- sr-users - kamailio.org

Error when using http_async_client $http_query_id var
by jenus＠cyberchaos.nl 24 Jul '18

24 Jul '18

Hello, I'm trying to build a async http call to a external server and was trying to use the $http_query_id var to identity the http call in the response route. But when i try the following: $http_req(suspend) = 0; $http_req(hdr) = "Content-Type: application/json"; $http_req(body) = "{'r_uri':'" + $rU + "', 'f_uri':'" + $fU + "'}"; http_async_query("http://10.10.10.1/", "ACC_HTTP_REPLY"); xlog("L_WARN", "ACC Query id: $http_query_id"); route[ACC_HTTP_REPLY] { xlog("L_WARN", "route[ACC_HTTP_REPLY]: received reply for query $http_query_id\n"); } I get errors in the kamailio log file : Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id" Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [d/100] in [$http_query_id] at [13 (0)] Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for query $http_query_id#012] Jul 23 11:34:14 sip03 /usr/sbin/kamailio[24645]: ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:5308 Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "http_query_id" Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$(http_query_id)] at [15 (1)] Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[route[ACC_HTTP_REPLY]: received reply for query $(http_query_id)#012] Jul 23 11:36:07 sip03 /usr/sbin/kamailio[24906]: ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg://etc/kamailio/kamailio.cfg:5308 Should i escape the var $http_query_id in some way? I'm running kamailio 5.1.3. Thanks, Jan

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Kamailio Calls Concurrency
by Abtin Mir 24 Jul '18

24 Jul '18

Hello Support, As a proxy server , How many concurrent calls can be processed by a Kamailio which is installed on a VM with 4 cores and 16GB of RAM? Your answer is much appreciated. -- *Regards,* *Abtin Mir*

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Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Thank you Konstantin, What I can’t understand is why media packets are sent correctly to the public IP , while early media (RBT) are sent on private. Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 1:38 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[4]: [SR-Users] NATHELPER issue Original INVITE contains private IP in c= parameter, so you need to use STUN or TURN or ICE, something which is supported in your system or in your softphone. See my comments in the previous mail. Best regrads Понедельник, 23 июля 2018, 12:25 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X <mailto:506251539613159370@X.X.X.X> >;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 where: X.X.X.X is Kamailio IP Y.Y.Y.Y is origination public IP Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <file://e.mail.ru/compose/%3fmailto=mailto%253apiligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <mailto:070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <mailto:CbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков. С уважением, Константин Поляков.

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messages dumped in wrong order sometime
by Philip Kucheev 23 Jul '18

23 Jul '18

Hello there! Using kamailio 5.0.1 and msilo module for storing offline messages I noticed that sometimes, especially when a big number of messages dumping for one user, some of them can arrive to receiver in wrong order. Lets say we have 20 messages and the order of receiving could be next 1..18..17..19..20. That could be a reason of that ? Thanks, Philip

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Forwarding to backup server
by Pali Rohár 23 Jul '18

23 Jul '18

Hello, can somebody help me how to configure Kamailio to route/forward requests to different (backup) server when Kamailio itself is not able to deliver them? I have working SIP gateway (on different host/port) which deliver received SIP messages to XMPP network and I want to achieve: Kamailio should try to send SIP messages directly to foreign SIP domains and when it fails (e.g. user not available or foreign SIP is not running), then forward message to that my SIP <--> XMPP gateway which will try to deliver it via XMPP. I guess that Kamailio should be able to configure for such custom routing/forwarding but I have not figured yet how to do it. I'm using Kamailio 4.4 with slightly modified default kamailio.cfg config. -- Pali Rohár pali.rohar(a)gmail.com

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Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 where: X.X.X.X is Kamailio IP Y.Y.Y.Y is origination public IP Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков.

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Re: [SR-Users] NATHELPER issue
by Ali Taher 23 Jul '18

23 Jul '18

Hi Konstantin, INVITE: INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> X-AUTH-IP: Y.Y.Y.Y Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0 Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Max-Forwards: 69 Contact: <sip:967123456@192.168.26.3:31556;alias=Y.Y.Y.Y~31556~1> To: "9613159370"<sip:9613159370@X.X.X.X:5065> From: "+967123456"<sip:967123456@X.X.X.X:5065>;tag=5269152d Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1003s stamp 31159 Content-Length: 434 v=0 o=- 5 2 IN IP4 192.168.26.3 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.26.3 t=0 0 m=audio 8602 RTP/AVP 0 8 101 a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602 a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602 a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602 a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF 200 OK 200 OK Via: SIP/2.0/UDP 192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556 Record-Route: <sip:X.X.X.X:5065;lr;ftag=5269152d> From: "+967123456" <sip:967123456@X.X.X.X:5065>;tag=5269152d To: "9613159370" <sip:9613159370@X.X.X.X:5065>;tag=agtHmrp3Dccrg Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM. CSeq: 1 INVITE Contact: <sip:9613159370@X.X.X.X:5060;transport=udp> User-Agent: FIKAR Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 200 Remote-Party-ID: "Outbound Call" <sip:506251539613159370@X.X.X.X>;party=calling;privacy=off;screen=no v=0 o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X s=FIKAR c=IN IP4 X.X.X.X t=0 0 m=audio 24496 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Regards, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru> Sent: Monday, July 23, 2018 12:08 PM To: Ali Taher <ataher(a)vanrise.com> Subject: Re[2]: [SR-Users] NATHELPER issue Hello Ali, Could you show your latest INVITE and OK? Check you SDP bodies in those messages, if you still have there private IPs then you need to solve usual NAT traversal task in your media software (not proxy). Kamailio doesn't process media as you know. Of course you can try fix_nated_sdp function from nathelper module but it will not work for symmetric NAT because you fix SDP with public IP toward proxy but not toward UA. Your media software or hardware should support STUN or ICE protocols to work correctly via NAT. Best regards. Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> >: Thank you for your reply. ACK is now sent correctly to public IP , yet I have another issue with RBT where it’s still sent to private IP .Yet the call media is sent correctly to the public IP. There is something strange , as I hear two short different ringing tones then complete silence till the called party answer the call and RTP flow normally. How can I handle this case? Thanks, Ali From: Konstantin Polyakov <piligrim_pk(a)mail.ru <mailto:piligrim_pk@mail.ru> > Sent: Friday, July 20, 2018 3:01 PM To: sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali Taher <ataher(a)vanrise.com <mailto:ataher@vanrise.com> > Subject: Re: [SR-Users] NATHELPER issue Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin Message: 23 Date: Fri, 20 Jul 2018 11:13:42 +0300 From: "Ali Taher" <ataher(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253aataher@vanrise.com> > To: <sr-users(a)lists.sip-router.org <file://e.mail.ru/compose/%3fmailto=mailto%253asr%252dusers@lists.sip%252drouter.org> > Subject: [SR-Users] NATHELPER issue Message-ID: <070901d42001$939be2b0$bad3a810$(a)vanrise.com <file://e.mail.ru/compose/%3fmailto=mailto%253a070901d42001$939be2b0$bad3a810$@vanrise.com> > Content-Type: text/plain; charset="utf-8" Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX <file://e.mail.ru/compose/%3fmailto=mailto%253aCbeX8453909200habfGhEfElPce@BC00.XXXXXXXXXXXXXX> CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher "Ali Taher" <ataher(a)vanrise.com <https://e.mail.ru/compose?To=ataher@vanrise.com> > С уважением, Константин Поляков.

