct '20

20 Oct '20

Hello, I'm trying to get Kamailio to retry INVITE. We have a Freeswitch which is sending INVITE to a sip user via Kamailio proxy using static ip/location. All is well, calls are flowing, but sometimes INVITE from kamailio to user is lost and no attempt to retry is occurring and so after 30 sec kamailio just returns 408 Request timeout. Call flow is like this: FS Kamailio User --- INVITE --> <-- 100 Trying --- INVITE ---> (Waiting 30 secs) <-- 408 Timeout --- ACK ---> So assume we have failed in handling retries somehow. Kamailio is running stateless, but other retries are working, so I assume this is not a problem? Can someone perhaps point us in the right direction on where to solve this? Kamailio is version 5.2.4 Thank you... /Allan

3 4

Re: [SR-Users] SRTP
by Yufei Tao 20 Oct '20

20 Oct '20

Hi, Like Olle said RTPEngine is perfectly capable of handling one leg in RTP the other in SRTP. When looking up RTPEngine's documentation, may be worth checking these out: SRTP SDES-unencrypted_srtp SDES-unencrypted_srtcp SDES-unauthenticated_srtp e.g. try apply these flags to rtpengin_offer() on the incoming RTP leg, to get RTPEngine to add encryption to the outgoing leg etc. Cheers, Yufei

1 0

Re: [SR-Users] Kamailio 5.2.5 - Need help with rtpengine error
by Yufei Tao 20 Oct '20

20 Oct '20

Hi Andrew, Like Alex said, have a look at the commands RTPEngine received and the responses it sent in the history of the call, and see if RTPEngine has ever received any commands regarding this call, or, if the call has been deleted from RTPEngine. You can raise the RTPEngine logging level to 6 for example so you can see the command/responses of RTPEngine. Cheers, Yufei

1 0

about registrar save()/lookup()
by Juha Heinanen 20 Oct '20

20 Oct '20

I didn't find any mention in registrar module README that save() function saves also branch flags and that lookup() retrieves them. Is that documented somewhere? -- Juha

3 5

dialog_active_dialogs erroneous value
by Sergiu Pojoga 20 Oct '20

20 Oct '20

Hi there, Every now and then, the value of active dialogs metric spikes to an erroneous huge number, messing up my otherwise beautiful grafana gauges :( Example: kamailio_dialog_active_dialogs 18446744073709551614 1602812239115 kamailio_dialog_early_dialogs 0 1602812239115 kamailio_dialog_expired_dialogs 2 1602812239115 kamailio_dialog_failed_dialogs 7361 1602812239115 Anybody faced the same symptoms, any clues what might be causing this? Thanks, --Sergiu

3 6

TOPOS module and fix_nated_contact
by Minh Phan 19 Oct '20

19 Oct '20

Hi List, I am using Topos module to hide network topology. I am having a problem of routing the ACK request for REINVITE. Here is the schema: A KAMAILIO B 1. --------- INVITE --------> 2. <---- 180 RINGING --- 3. <---- 200 OK --- 4. ---- ACK ---> 5. <--------- (RE) INVITE -- 6. --------- OK ---> 7. <---- ACK ----- (THIS ACK IS SENT TO PRIVATE IP OF A) I use TOPOS module along with fix_nated_contact as I would do normally for natted UAs. The first ACK (4) is routed correctly but not the second one (7), which is routed to the PRIVATE IP of A instead of routing to its public address. Has anyone run into this problem while using TOPOS module? How could I make them work together? Best Regards, Minh Phan

2 1

hold_on_music over P2P call [Kamailio]
by Amit Pal 19 Oct '20

19 Oct '20

Hi Team, Pls guide me on following point. :- By using kamailio (P2P call), want to play hold on music during hold call. Shall it be possible without any external media server like (RTPProxy/media proxy). Regards Amit Pal

2 1

Kamailio IMS RTP Engine
by Pavithra Mohanraja 19 Oct '20

19 Oct '20

Hi, I have followed below link to setup rtp engine. https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ RTP engine is started without any errors and it is enabled in pcscf also. I am using the config files of pcscf as per the above link .. still rtp packets are not reaching pcscf from UE . What could be the problem? The only difference is that i have deployed ims in kubernetes environment as pods. Kindly help. Thanks, Pavithra

2 2

Libphonenumber - Updating metadata
by Zach Nielsen 18 Oct '20

18 Oct '20

Hey all, I've been using the PHONENUM module, which uses Google's libphonenumber library, to perform number validation among other handy things within the route script. One problem I've run into however is the metadata in the latest stable (5.4) repo is pretty stale. I'm curious if anyone is running this module, and if so how they've gone about updating the metadata used by the underlying library. If I have to ditch the repo and compile everything manually, that's not a huge deal. Just hoping there's another way. Thanks! Zach