1 0

KEMI: accessing "keyword" variables from core
by Enrico Bandiera 23 Jul '18

23 Jul '18

Hi, anyone how to access keyword variables like dst_port or src_port from kemi? (specifically python in our case). Thanks, Enrico.

2 3

Re: [SR-Users] NATHELPER issue
by Konstantin Polyakov 23 Jul '18

23 Jul '18

Hello Ali, ACK is sent by UAC to the Contact which is received in OK from UAS. So from my point of view you need to fix that Contact in OK on your proxy. My NATDETECT looks like: if (nat_uac_test("19")) { fix_nated_contact(); } I call it for requests and responses. Best regards. Konstantin >Message: 23 >Date: Fri, 20 Jul 2018 11:13:42 +0300 >From: "Ali Taher" < ataher(a)vanrise.com > >To: < sr-users(a)lists.sip-router.org > >Subject: [SR-Users] NATHELPER issue >Message-ID: < 070901d42001$939be2b0$bad3a810$(a)vanrise.com > >Content-Type: text/plain; charset="utf-8" > >Hello, > > > >I'm using Kamailio 4.2 as proxy with nathelper enabled. > > > >Yet , the ACK packet sent from the proxy to the origination's private IP. > > > >The ACK is sent as reply on the following 200 OK sent from the origination : > > > > >8m2EJN41BN/6WSIP/2.0 200 OK > >From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze > >To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 > >Via: SIP/2.0/UDP >X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= >X.X.X.X,SIP/2.0/UDP >X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN > >Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> > >Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX > >CSeq: 125698370 INVITE > >Accept: application/sdp > >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE > >P-Charging-Vector: >icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX >;orig-ioi=MXXXXXXXXXXXXXX > >Content-Type: application/sdp > >Contact: <sip:172.16.45.65:5060;transport=UDP> > >Content-Length: 268 > > > >v=0 > >o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX > >s=- > >c=IN IP4 172.16.45.144 > >t=0 0 > >a=sendrecv > >m=audio 47588 RTP/AVP 18 96 > >c=IN IP4 172.16.45.144 > >a=rtpmap:18 G729/8000 > >a=fmtp:18 annexb=yes > >a=rtpmap:96 telephone-event/8000 > >a=fmtp:96 0-15 > >a=maxptime:20 > > > >Following is the header of the sent ACK packet: > > > >Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 > >Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> > >Via: SIP/2.0/UDP >X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 > >Via: SIP/2.0/UDP >X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc > >Max-Forwards: 69 > >From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze > >To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 > >Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX > >CSeq: 125698370 ACK > >Content-Length: 0 > > > >Where X.X.X.X is Kamailio server public IP. > > > >Following is part of my config file : > > > >route { > >route(NATDETECT); > >record_route(); > > > > if(!mf_process_maxfwd_header("10")) { > > sl_send_reply("483", "Too Many Hops"); > > exit; > > } > > > > # Maybe some sanity_check() here. > > > > if(has_totag()) { > > > > if(loose_route()) { > > route(DLGURI); > > if(!t_relay()) > > sl_reply_error(); > > > > exit; > > } else { > > if(is_method("ACK")) { > > route(DLGURI); > > if(t_check_trans()) { > > t_relay(); > > } > > } else > > sl_send_reply("403", "Forbidden"); > > } > > exit; > > } > > > >..... > >} > > > > > >route[NATDETECT] { > >#!ifdef WITH_NAT > > force_rport(); > > if (nat_uac_test("19")) { > > if (is_method("REGISTER")) { > > fix_nated_register(); > > } else { > > add_contact_alias(); > > } > > setflag(FLT_NATS); > > } > >#!endif > > return; > >} > > > >route[DLGURI] { > >#!ifdef WITH_NAT > > if(!isdsturiset()) { > > handle_ruri_alias(); > > } > >#!endif > > return; > >} > > > >Can you please check why the ACK is still sent on private IP ? > > > >Thanks > >Ali Taher > > "Ali Taher" < ataher(a)vanrise.com >

2 1

Selectively enable / disable cdrs using acc module
by M S 23 Jul '18

23 Jul '18

Hi, I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is there a way to enable / disable cdrs on per call basis in kamailio script? For example i want to allow cdrs only for offnet calls and disable them for onnet calls? May be a dialog flag configurable in module parameters to signal acc module whether or not to generate cdr for current dialog at the end of call?

2 5

Options for SIP traffic duplicating to third party tcp port
by Максим Баранов 23 Jul '18

23 Jul '18

Hello, what are the options for duplicating sip traffic to arbitrary tcp port in Kamailio?