3 3

No trusted table found
by Jon Bonilla (Manwe) 17 Oct '20

17 Oct '20

Hi all Just installed a kamailio 5.4.1 for a simple proxy scenario. I created the version,address and trusted tables, and added some ips to the trusted via kamctl. If I execute "kamctl trusted show" I see the list of servers. But with "kamctl trusted dump" I receive { "jsonrpc": "2.0", "error": { "code": 500, "message": "No trusted table" }, "id": 13063 } when I start kamailio with debug I see these messages: b_do_submit_query(): submitted query: select `table_version` from `version` where `table_name`='trusted', result 0 my version table looks like this: +----+------------+---------------+ | id | table_name | table_version | +----+------------+---------------+ | 1 | version | 1 | | 2 | trusted | 6 | | 3 | address | 6 | +----+------------+---------------+ any hints? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

2 5

TLS connection from Chrome to Kamailio fails
by Juha Heinanen 17 Oct '20

17 Oct '20

White testing WebRTC, I noticed that TLS connection from Chrome browser 85.0.4183.121 to Kamailio's (master version) TLS listening port fails. Chrome sends Client Hello and K responds with Server Hello. After that Chrome terminates the handshake by sending TCP FIN. Connection setup from Firefox works fine. After receiving Server Hello, is sends Change Cipher Spec, etc. My K is using Let's Encrypt server certificate and is server tls_method is TLSv1.2+. Any ideas what goes wrong? Has anyone else experienced this? -- Juha

3 6

How to increase private & shared memory in kamailio Container?
by Hamid Hashmi 16 Oct '20

16 Oct '20

Dear Users How can I increase private memory in the Kamailio container? the service was started with the following command docker run -d --restart unless-stopped --network host -v /root/volumes/scscf/:/etc/kamailio --name scscf -e IP=10.161.224.90 -e INT_IP=10.161.224.90 10.161.224.196:5000/kamailio-ngvoice PS: I don't have a docker file to build the image again. Regards Hamid R. Hashmi

2 1

usrloc_dmq - Suddenly stopped usrloc replication
by José Seabra 16 Oct '20

16 Oct '20

Hello there, Suddenly the usrloc replication through DMQ stopped, but the interesting thing is that DMQ cluster status is OK, htable replication still working only usrloc_dmq stopped working. I increased the kamailio loglevel to 3 in order to see if there was some log related and then registered a peer, but the only DMQ logs that I see are related with notification_peer and htable replication, nothing related with usrloc_dmq. What I can do more here to try to understand what went wrong? I haven't restarted the cluster yet, because I would like to try to understand the root cause of this issue. Any advice would be appreciated Thank you -- Cumprimentos José Seabra

3 8

Kamailio 5.2.5 - Need help with rtpengine error
by Andrew Chen 15 Oct '20

15 Oct '20

Hi all, Today I saw the following errors reported by rtpengine module: Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) DEBUG: rtpengine [rtpengine.c:2453]: rtpp_function_call(): proxy reply: d6:result5:error12:error-reason15:Unknown call-ide Oct 15 15:02:10 ashprodkama52 kamailio[8028]: 30(8086) ERROR: rtpengine [rtpengine.c:2474]: rtpp_function_call(): proxy replied with error: Unknown call-id This was on the 200 OK response back from our Freeswitch. Any idea what would cause this? 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.*

2 2

Regd : SIP Configuration in Kamailio
by Santanu B 15 Oct '20

15 Oct '20

Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu

5 5

sdpops - modifying sdp codecs' presedence list - order
by Melek Oktay 15 Oct '20

15 Oct '20

Hello, Kamailio *sdpops *module consist of several useful functions for manipulating SDP on Proxy side such as *sdp_keep_codecs_by_name*() . This function *sdp_keep_codecs_by_name*("PCMU,PCMA,GSM") finds codecs id [0,8,3] and if incoming packet consist of other than these codecs, they are removed ! For example, assume incoming packet SDP --> [8 0 3 9 101] after applying this function 9 and 101 are removed and outgoing packet SDP consist only PCMU,PCMA,GSM [0,8,3]. This function does taking given codec *precedence/priority* into account. As you see above message SDP is 8 0 3, different priority than inputted 0 8 3. I need to change *order *of codecs too ! Would you help me to do it ? I couldn't find *add *operation on *sdpops *module Regards, Melek