3 5

forbidden domain not served
by eyas barhouk 22 Jul '18

22 Jul '18

Dears i'm using kamilio 5.2.0 as IMS with Zoiper client and RTP Engine when i'm trying to terminate a video call ,BYE message does not arrived to destination with error from S-cscf side "forbidden domain not served " has any one had the same issue before ? and is there any ideas about how to solve it ? ? thanks in advance

2 3

dialog module with dmq
by E. Schmidbauer 22 Jul '18

22 Jul '18

Is anyone using dialog module with DMQ to share dialogs across multiple kamailio instances? i've been testing it and it seems unreliable. sometimes the dialog data is sent from nodeA to nodeB, sometimes it is not. im wondering if im missing some settings or it's a bug. I tested latest master branch and version 5.1.4 Thanks! Emmanuel

3 2

$var(name) - Private memory variables
by Sergey Safarov 21 Jul '18

21 Jul '18

Hello I have faced when case when private var defined as $var(json_request) is share between different SIP messages. In my case first time this var is used from registration processing route. Then var with same name is used PUBLISH messge processing. And i be able to this value defined from registration processing. I expecting that vars is not shared between threads and message handling routes for var defined as $var(name). Is that not true? How i can prevent race condition between different message processing routes when two thread simultaneously change same variable? Sergey

3 5

uri==myself
by Karthik Srinivasan 20 Jul '18

20 Jul '18

Hello, Can someone explain to me why statement A returns True and statement B returns False. Statement A: is_myself("$ru") Statement B: uri == myself Isn't uri and $ru referencing the same data? With my current config i am expecting the is_myself to return True; i was expecting the same for Statement B but unfortunately it is returning False. Thanks, Karthik

2 11

how to configure ims in kamailio
by vinod mn 20 Jul '18

20 Jul '18

Hi what are the requirements to configure the ims to work with kamailio. -- Thanks and regards Vinod.M.N

3 2

NATHELPER issue
by Ali Taher 20 Jul '18

20 Jul '18

Hello, I'm using Kamailio 4.2 as proxy with nathelper enabled. Yet , the ACK packet sent from the proxy to the origination's private IP. The ACK is sent as reply on the following 200 OK sent from the origination : 8m2EJN41BN/6WSIP/2.0 200 OK From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received= X.X.X.X,SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 INVITE Accept: application/sdp Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE P-Charging-Vector: icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XXXXXXXXXXXXXX.XX ;orig-ioi=MXXXXXXXXXXXXXX Content-Type: application/sdp Contact: <sip:172.16.45.65:5060;transport=UDP> Content-Length: 268 v=0 o=- 5838243 5838244 IN IP4 BC00.XXXXXXXXXXXXXX s=- c=IN IP4 172.16.45.144 t=0 0 a=sendrecv m=audio 47588 RTP/AVP 18 96 c=IN IP4 172.16.45.144 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 a=maxptime:20 Following is the header of the sent ACK packet: Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0 Record-Route: <sip:X.X.X.X:5065;lr;ftag=XQBQNjvjgp4Ze> Via: SIP/2.0/UDP X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0 Via: SIP/2.0/UDP X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc Max-Forwards: 69 From: <sip:+4444331234567@X.X.X.X;user=phone>;tag=XQBQNjvjgp4Ze To: <sip:+905362695933@172.16.45.65;user=phone>;tag=12033368836000 Call-ID: CbeX8453909200habfGhEfElPce(a)BC00.XXXXXXXXXXXXXX CSeq: 125698370 ACK Content-Length: 0 Where X.X.X.X is Kamailio server public IP. Following is part of my config file : route { route(NATDETECT); record_route(); if(!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; } # Maybe some sanity_check() here. if(has_totag()) { if(loose_route()) { route(DLGURI); if(!t_relay()) sl_reply_error(); exit; } else { if(is_method("ACK")) { route(DLGURI); if(t_check_trans()) { t_relay(); } } else sl_send_reply("403", "Forbidden"); } exit; } ..... } route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { add_contact_alias(); } setflag(FLT_NATS); } #!endif return; } route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } Can you please check why the ACK is still sent on private IP ? Thanks Ali Taher

1 0

The problem of get_aor_hash function
by tyd 20 Jul '18

20 Jul '18

Dear all, I'm trying Kamailio 5.1.4 and IMS module. When registering to Kamailio IMS using the same ip and port for 30 users (different contact user part), the P-CSCF get the same hash number and aor value. The function get_aor_hash(_d, &_ci->via_host, _ci->via_port, _ci->via_prot) in ims_usrloc_pcscf pcontact.c seems the problem. Because hashing with host, port, prot will get the same hash value. Anyone can help this ?

3 3

dialog module : questions about dialog states
by Abdoul Osséni 19 Jul '18

19 Jul '18

Hello list, When the dialog state for a call is < 4, is it possible to use dlg_bye() function to end the call? Best regards Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

2 5

Locking htable entries
by Alex Balashov 19 Jul '18

19 Jul '18

Hi, Sorry if this question has been answered before, but I'm rather puzzled as to the intent behind sht_lock() / sht_unlock(). Presumably, "thread"-safe / shm mutex locking is already implemented in htable query and mutation operations, otherwise htable would be unusable in a route script context. So what is the purpose of this higher-level lock? Is it to create a way to block config script execution in multiple processes as to serialise operations? Why would one want to do this under any ordinary circumstances? Perhaps it is a failure of my imagination... Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 4

executing python routine on demand
by Walter Martín Villalba 18 Jul '18

18 Jul '18

Hello, TL;DR How can I execute a routine in my Kamailio python module (kamailio_python.py) on demand? For example, using kamcmd from the command line. I know you can execute *app_python.reload *but I don't need to reload the whole python module, and furthermore, docs for this call say it's unstable and can even crash your kamailio instance. So my goal is to execute one single python routine on demand. https://www.kamailio.org/docs/modules/5.1.x/modules/app_python.html#app_pyt… Full explanation: I'm currently implementing OIDC token authentication for Kamailio. I have a python routine that extracts the token from a custom SIP header, and validates the token as per the OIDC spec (I'm using pyjwt plus other custom checks for this). In order to do offline validation of the token, you need the Identity Provider (IdP) Server's certificate, which I get in advance and cache in a local file on my Kamailio server. I also have an external python script which I can run to get the latest certificate from the IdP server and update it locally. The part I'm missing is how to tell Kamailio to reload the value of the certificate from the local cache, WITHOUT stopping kamailio or interrupting any sessions. This is needed in case the certificate is updated (e.g. due to expiration or compromise) on the IdP server; if you don't update it so that Kamailio can use the latest one, all token verification will fail (tokens signed by IdP server using latest private key corresponding to latest certificate, but Kamailio trying to validate the signature using the public key extracted from previous certificate -- this results in a signature error, as Kamailio needs to be using the public key corresponding to the private one that was used to sign the token). By being able to execute one single python routine on demand (e.g. using kamcmd), I can tell Kamailio to reload the certificate value from the local cache, which I just updated by running my other (external and unrelated to Kamailio software) python script. Hope this makes sense, and hopefully I can get some tips on how to achieve this. Thanks, Martín.