1 0

Kamailio Relay Replies Issue
by me.projects＠yahoo.com 15 Oct '20

15 Oct '20

I have DID provider which is sending INVITES to Kamailio and kamailio has UAC module with asterisk to forward Calls to Asterisk with uac_auth() 1-DID Provider ----> (TLS)2-Kamailio (UDP) -----> (UDP) 3-Asterisk Call from Kamailio to Asterisk is forwarded and answered but NON of the reply received in "MANAGE_REPLY" route is not relayed back to DID provider call get disconnected. =================SIP DUMP INVITE FROM DID INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 70VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih -----------------------Invite to Asterisk with AUTH INVITE sip:+1700@KAMAILIOSERER.com:5061;user=phone;transport=tls SIP/2.0FROM: Alga Tech<sip:+17900000002@DIDPROVIDER.com:5061;user=phone>;tag=28989e96e28d47d891b4d8d21d85fa92TO: <sip:+1700@KAMAILIOSERER.com:5061;user=phone>CSEQ: 1 INVITECALL-ID: 936ba28f302f5a5d9b09f32e2e7c41f2MAX-FORWARDS: 69VIA: SIP/2.0/TLS DID-IP:5061;TH=div;branch=z9hG4bK77976715RECORD-ROUTE: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>CONTACT: <sip:DIDPROVIDER-CONTACT.com:443;x-i=b9483d0b-e3aa-4487-8dec-68de62d93a35;x-c=936ba28f302f5a5d9b09f32e2e7c41f2/d/8/26d69f6b729942a799552629a315d5d1>CONTENT-LENGTH: 1103MIN-SE: 300SUPPORTED: timerUSER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.10.12.5 i.USEA.0CONTENT-TYPE: application/sdpALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFYSESSION-EXPIRES: 3600TH: dih ------------------ >From Asterisk to Kamailio SIP/2.0 100 TryingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>CSeq: 1 INVITEServer: AsteriskContent-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskContact: <sip:66.232.107.106:7790>Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContent-Length: 0 ---------------------- >From Asterisk to Kamailio SIP/2.0 180 RingingVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Content-Length: 0TH: uch ---------------------- >From Asterisk to Kamailio SIP/2.0 200 OKVia: SIP/2.0/UDP KAMAILIO-IP:7790;TH=ucv;rport=7790;received=KAMAILIO-IP;branch=z9hG4bK7317.fe3491996f35f580f2efa1c50081a62c.1;i=7Via: SIP/2.0/TLS DID-IP:5061;rport=8256;TH=div;branch=z9hG4bK77976715Record-Route: <sip:KAMAILIO-IP:7790;lr;r2=on;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:KAMAILIO-IP:5061;transport=tls;r2=on;lr;ftag=28989e96e28d47d891b4d8d21d85fa92;vsf=AAAAABoEBwFwQUMeU1FcXjMDGx9bBF0XAQV1Yi5taWNyb3NvZnQuY29tOjUwNjE7dXNlcj1waG9uZQ--;nat=yes>Record-Route: <sip:DIDPROVIDER-ROUTE.com:5061;transport=tls;lr>Call-ID: 936ba28f302f5a5d9b09f32e2e7c41f2From: "Alga Tech" <sip:1508@qs.callsprout.com>;tag=28989e96e28d47d891b4d8d21d85fa92To: <sip:+1700@KAMAILIOSERER.com;user=phone>;tag=17f0e0ba-c1c0-457a-b0dd-9a9729d312e7CSeq: 1 INVITEServer: AsteriskAllow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFERContact: <sip:66.232.107.106:7790>Supported: 100rel, timer, replaces, norefersubSession-Expires: 3600;refresher=uacRequire: timerContent-Type: application/sdpContent-Length: 299TH: uch

1 0

To Header Manipulation
by Sam Ware 15 Oct '20

15 Oct '20

Wondering if this is intended behavior. We have a customer doing some craziness with the TO header. They are including their tech prefix in the TO header as well as the RURI. It doesn't cause any issue in our systems but apparently it does with one of our vendors. In an effect to prevent the issue, we thought we would check if the tech prefix was added on the TO header and remove it. Initially, I tried this by changing the $tU variable. $avp(tp_len) = $(avp(techprefix){s.len}) + 1; # Get the Length of the Tech Prefix and add 1 for the # or * $tU = $(tU{s.strip, $avp(tp_len)}); The result ended up with the duplication of the DNIS: sut <sip:911999# 1812555111118125551111@172.16.3.45:5060> Where 911999# was the tech prefix and 18125551111 is the DNIS which was still an issue for the vendor. Through additional reading, I found the suggestion to use the uac_replace_to function in the UAC module which I implemented as follows: $avp(new_to_hdr_uri) = $(tu{re.subst,/sip:(.*[\*#])(.*)/sip:\2/}); uac_replace_to("$avp(new_to_hdr_uri)"); This resulted in the original TO header username to be appended at the end of the uri: sut <sip:18125551111@172.16.3.45:5060911999#18125551111> Wondering if I am doing something wrong or this is the way these functions are designed to protect the TO header contents for Dialog matching. -- Sam D Ware

2 1

Error in RTP engine setup
by Pavithra Mohanraja 15 Oct '20

15 Oct '20

Hi, I tried to setup rtpengine to make my sip clients communicate through proxy following https://open5gs.org/open5gs/docs/tutorial/02-VoLTE-setup/ I started sip communication since the service was running and saw that the RTP packets go to exposed rtp interface instead of other sip client. But i don't receive packets from the other end. When analyzed I noticed the rtp-engine daemon gives a few errors. *modprobe: FATAL: Module xt_RTPENGINE not found in directory /lib/modules/4.15.0-118-generic* *iptables: No chain/target/match by that name.ip6tables: No chain/target/match by that name. * Starting RTP/media proxy ngcp-rtpengine-daemon [1602690253.130392] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED* *(Note: *uname -r gives 4.15.0-118-generic and /lib/modules/ has only 4.15.0-115-generic*)* Please help me with this issue. Thanks in Advance. Regards, Pavithra