1 0

in-dialog requests
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using Kamailio as a proxy behind freeswitch to alter some packets (183 packets precisely). Here is my scenario: (---Customer---) --> ((port 5065)---Kamailio---) --> ((port5060)---Freeswitch---) --> (---Supplier---) Here is my request route : route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(X.X.X.X,5060); } Where X.X.X.X is the Freeswitch IP This script is working normally as long as the customer is sending the requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or freeswitch initiated the request , above script causes loop where the packet is bounced between kamailio and freeswitch. As I know , loose_route is used to handle such requests , but I'm not sure how to forward these packets to the customer after receiving them from freeswitch. I'm not sure also if what's I'm doing is statefull or stateless. Any help is appreciated. Thanks, Ali Taher

2 4

Kamailio dialog presence (BLF)
by Michal Škuta 18 Jul '18

18 Jul '18

Hello, I am trying make presence module work with Kamailio. Right now I am testing these modules: presence, presence_xml, presence_dialoginfo, pua and pua_dialoginfo. I had some success, the BLF is working only when the monitored extension is calling. When the monitored extension is calling, then the BLF changes its color (red). After end of the call the BLF changes color to green. My problem is when a User comes on-line then it sends SUBSCRIBE with *Event: dialog* and don't get notified of its subscribers state and BLF is not changed. I found similar problem in this mail list : https://lists.kamailio.org/pipermail/sr-users/2017-March/096314.html But is not working for me. Second question is about BLF and offline state. I can not found anything in RFC, there is no offline state, no way to turn off the BLF lamp from green to OFF. I found only this pdf from grandstream: http://www.grandstream.com/sites/default/files/Resources/GXP21x0_Eventlist_… mentioning offline state on page 8. Does anyone have experience with it?

2 2

Using the SIP related functions in a lib
by Bobby Mah 18 Jul '18

18 Jul '18

Hello all, Is it worth my time to try and create a SIP lib out of the SIP functionality available in Kamailio? Thanks for all your input, Bobby

2 1

SIP re-invite Packets
by Ali Taher 18 Jul '18

18 Jul '18

Hello, I'm using below routing logic to change the 183 reply to 180 and send it back to the origination. Noting that Kamailio is listening on 5065. route{ if (is_method("INVITE")) { xlog("L_INFO","INSIDE request route $si \n"); insert_hf("X-AUTH-IP: $si\r\n"); } route(FWD); } reply_route { #xlog("L_INFO","INSDE reply \n"); if (status == "183") { change_reply_status(180, "Ringing"); exit; } } route[FWD] { #xlog("L_INFO","INSIDE FWD \n"); forward(x.x.x.x,5060); } Below is the scenario , where the customer is sending packets to Kamailio , and Kamailio is forwarding them the supplier (with IP x.x.x.x) , then the supplier reply back to kamailio and kamailio relay the reply to the customer ----Customer---- --> ----Kamailio ---> ---Supplier--- I have a case where the supplier sends in some cases an in-dialog invite (Re-Invite) , which is considered I think as SIP-Request and thus as per above routing logic , kamailio is sending it back to the supplier , yet I need to send it to the customer. How can I solve this issue? Thanks Ali Taher

3 6

Error on Kamailio 4.3.7 and dialog
by Laura 18 Jul '18

18 Jul '18

Hi guys, on a test server with kamailio 4.3.7 statefull configuration with dialog I see a lot of messages like this.. Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 /usr/sbin/kamailio[31288]: ERROR: dialog [dlg_req_within.c:340]: send_bye(): terminating non-confirmed dialogs not supported Jul 16 15:20:26 vm-nextaitz-01 media-dispatcher[31555]: error: failed to end dialog: 500 Operation failed I'm not find the issue and the way for fix.. Any idea ? Regards Laura

3 8

ACK Cseq incrases after 404 on uac_auth()
by Yuriy Gorlichenko 17 Jul '18

17 Jul '18

Hi. I found some additional scenario when kamailio works incorrect with ACK Cseq after succesfull auth with uac_auth but unsuccesfull call: 1.Caller via kamailio as proxy (Cseq 1) 2.Kamailio resends call to provider with Auth method (Cseq 1) 3. Provider answers 401/407 (Cseq 1) 4. Kamailio inrases Cseq and sends invite with credentians (dialog track_cseq_updates = 1, uac_auth()) (Cseq 2) 5. Provider says - 404. not found (Cseq 2) 6. Kamailio resends INVITE to another server (lets say VM). It contans not tracked yet Cseq (Cseq 1) 7. VM server says 200 (Cseq 1) 8. Kamailio resends 200 to the caller (Сseq 1) 9. Caller says ACK (Cseq 1) 10. Kamailio incrases CSeq of ACK and resends t VM server (Сseq 2) So in this case kamialio tries to track Sceq because thining that this ACK belongs to the dialog that was made to the auth server So as I see here is a misunderstanding on the dialog module Any suggestions how to avoid it or exclude resent INVITE from the dialog module tracking?

2 4

RTPEngine serial forking with early media
by Alex Balashov 16 Jul '18

16 Jul '18

Hi, I've got a scenario where call branch #1 goes to an in-call announcement server; the message is played out via unidirectional early media (183+SDP), then that branch is disposed of with a 503. A subsequent call then goes out to the real SIP provider that will ultimately answer the call. There's some trouble handling this with RTPEngine: Two subsequent calls to rtpengine_offer() in a normal request route context will result in two SDP message lumps concatenated in the second INVITE (to the provider). This can be fixed by putting the rtpengine_offer() invocations into branch routes. However, the next problem is that then the identical offer goes to the second callee with no regard for RTPEngine's internal state, so RTPEngine ends up transmitting from a different port to the one that is declared in the SDP offer, requiring NAT latching from the provider to achieve proper symmetry in the return stream. This can be fixed with: rtpengine_offer("replace-origin replace-session-connection ICE=remove via-branch=1 internal external"); And in the reply handling: rtpengine_answer("replace-origin replace-session-connection ICE=remove via-branch=2"); An alternative to using via-branch[1] is to simply delete the call before initiating branch #2: rtpengine_delete("delete-delay=0"); rtpengine_offer("replace-origin replace-session-connection ICE=remove internal external"); However, both of these lead to the problem that caller receives two distinct SDP answers on the respective branches. I am not sure offhand whether this is a protocol semantics violation. On the one hand, RFC 3261 § 13.2.1 ("Creating the Initial INVITE") says: If the initial offer is in an INVITE, the answer MUST be in a reliable non-failure message from UAS back to UAC which is correlated to that INVITE. For this specification, that is only the final 2xx response to that INVITE. That same exact answer MAY also be placed in any provisional responses sent prior to the answer. The UAC MUST treat the first session description it receives as the answer, and MUST ignore any session descriptions in subsequent responses to the initial INVITE. So, I always understood that the first answer is the final answer, absent a session-updating request cycle. On the other hand, RFC 3960 ("Early Media and Ringing Tone Generation in the Session Initiation Protocol (SIP)") Section 4 says: The application server model consists of having the UAS behave as an application server to establish early media sessions with the UAC. The UAC indicates support for the early-session disposition type (defined in [2]) using the early-session option tag. This way, UASs know that they can keep offer/answer exchanges for early media (early-session disposition type) separate from regular media (session disposition type). I take this, along with RFC 3959 Section 3, to imply an amendment to 3261 § 13.2.1, but I'm not sure. Regardless, I'm not convinced all UAs will handle this situation. So, what I would most prefer is a means of recycling the same answer port on the "back" side of RTPEngine while pivoting it on the front across branches/multiple offers. Is that possible? If so, how can it be achieved? Many thanks, -- Alex [1] As best as I understood how to use it; the documentation is a bit unclear! -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

Convert 183+SDP and 180+SDP into a single reply w/SDP
by yu＠yu-boot.ru 16 Jul '18

16 Jul '18

Hello. My uplink usually gives ringback as both 183 Progress with SDP and 180 Ringing also with (different) SDP. One of my clients can't handle that behavior and it just don't plays ringback at all. Is it possible to relay only last provisional response with SDP and drop anything else 183? Any ideas plz? Can't figure out even a logic itself. Thanks. -- sent from myMail for Android

5 7

Encapsulate and de-encapsulate SIP traffic
by William Phillips 16 Jul '18

16 Jul '18

I have a need to encapsulate SIP traffic depending on destination and obviously de-encapsulate traffic incoming from those destinations. The approach to dealing with outbound SIP traffic seems pretty straight forward, I create an appropriate routing block that determines if I need to encap the SIP message, if I do, I forward it off to my module. The incoming traffic isn't particularly complicated, I'm just not 100% sure on the way to give that de-encapsulated SIP message back to kamailio to be processed. It looks like I can achieve this using the TM module with the uac_req_t struct and tm_bind.request but I really only see it used for SIP text messages in modules like XMPP and SMS. Is this the appropriate approach to achieve what I need? Will

2 1

5.1.3 issue with rtpengine.reload
by Marcin Kowalczyk 16 Jul '18

16 Jul '18

Hi, I have Kam 5.1.3 running on top of Ubuntu Xenial docker image. Today I've added new RTPEngine set and tried to reload rtpengine from db but no luck. Command executes without any errors (but way too fast) root@dfa7aae0b6ef:/# time kamcmd rtpengine.reload real 0m0.002s user 0m0.000s sys 0m0.000s root@dfa7aae0b6ef:/# Last option would be restart but I would prefer not as has multiple WSS connections active. Do you have idea what I could try to reload rtpengine set?

2 1

File transfer with Kamailio IMS??
by Ashok kumar Marra 16 Jul '18

16 Jul '18

Hi, I m using cent-os 6.2, kamailio-5.1.2 and openims-hss. I am using rtpengine to relay voice, but I am unable to send files or chat messages using it. I looked at rtpengine logs, it was saying unknown codec for chat and file-transfer requests. Then, I thought of using Kamailio msrp , tried but no success. Does current Kamailio IMS support file transfer? If so, please give me suggestions on doing it. -- With Best Regards, Ashok Kumar.

2 1

rtpengine/kamailio teardown method
by Abdoul Osséni 16 Jul '18

16 Jul '18

Hello all, There are some rtpengine parameters that permit to send an RPC command to kamailio in order to terminate a dialog. - timeout = xx - b2b-url = http://xxxx:xx/RPC - xmlrpc-format = 1 Each rpc command sent to Kamailio exited with status 256. By checking the code of rtpengine, I see rtpengine seems to use "teardown" method. I think Kamailio does not know this method. Can you help me please ? Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167

2 1

Communication between Kamailio Proxy and Registrar Server
by amit Kumar 16 Jul '18

16 Jul '18

Hi All, As per our requirement, we want to configure one Kamailio as a " Kamailio PROXY" and 2nd Kamailio as a " Kamailio REGISTRAR". Whenever any INVITE request come at " Kamailio PROXY" we want to get the UA information from " Kamailio REGISTRAR " and send INVITE to particular UA. Could you please let us know how achieve this. Thanks, Amit

2 1

SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM)
by Aqs Younas 13 Jul '18

13 Jul '18

Greeting list, I am using two async functions(http_async_query) in configuration one for authentication and other one for push notifications. I am also using tsilio module to suspend the transaction and resumed it. I am resetting this flag 'T_ASYNC_SUSPENDED' before suspending the transaction. Everything is working fine except. When caller send CANCEL on the call, kamailio successfully completes sip flow at callee(B party) side but send back below packet towards caller. 2018/07/10 19:47:41.125893 172.31.21.87:5060 -> 39.38.156.111:59969 SIP/2.0 500 I'm terribly sorry, server error occurred (1/TM ) Via: SIP/2.0/UDP 39.38.156.111:59969 ;branch=z9hG4bK-524287-1---91c6114330fe6293;rport=59969 To: <sip:923335550609@sip.talkhomeappcall.com ;transport=UDP>;tag=9fe2d4c84d97a4af81aa65ca8fc51b3d-62cb From: <sip:923135082344@sip.talkhomeappcall.com;transport=UDP>;tag=d72eaf4c Call-ID: VkAwxpSgkKSGn5dXaXgn5A.. CSeq: 2 INVITE Content-Length: 0 root@sip:~# kamailio -V version: kamailio 5.1.3 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, 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 08:06:26 May 2 2018 with gcc 4.9.2 Could someone please help me out how to fix this. Cancels are handles automatically so, i cannot hardcore anything to resolve it. Best Regards, Aqs Younas