2 2

Re: [SR-Users] Regd : SIP Configuration in Kamailio
by Henning Westerholt 14 Oct '20

14 Oct '20

Hello, please keep the list in CC. I can see Kamailio in your log: Oct 14 11:21:59 nickar /usr/sbin/kamailio[24567]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received Oct 14 11:21:59 nickar systemd[1]: Stopping Kamailio (OpenSER) - the Open Source SIP Server... Oct 14 11:21:59 nickar /usr/sbin/kamailio[24568]: INFO: <core> [main.c:842]: sig_usr(): signal 15 received About tracing/debugging, one good option is sngrep. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Santanu B <santanu.br(a)gmail.com> Sent: Wednesday, October 14, 2020 1:41 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Hi, My syslog file is enclosed for your reference. Thanks, Santanu On Wed, Oct 14, 2020 at 1:07 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, have a look to /var/log/messages or /var/log/syslog, depending on the distribution. Kamailio might be logged there. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of Santanu B Sent: Wednesday, October 14, 2020 7:09 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>> Subject: Re: [SR-Users] Regd : SIP Configuration in Kamailio Dear David, In /var/log folder I am unable to see any kamailio log. Only syslog file is available. Please help me to find the Kamailio log. Also please guide me to find out the trace also as I am very new to SIP Server. Thanks a lot. Regards, Santanu On Tue, Oct 13, 2020 at 4:05 PM David Villasmil <david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com>> wrote: Hello, You will need to provide more information. What happens? What errors do you see in kamailio’s log? A trace would also be nice. On Tue, 13 Oct 2020 at 06:24, Santanu B <santanu.br(a)gmail.com<mailto:santanu.br@gmail.com>> wrote: Hi, We are using Jitsi Meet for Video Conference Service. We want our SIP endpoint to join in the conference room in Jitsi. For that i am trying to configure Kamailio SIP Server in our environment. I have installed the Server and my endpoints are registered in my SIP server. Both endpoints are able to connect to each other. But I am unable to configure the SIP endpoints to join in the Meet Conference Room. Please guide me to configure. Thanks and regards, Santanu _______________________________________________ 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 -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 _______________________________________________ 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

1 0

SRTP
by Andy Kama 14 Oct '20

14 Oct '20

Hi All, I have setup a tls connection to a sip provider and it works perfectly But they are requesting i send srtp i can find any documentation how i do it with rtpengine etc? any guidance would be appreciated

3 3

pstn_route add_route(): route name <IP> not defined error
by sai Sudheer 14 Oct '20

14 Oct '20

Hii,I am trying to use pstn_route module for the first time, and i want to route the invite to specific pstn server based on its prefix.I have loaded the module & setup the DB, and it's giving error saying route name (from DB) is not defined after adding the a prefix into DB table., and there were no errors when DB table is empty.Not sure how to defined it, or i dont know what did i miss. Please help. #### CONFIG ##########loadmodule "prefix_route.so"modparam("prefix_route", "db_url", "mysql://xxxxx:xxxxx@localhost/kamailioDB")modparam("prefix_route", "db_table", "new_prefix_route") #PSTN Routing table contains if (!prefix_route("+44")) { xlog("L_INFO", "+44 prefix didnt match with prefix_route DB\n"); } ########## DB####MariaDB [kamailio]> select * from new_prefix_route;Empty set (0.00 sec) MariaDB [kamailio]> exit;########## Kamailio looks good untill here, no errors.. # Now i have added a prefix into sql DBINSERT INTO new_prefix_route VALUES ("+44", "10.10.1.1:5060", "route for this prefix");###########Error log after adding a prefix into new_prefix_route table & restart of kamailio 0(5802) CRITICAL: prefix_route [prefix_route.c:72]: add_route(): route name '10.10.1.1:5060' is not defined 0(5802) NOTICE: prefix_route [prefix_route.c:170]: pr_db_load(): Total prefix routes loaded: 1 0(5802) ERROR: prefix_route [prefix_route.c:174]: pr_db_load(): error flushing tree 0(5802) CRITICAL: prefix_route [prefix_route.c:212]: mod_init(): db load failed I have tried a "name" instead of IP (though i dont know how to convert that "name" into a IP:Port), But i am still getting same error using "name" instead of IP:port @ route field in DB. Please help.

2 2

how to get a value from SIP INVITE Header
by sai sudheer 14 Oct '20

14 Oct '20

Hii Please help me to get the a value from SIP INVITE header reached to kamailio like INVITE sip:+341930203454@sub.domain.com;myid=+34@sub.domain.com SIP/2.0 and i want to save the myid value +34 into a variable, without the domain name. $var(uri) = $sel(ruri); xavp_params_explode("$(var(uri){s.unbracket})", "uri"); xlog("L_INFO", "$var(uri) Received converted to $xavp(uri=>myid[0])\n"); I tried above and it prints +34(a)sub.domain.com But i want to just save +34 into a variable to further check the prefix based routing from the database. 1) Could you please help how to get it or If there is any alternate/single line approach to get this value? 2) $var & $xavp are process-local variables, and they cant be shared with other calls? Right 3) What is the best module to route calls based their prefix ? pstn_route or dynamic routing or any suggestions? Please help, thanks in advance.

3 7

'302 Moved temporarily' contact parsing error
by David VILLAUME 14 Oct '20

14 Oct '20

Hello, I see parsing errors on 302 contact header that seems compliant to RFC : Contact:<sip:172.16.0.21:5060>;q=0.5,<sip:172.16.0.22:5060>;q=0.25 <core> [core/parser/parse_addr_spec.c:479]: parse_to_param(): invalid character ',' in status 27: [;q=0.5] Is there an error in my contact or is it truly a parsing error ? Regards, David

3 6

Wants to know kamailio supports IMS with SRVCC Handover or not
by Anwesh Babu 14 Oct '20

14 Oct '20

Hello Team, I am working for a telecom company and want to use Kamailio for my VoLTE testing. I had earlier used these tools for my VoLTE testing using a Polaris CNE. But i have a requirement to perform SRVCC Handover for CS/CS+PS. will Kamailio support the SRVCC Handover? If supported, can it be possible to share the details ? -- Best Regards, Anwesh Babu T

1 0

routing calls based on prefix
by Andy Kama 13 Oct '20

13 Oct '20

Hi All, What is the best way to approach this? example if a call comes into kamailio with a specific number it example to:123456789 it must go to the dispatcher module but if a call comes with any other number it must route to a uac or can you use uac with dispatcher? basically i want to route inbound calls to a dispatcher and calls from that dispatcher out a uac

2 2

Caseless regex
by Marat Gareev 13 Oct '20

13 Oct '20

Hello, Is it possible write more readable regular expression if ($tU =~ "(?i)^test$") { ... } instead of "^[tT][eE][sS][tT]$"? On the first option, I get the error: 0(47250) ERROR: <core> [core/rvalue.c:3038]: fix_match_rve(): Bad regular expression Regards, Marat

3 5

about $sht(htable=>key) htable name
by Juha Heinanen 13 Oct '20

13 Oct '20

Is there some good reason why htable in $sht(htable=>key) cannot be a dynamic string containing pseudo variables? -- Juha

2 2

Re: [SR-Users] Local cache registration
by David VILLAUME 13 Oct '20

13 Oct '20

I read again the uac doc and didn’t find a way to perform this using the functions that seems to require credentials that I don’t have on the kamailio host. Is the uac module really designed to perform this operation is that way ? Regards, David El Sun, 11 Oct 2020 20:07:58 +0000 David VILLAUME <david.villaume at sewan.fr<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> escribió: > Hello, > > I try to perform a registration caching in order to have a short interval > registration on phone <> Kamailio (5 minutes) and a longest interval on the > leg Kamailio <> registrar(1h). > > I’m not so sure about the best way to proceed, does one of you have a working > sample or some hints about the way to have it working ? > > the uac module does that. You can handle upstream registrations in different intervals than the ones you receive.

4 9

Permissions address
by David Villasmil 13 Oct '20

13 Oct '20

Hello guys, Is there any way to reload the address table in the cfg? I always need to use "kamctl address reload"... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Dispatcher stops working
by Sergio Charrua 12 Oct '20

12 Oct '20

Hello, Using Kamailio 5.1.6 for load balancing & failover, I noticed that from time to time, and very rarely, the dispatcher module stops dispatching to the Asterisk servers. I have to execute a "kamcmd dispatcher reload" for it to (re)start working. This is my dispatcher.list : 1 sip:10.19.XXX.YYY:5060 0 1 duid=sipgw01;maxload=100 1 sip:10.19.XXX.YYY:5060 0 2 duid=sipgw02;maxload=100 1 sip:10.19.XXX.YYY:5060 0 3 duid=sipgw03;maxload=100 1 sip:10.19.XXX.YYY:5060 0 4 duid=sipgw04;maxload=100 This is my dispatch routes: # Dispatch requests route[DISPATCH] { if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } I suspect that the ds_mark_dst("ip") command in RTF_DISPACTH is the cause. Is there any think I can improve on this? Or any know bug with the dispatcher in Kamailio 5.1.6 version? Thanks in advance, *Sérgio Charrua*