3 8

Re: [SR-Users] sr-users Digest, Vol 158, Issue 12
by Ne Mo 13 Jul '18

13 Jul '18

Thanks Daniel! I'm folow: https://www.kamailio.org/docs/modules/stable/modules/ dispatcher.html#dispatcher.ex.install but my kamailio get errors: parameter <dstid_avp> of type <1> not found in module <dispatcher> parameter <attrs_avp> of type <1> not found in module <dispatcher> parameter <sock_avp> of type <1> not found in module <dispatcher> parameter <cnt_avp> of type <1> not found in module <dispatcher> I have no infor how to fix it Can you show me more detail about how to do: - *Fix error* * - Set the socket attribute to your private ip socket to do bridge from external to internal networks. * (just add to "attrs(str,opt)", and in my situation, it is my private IP of Kamailio server1 ? forexample: 1 sip:<PublicIP>:<Port> 0 0 duid=kamilioserver1;socket=<protocol>:<KamailioServer1_PrivateIP>:<Port>;my=xyz) (I don't understand what my=xyz mean..) * - Do i have to do something in Kamailio server2 and KamailioServer3 (or just create kamailio database each server and synchronyze kamailio database between them?)* *Thanks!* And thanks Mojtaba! I had read your recomment tutorial with OpenSIP, but do not know how to apply to my system. Have you got an other more clearly tutorial for this? On Thu, Jul 12, 2018 at 5:00 PM, <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. UAC register (Ben Hood) > 2. Re: UAC register (Daniel-Constantin Mierla) > 3. Re: UAC register (Daniel-Constantin Mierla) > 4. dispatcher module complete tutorial (SamSung TV) > 5. Re: UAC register (Ben Hood) > 6. Re: UAC register (Ben Hood) > 7. Re: dispatcher module complete tutorial (Mojtaba) > 8. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Aqs Younas) > 9. Re: dispatcher module complete tutorial (Daniel-Constantin Mierla) > 10. Re: UAC register (Daniel-Constantin Mierla) > 11. Re: UAC register (Daniel-Constantin Mierla) > 12. Re: SIP/2.0 500 I'm terribly sorry, server error occurred > (1/TM) (Daniel-Constantin Mierla) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 11 Jul 2018 12:36:36 +0100 > From: Ben Hood <ben(a)relops.com> > To: sr-users(a)lists.sip-router.org > Subject: [SR-Users] UAC register > Message-ID: <7B1A15CF-7BCF-4BCC-8DF0-20E2906BA1DA(a)relops.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > I’m wondering if there is a flag somewhere in the UAC module config that > prevents remote registration from happening. > > It looks like I’ve loaded the uacreg table but I can’t see any REGISTER > methods being sent from Kamailio to the remote peer. > > The remote registrants are loaded using this config: > > listen=udp:172.16.176.150:5060 > > loadmodule "db_postgres.so" > loadmodule "kex.so" > loadmodule "corex.so" > loadmodule "tm.so" > loadmodule "tmx.so" > loadmodule "sl.so" > loadmodule "rr.so" > loadmodule "pv.so" > loadmodule "maxfwd.so" > loadmodule "textops.so" > loadmodule "siputils.so" > loadmodule "xlog.so" > loadmodule "sanity.so" > loadmodule "ctl.so" > loadmodule "cfg_rpc.so" > loadmodule "counters.so" > loadmodule "uac.so" > > modparam("uac", "reg_db_url", DBURL) > modparam("uac", "reg_contact_addr", “172.16.176.150:5060") > modparam("uac", "reg_db_table", "uacreg") > modparam("uac", "reg_timer_interval", 10) > modparam("uac", "reg_retry_interval", 10) > > request_route { > route(RELAY); > } > > route[RELAY] { > > t_on_reply("MANAGE_REPLY"); > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > And it looks like this has been consumed by Kamailio: > > $ kamcmd uac.reg_dump > { > l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > l_username: 126231943099723776 > l_domain: 172.16.176.150 > r_username: 1000 > r_domain: 172.16.176.129 > realm: 172.16.176.129 > auth_username: 1000 > auth_password: password > auth_proxy: sip:172.16.176.129:5080 > expires: 0 > flags: 0 > diff_expires: -1531235076 > timer_expires: 0 > reg_init: 1531234841 > reg_delay: 0 > } > > Doing a refresh successfully reloads the uacreg data > > $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f > > I can get Kamailio to forward an OPTIONS message to the remote peer () to > verify network connectivity, e.g. : > > $ru = "sip:1000@172.16.176.129:5080"; > $fs = "udp:172.16.176.150:5060"; > > But Kamailio does not appear to generate REGISTER messages either after > first boot or as a result of the manual refresh. > > This is using an instance of Kamailio 5.0.1. Should I try with a newer > version? > > Is there a way to get extra debugging of the uac module? > > If I increase the debug verbosity, I get quite granular detail for the DB > traffic, but the uac module doesn’t appear to emit much diagnostics. > > Many thanks, > > Ben > > > > > >