4 4

Logging problem
by Jan Rozhon 12 Oct '20

12 Oct '20

Hi, sorry for quite stupid and newbie questions, but I am out of ideas. 1. On one of my Debian 10 servers I am unable to stop logging kamailio messages to /var/log/syslog. It is funny, because I have 3 more servers where it works but on this one, it doesnt and no matter what I try it still keeps sending logs to syslog (and to the file configured). My related configs: /etc/kamailio/kamailio.cfg log_stderror=no log_facility=LOG_LOCAL7 /etc/rsyslog.conf local6.* /var/log/kamailio/cdr.log local7.* /var/log/kamailio/kamailio.log /etc/rsyslog.d/kamailio.conf if $programname contains 'kamailio' then /var/log/kamailio/kamailio.log & stop /etc/systemd/system/multi-user.target.wants/kamailio.service SyslogIdentifier=kamailio 2. It is probably related to 1. I am unable to get the CDRs logged to anything else but /var/log/syslog. I tried database (nothing) and setting cdr_facility and again nothing. It just keeps sending everything to /var/log/syslog. Related configs: modparam("acc", "cdr_enable", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_facility", "LOG_LOCAL6") Example output in syslog file: Sep 30 09:21:41 sip2 /usr/sbin/kamailio[1151]: NOTICE: {2 21 INVITE ZPZI7Hqytw} acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1601450501;method=INVITE;from_tag=1GH5hGQxN;to_tag=QwcYFet;call_id=ZPZI7Hqytw;code=486;reason=Busy here;src_user=XXXXXXXXX; Is there something trivial I am missing? Thank you. Regards, Jan

4 7

Local cache registration
by David VILLAUME 11 Oct '20

11 Oct '20

Hello, I try to perform a registration caching in order to have a short interval registration on phone <> Kamailio (5 minutes) and a longest interval on the leg Kamailio <> registrar(1h). I’m not so sure about the best way to proceed, does one of you have a working sample or some hints about the way to have it working ? Thanks in advance , David

2 1

Kamailio Bridge
by conradcordeiro＠gmail.com 09 Oct '20

09 Oct '20

Hello everyone, Please, see this: https://github.com/fredposner/scripts/blob/master/kamailio/kamailio-tls_srt…. I am researching to understand how this TLS-bridge configuration works and have a main question about how passwords are handled in this scenario. If the PBX in the equation requires password authentication, when SIP clients connect to the Kamailio bridge with that username and password, does Kamailio pass through the username and password to the PBX? Many thanks, Conrad

2 2

jsonrpcs - safest way to get data
by Ilie Soltanici 09 Oct '20

09 Oct '20

Hello, I'm trying to periodically get all registered contacts from the Kamailio memory in the json format. JSONRPCS module have 3 different types of transport to get this data. I just wonder what will be the safest transport for the Kamailio to get all this data? Because all this data it's not so important, I want to gather it without affecting the main process or with the minimal effect on the Kamailio SIP Processing module. 1. The HTTP transport - very useful, working async, but because it depends on the xhttp module - which works synchronously I'm afraid that for big data with contacts this type of transport can affect the kamailio performance. 2. FIFO - this transport also is very fast, working locally - but it's synchronous, I'm not sure how it will affect Kamailio SIP Processing when the local database will have few millions of users registered. 3. DGRAM - also seems to be very fast, working locally or remotely through the UDP protocol, but again it's synchronous and I'm not sure that this transport is returning any output for the request, I tried - but didn't get back anything, while for changing something it worked fine. Maybe it was a misconfiguration in my config file. What would be your recommendation for this case? Thank you.

3 3

modify Proxy-Authenticate header parameter
by David Villasmil 09 Oct '20

09 Oct '20

Hello guys, I'm receiving a Proxy-Authenticate header with a special parameter, call if "myparam", something like: Proxy-Authenticate: Digest realm="sip.mydomain.com", nonce="...", algorithm=MD5, qop="auth", opaque="XXXX/YYYY" I can get the parameter like so: $var(myparam) = @proxy_authorization["sip.mydomain.com"].myparam; And this is great! But i need to replace part of the parameter "myparam"m and I can't figure out how to do it... Can anyone help me with this? Thanks!, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Caller-ID and Dbaliases
by HS 08 Oct '20

08 Oct '20

Hi all. We want to manage users via dbaliases so that one subscriber can have multiple aliases. We have an internal username in the subscriber table. We authenticate with the internal username, the call goes through with the internal username. Is there anyway to specify: 1. Caller-ID to be the actual email_address (added to subscriber table), 2. Authenticate with dbaliases? Any other thoughts please? Appreciate the help. Best wishes.

1 0

Result limitation on sqlops
by Duarte Rocha 07 Oct '20

07 Oct '20

Hey, I just upgrade to Kamailio 5.4 and i see that now sqlops has a limitation of 32 results containers which is not enough for my proxy. Is it possible that this limitation can be configured via module parameters? In the meanwhile i've changed the value in sql_api.c but i would like to have my source code as close as the original as possible. Best Regards, Duarte Rocha

3 3

SIP client - Linphone
by Pavithra Mohanraja 06 Oct '20

06 Oct '20

Hi, I am a newbie to sipclient Linphone. I know this issue is not related with kamailio.Yet If anybody have tried it before , kindly provide me with some links or procedures to be followed for call registration and connectivity with kamailio since in my case, request is going but response is not coming back resulting in timeout issue. This happens only in linphone.If i do the samething in PJSIP/zoiper, call registration and call establishment happens perfectly fine. Kindly help. Thanks, Pavithra

4 10

kamailio SBC - session timers
by marek 06 Oct '20

06 Oct '20

hi, i have kamailio+rtpengine acting as a SBC (private/public net interfaces) kamailio 5.4.x/rtpengine 9.x/centos 7 using TOPOH module modparam("topoh", "mask_ip", "public_IP") i'm rewriting IPs in INVITE like this $var(ruri) = "sip:" + $(ru{re.subst,/private_IP/public_IP/g}); $ru = $(ru{re.subst,/private_IP/public_IP/g}); $tu = $(tu{re.subst,/private_IP/public_IP/g}); $fu = $(fu{re.subst,/remote_private_IP/public_IP/g}); the same for route[WITHINDIALOG] and INVITE i have problem with session timers using re-INVITE method in the first INVITE SDP is correct c=IN IP4 public_IP but in the reinvite is c=IN IP4 private_IP and remote SBC drop with 422 because SDP is changed and SDP version is not. any tips / best practice? thanks Marek

2 2

I want separate subscribers in groups
by ahmed moghazy 06 Oct '20

06 Oct '20

Hello sir, I got the the log from kamailio as u explained thank you , now i want to put subscribers in groups so subscribers in group (a)could not call subscribers in group(b) , and i want to make subscriber could call any one but no one could call him , thank you.

2 1

Drop calls with response 200 without previous 180/183 reponses
by Valter Nogueira 05 Oct '20

05 Oct '20

I would like to use Kamailio to drop every call that is answered with no previous 180 / 183 responses or with 180 / 183 responses with no media attached (no SDP?). Is it possible? How could I accomplish this? Thanks, Valter

4 6

SIP message generator for Kamailio
by luke devon 05 Oct '20

05 Oct '20

Hi Team, May I check with you, Is there a SIP message generator available for the Kamailio? I want to test my setup using some test SIP messages. Many ThanksLuke.

5 6

xcap_client: storage of document in local DB
by Valentin Christoph 05 Oct '20

05 Oct '20

Hi all, Does anybody know, when the xcap_client deletes a registered document from the database? Is it possible to deregister a document? Are you sure, the documented get_elem API function is actually implemented? All the best Christoph

3 5

dlg.list_match - Invalid matching key parameter
by Ilie Soltanici 03 Oct '20

03 Oct '20

Hello, The dlg.list_match rpc function from the dialog module is not able to find dialogs matching by the callid key: kamcmd dlg.list_match callid eq "01234567890(a)192.168.0.1" getting back: error: 500 - Invalid matching key parameter I think it's because of this check: 2945:dialog.c: } else if(mkey.len==5 && strncmp(mkey.s, "callid", mkey.len )==0) { the mkey.len - should be 6 in this case, I recompiled locally the dialog module and seems to be working ok. Thank you.

3 2

Kamailio 5.2.8 released
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, Kamailio SIP Server v5.2.8 stable release is out. This is a maintenance release of the old stable branch (5.2), that includes fixes since the release of v5.2.7. There are no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.8. For more details about version 5.2.8 (including links and guidelines to download the tar file or from GIT repository), visit: https://www.kamailio.org/w/2020/10/kamailio-5-2-8-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/>

1 0

403 Not relaying (DNS)
by Justin Bookshar 02 Oct '20

02 Oct '20

Hello, I'm trying to use Kamailio as a proxy and registrar component in our network and I would like to be able to do several things. I have an MCU capable of making SIP calls and using Kamailio as a proxy but it is not able to register with Kamailio. I would like to have Kamailio resolve addresses using its DNS component. I have attempted to do this, but I am getting 403 Not relaying responses when attempting to make calls through Kamailio that require the destination URI to be resolved. I have enabled debug and I don't see anything that would cause this. I have verified our /etc/resolv.conf file. When I use the primary DNS server in the list on an endpoint, I can get the address to resolve and the call completes correctly. I know I have working equipment on each end and the DNS server and URI are correct but going through Kamailio fails for both the MCU and endpoint when using it as a proxy. I have also tried to disable the internal resolver by adding the use_dns_cache=no option to the kamailio.cfg file. This produced the same result but as I mentioned I have independently verified the DNS server so I know the issue is isolated to Kamailio. Is it possible for Kamailio to do this or am I wrong in believing it can? The second thing I would like to do is to route calls to the MCU based on a prefix. I think I roughly have this working by adding a custom routing block to the request_route operation. I would also like to be able to make calls to local equipment that is not registered with the registrar and I am not able to do this either. I get the same 403 Not Relaying error when I attempt to call a local endpoint by IP address through the proxy that is also not registered with the registrar. Any info is greatly appreciated. version: kamailio 5.2.5 (x86_64/linux) 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, 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: unknown compiled on 09:56:06 Oct 17 2019 with gcc 8.3.0 Thanks, Justin