2 1

UAC register
by Ben Hood 12 Jul '18

12 Jul '18

Hi, I’m wondering if there is a flag somewhere in the UAC module config that prevents remote registration from happening. It looks like I’ve loaded the uacreg table but I can’t see any REGISTER methods being sent from Kamailio to the remote peer. The remote registrants are loaded using this config: listen=udp:172.16.176.150:5060 loadmodule "db_postgres.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "counters.so" loadmodule "uac.so" modparam("uac", "reg_db_url", DBURL) modparam("uac", "reg_contact_addr", “172.16.176.150:5060") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) request_route { route(RELAY); } route[RELAY] { t_on_reply("MANAGE_REPLY"); if (!t_relay()) { sl_reply_error(); } exit; } And it looks like this has been consumed by Kamailio: $ kamcmd uac.reg_dump { l_uuid: 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f l_username: 126231943099723776 l_domain: 172.16.176.150 r_username: 1000 r_domain: 172.16.176.129 realm: 172.16.176.129 auth_username: 1000 auth_password: password auth_proxy: sip:172.16.176.129:5080 expires: 0 flags: 0 diff_expires: -1531235076 timer_expires: 0 reg_init: 1531234841 reg_delay: 0 } Doing a refresh successfully reloads the uacreg data $ kamcmd uac.reg_refresh 2cc6776d-61ed-4ea1-a3ae-055f89bd4d3f I can get Kamailio to forward an OPTIONS message to the remote peer () to verify network connectivity, e.g. : $ru = "sip:1000@172.16.176.129:5080"; $fs = "udp:172.16.176.150:5060"; But Kamailio does not appear to generate REGISTER messages either after first boot or as a result of the manual refresh. This is using an instance of Kamailio 5.0.1. Should I try with a newer version? Is there a way to get extra debugging of the uac module? If I increase the debug verbosity, I get quite granular detail for the DB traffic, but the uac module doesn’t appear to emit much diagnostics. Many thanks, Ben

2 3

dispatcher module complete tutorial
by SamSung TV 12 Jul '18

12 Jul '18

I want to use kamailio as load balancing server my topo: (Private IP) Kamailio server2 / / (public iP)Kamailio server1(load balance)(Private IP) \ \ (Private IP) Kamailio server1 Any body can recomment me some tutorial to do this? Thanks any advance!

3 2

Kamailio, forking and multiple answer
by Marino Mileti 10 Jul '18

10 Jul '18

Hi guys, i've a Kamailio with forking scenario using addbranch technique.Now I've a problem; i know it's very very very rare but it could be possible so it's better for me to find a solution. A -> B,C,D in forking parallel mode, so B C and D rings simultaneulsy. What is happen if B anser the call, and at the same time C decline the call?It seems that, using my cfg scripts, C sends DECLINE but this message never arrive to A because it receives a 200OK of B. How I can prevent this? I would like that if DECLINE is the first message, all other answers will be ignored (first answer win). Any help or comment? ThanksMarino

2 1

TLS CRL configuration
by Amarnath Kanchivanam 10 Jul '18

10 Jul '18

Hi All, I'm trying to configured kamailio as TLS server with below configuration (tls.cfg) and TLS server is started successfully. [server:default] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = ./sip/server.key certificate = ./sip/server.crt ca_list = ./bundle.crt crl = ./sip_crl.pem verify_depth = 9 [client:default] verify_certificate = no require_certificate = no TLS connection works fine. Later i have updated the sip_crl.pem with server certificate revoked details and performed tls.reload command to load the latest update. After this I expect any TLS client trying to establish TLS connection should fail, as the client and server certificates are signed by same authority and server certificate is revoked. But the clients are able to establish TLS connection without any errors. I'm not getting any traces to confirm CRL validation has been performed before accepting the TLS connection. Any advice would be help to proceed with evaluating CRL functionality. -Amar

3 5

CPS
by Sergiu Pojoga 10 Jul '18

10 Jul '18

Hi folks, Trying to implement CPS throttling based on this <https://lists.kamailio.org/pipermail/sr-users/2013-September/079713.html>article, having some trouble. modparam("htable", "htable", "rhs=>size=32;initval=0;autoexpire=300;" modparam("htable", "htable", "rhm=>size=32;initval=0;autoexpire=1800;") route[CPS] { $var(rateHashSec) = $si+":sec:"+$timef(%Y/%m/%d_%H_%M_%S); $var(rateHashMin) = $si+":min:"+$timef(%Y/%m/%d_%H_%M_00); xlog("L_INFO", "var(rateHashSec)=$var(rateHashSec); var(rateHashMin)=$var(rateHashMin)\n"); $var(ratePerSec) = $shtinc(rhs=>$var(rateHashSec)); $var(ratePerMin) = $shtinc(rhm=>$var(rateHashMin)); xlog("L_INFO", "var(ratePerSec)=$var(ratePerSec); var(ratePerMin)=$var(ratePerMin)\n"); $var(limitPerSec) = 1; $var(limitPerMin) = 1; if ($var(ratePerSec) > $var(limitPerSec) || $var(ratePerMin) > $var(limitPerMin)) { xlog("L_INFO", "CPS Limit on IP $si\n"); send_reply("503", "CPS Limit on IP $si"); exit; } } Logs show hashes are created but *$var(ratePerSec/Min)* won't increment no matter how many calls I place per second or minute: Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(rateHashSec)=65.39.XX.XX:sec:2018/07/09_14_35_52; var(rateHashMin)=65.39.XX.XX:min:2018/07/09_14_35_00 Jul 9 14:35:52 proxy /usr/local/sbin/kamailio[1484]: INFO: <script>: var(ratePerSec)=0; var(ratePerMin)=0 Much obliged. --Sergiu

2 3

Delay in 100 Trying response for INVITE when the load is 70 INVITE's per sec
by vivek＠advaitamtech.com 09 Jul '18

09 Jul '18

Hi All, When we are using the kamailio to handle around 70 INVITE's per second, kamailio is sending 100 Trying with the delay of around 3-4sec for few INVITE's. How can I reduce the delay in 100 Trying? Also, I tried increasing the number of threads for SIP routing processes. This doesnt seem to help. Also tried "modparam("tm", "auto_inv_100", 0)" . This is not working. Please help. Thank you, Vivek.