1 0

MS Teams Serivce unavailable
by Andy Kama 02 Oct '20

02 Oct '20

Hi All, I have done all the changes to get teams working according to https://skalatan.de/en/blog/kamailio-sbc-teams#:~:text=Extend%20the%20MS%20… . but i find when i send a call to teams they send a response back with service unavailable. Can anyone guide me what I am doing wrong? I have added the record route and confirm it is being sent My setup is freeswitch -- kamailio --- teams

3 13

Releasing Kamailio 5.2.8
by Henning Westerholt 02 Oct '20

02 Oct '20

Hello, I am planning to release next week (likely on Friday 2nd October) the next minor version out of branch 5.2, version 5.2.8. 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). Unless major regressions are discovered after the release, this will be the last release of the 5.2 stable branch. The current stable branches 5.3 and 5.4 will continue to be maintained. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio fails to start when tls module parameter 'config' contains a folder name
by Leonid Fainshtein 02 Oct '20

02 Oct '20

Hi, I am using Kamailio 5.4.1 installed from the packages on Ubuntu 18.04. When I define 'modparam("tls", "config", "<folder>")', Kamailio fails to start: ERROR: tls [tls_config.c:418]: tls_load_config(): cannot make tmp file configrcCczj ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module tls (/usr/lib/x86_64-linux-gnu/kamailio/modules/tls.so) I checked the tls_load_config() function and found that when a folder is defined in the 'config' parameter, the function tries to open a temporary file in the working directory. I believe that the working directory at that point is the folder where Kamailio was started: /usr/sbin. I succeeded to launch Kamailio in two ways: Method 1: copy 'kamailio' to /tmp and modify kamalio.service accordingly. This experiment result made me think that the working folder is the folder where kamailio is started. Method 2: add Kamailio command line parameter "-w /tmp" (I also tried to define the Systemd 'WorkingDirectory' parameter but it didn't have any effect.) I think that the ultimate solution for this problem is to modify the tls_load_config() function in such a way that it will open the temporary file in a folder that is intended for the temporary files. For example, the /tmp folder in Linux and *BSD systems. I guess that /tmp is a good place in MacOS as well. Best regards, Leonid Fainshtein

2 3

TOPOS issue
by David Villasmil 01 Oct '20

01 Oct '20

Hello guys, I have this scenario where I have an ongoing transaction. I receive a 180 (non 100rel) with some to_tag, which i forward back to the caller. Then I receive an 183 (100rel) which is forwarded back to the caller, who in turn PRACKs it. It is then that Kamailio (apparently) can't match that transaction internally and replies to the caller (the one PRACKing the 183 (100rel)), with 493 FORBIDDEN. I'm attaching a trace flow. (Kamailio is 10.255.0.1) you need to open the email really wide to see it properly. Anyone ever encountered this? Is this valid and should work? |Time | 10.15.169.4 (FS) 10.255.0.1 (KAM) | (KAM) 1.1.1.10 1.2.3.13 | | | | | | | | | | | | | |0.000000 | INVITE SDP (g722 g71 | | | | |(6000) ------------------> (5060) | | | |0.000941 | 100 Telnyx trying | | | | |(6000) <------------------ (5060) | | | |0.046558 | | | INVITE SDP (g722 g71 | | | | |(33033) ------------------> (5060) | |0.102078 | | | 100 Trying, boss | | | | |(33033) <------------------ (5060) | |0.386867 | | | 180 Ring a ling a li | | | | |(33033) <------------------ (5060) | |0.387578 | 180 Ring a ling a li | | | | |(6000) <------------------ (5060) | | | |3.413368 | | | 181 Prending to forw | | | | |(33033) <------------------ (5060) | |3.414373 | 181 Prending to forw | | | | |(6000) <------------------ (5060) | | | |6.441704 | | | 183 Seeing what I ca | | | | |(33033) <------------------ (5060) | |6.442896 | 183 Seeing what I ca | | | | |(6000) <------------------ (5060) | | | |6.890173 | PRACK | | | | | |(6000) ------------------> (5060) | | | |6.895497 | 403 Forbidden R1 | | | | |(6000) <------------------ (5060) | | | |9.473035 | | | 486 Actually, I'm qu | | | | |(33033) <------------------ (5060) | |9.473561 | | | ACK | | | | | |(33033) ------------------> (5060) | |9.473902 | 486 Actually, I'm qu | | | | |(6000) <------------------ (5060) | | | |9.921200 | ACK | | | | | |(6000) ------------------> (5060) | | | Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 2

Record Route
by Andy Kama 01 Oct '20

01 Oct '20

Hi Guys I am trying to add a custom record route so i have added record_route_preset("mydomain:5061;transport=tls", "myip:5060"); yet kamailio is still not adding it in the sip message what am i doing wrong?

2 3

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sr-users October 2020