3 2

RTPEngine-Recording-daemon
by Mojtaba 09 Jul '18

09 Jul '18

Hello, In RTPEngine recording-daemon, I enabled "forward-to" option to unix_socket. All things work right, I got metadata at first, then media packets (RTP-UDP-IP). The metadata for each calls is received just at first packet, but i want to know, if i have more than one concurrent active call in RTPEngine, How can i distinct which media packet was from which call? Thanks with Regards. --Mojtaba Esfandiari.S

2 2

ul.add | path & socket not correctly set
by Laurent Schweizer 09 Jul '18

09 Jul '18

Dear all, I have kamailio 5.0.6 and I have an issue to add permanent location When I add a location with : kamcmd ul.add location_ch demo(a)peoplefone.ch sip: demo@95.128.80.122:5060 0 0.0 . 0 128 6111 . udp:95.128.80.3:5060 then I get the following insertion: kamcmd ul.lookup location_ch demo@ { AoR: demo Contacts: { Contact: { Address: sip:demo@95.128.80.122:5060 Expires: permanent Q: 0.000000 Call-ID: dfjrewr12386fd6-343(a)kamailio.mi CSeq: 1 User-Agent: SIP Router MI Server Received: [not set] Path: . State: CS_SYNC Flags: 0 CFlags: 128 Socket: [not set] Methods: 6111 Ruid: ulcx-5b243cb5-67c6-31 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: 0 Keepalive: 0 Last-Keepalive: 0 Last-Modified: 0 } } As you can see the Path value is set to "." But it must be set to [not set] The Socket is also not set. Any idea ? BR Laurent

2 3

RFC: database unsigned number types
by Daniel-Constantin Mierla 07 Jul '18

07 Jul '18

Hello, starting here a discussion about an issue with the database API and signed/unsigned number type to see how to address it. So far, the database API supported only signed types for numbers, respectively DB1_INT (int in c) and DB1_BIGINT (long long in c). However, many database table columns are defined as UNSIGNED INT or UNSIGNED BIGINT. The database connector modules are mapping UNSIGED values retrieved from database over signed fields in DB API structure. The other way around is done as well: signed values are the inserted in the unsigned columns. There is no issue if the value in C is a positive number, however, if the value goes over MAX_UINT/2 (over 2147483647), it becomes negative and inserting the value in database results in an exception and 0 being stored instead. So far, I guess the issue was rarely exposed, if at all, because no report on it, even these data types for DB1 are since the SER project was started in 2001. As I looked at database definition schema, most of unsigned columns are for internal flags or ids (e.g., lcr_id), where I guess no large values were used or needed so far. However, it can bite at any time and needs to be addressed. So far, two solutions come in mind: 1) drop using UNSIGNED INT for those db columns, use only INT and do unsigned cast in the C code when reading and cast to int when writing. 2) update the db connector modules to support unsigned types -- I added support for them in DB API, but each db_* module has to be updated. The also each module that uses UNSIGNED DB columns must be updated 1) should be simpler, 2) more work but better in long term Any other ideas? Which of the options you prefer to go for? Not to forget: somehow related, probably we have to switch from int to long for PV number values, otherwise timestamp variables can go negative once unix timestamp approaches MAX_UINT/2 (still plenty of time, but should not be delayed for long ...). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

2 1

sdpops / keep_codecs_by_name not modifying sdp unless i call msg_apply_changes
by Enrico Bandiera 06 Jul '18

06 Jul '18

Hi, I'm encountering an issue where calling keep_codecs_by_name does not modify the SDP on the INVITE unless I call the "evil function" msg_apply_changes Any help is welcome, Thanks, Enrico.

2 2

rtpengine installation dependencies
by Aqs Younas 05 Jul '18

05 Jul '18

Greetings list. This might not be related to Kamailio but I thought someone would be able to give a hand in resolving dependency during the installation of rtpengine on Debian 8 Jessie. It is complaining about below dependencies. root@debian-769mb-miami-01:/usr/src/rtpengine# dpkg-buildpackage dpkg-buildpackage: source package ngcp-rtpengine dpkg-buildpackage: source version 6.3.0.0+0~mr6.3.0.0 dpkg-buildpackage: source distribution unstable dpkg-buildpackage: source changed by Sipwise Jenkins Builder < jenkins(a)sipwise.com> dpkg-buildpackage: host architecture amd64 dpkg-source --before-build rtpengine dpkg-source: info: using options from rtpengine/debian/source/options: --extend-diff-ignore=.gitreview dpkg-checkbuilddeps: *Unmet build dependencies: debhelper (>= 10~) libbcg729-dev* dpkg-buildpackage: warning: build dependencies/conflicts unsatisfied; aborting dpkg-buildpackage: warning: (Use -d flag to override.) I tried to install debhelper from deb but complained from dh-autoreconf and that itself complained abount debhelpler. So, no one is being installed. Also, If someone can help me how to resolve dependency for libbcg729-dev too. Any pointer or hint or link to some installation guide would be more than welcome. Br, Aqs.

9 17

RTPEngine Recording does not work
by Mojtaba 05 Jul '18

05 Jul '18

Hi all, I installed RTPEngine (Version: 6.4.0.0+0~mr6.4.0.0 git-master-4eb80da) in my VM machine (Debian GNU/Linux 8 (jessie), i also installed kamailio 5.1. In kamailio routes, i have: route { ... #rtpengine_manage("record-call") rtpengine_manage(); start_recording(); ... } Here is all other configuration files: In /etc/default/ngcp-rtpengine-daemon file: RUN_RTPENGINE=yes CONFIG_FILE=/etc/rtpengine/rtpengine.conf CONFIG_SECTION=rtpengine PIDFILE=/var/run/ngcp-rtpengine-daemon.pid MANAGE_IPTABLES=yes TABLE=0 In /etc/default/ngcp-rtpengine-recording-daemon file: RUN_RTPENGINE_RECORDING=yes CONFIG_FILE=/etc/rtpengine/rtpengine-recording.conf CONFIG_SECTION=rtpengine-recording PIDFILE=/var/run/ngcp-rtpengine-recording-daemon.pid MUST_NFS=no NFS_HOST=192.168.1.1 NFS_REMOTE_PATH=/var/recordings NFS_LOCAL_MOUNT=/var/lib/rtpengine-recording # must match output-dir if used NFS_OPTIONS=hard,intr,tcp In /etc/rtpengine/rtpengine.conf file: [rtpengine] table = 0 interface = 192.168.122.200 listen-ng = 127.0.0.1:2223 recording-dir = /var/spool/rtpengine/ recording-method = proc In [rtpengine-recording] table = 0 # output-storage = db (use default) # output-format = mp3 (use default) # output-mixed = 1 spool-dir = /var/spool/rtpengine/metadata output-dir = /var/spool/rtpengine/recording But the recording is not work, I have these issues in syslog: [1530178531.860171] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:38654 [1530178531.860191] NOTICE: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Turning on call recording. [1530178531.860239] WARNING: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Call recording through /proc interface requested, but kernel table not open [1530178531.860262] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860269] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860275] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860281] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860287] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860292] ERR: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Failed to open recording metadata file '(null)' for writing: Bad address [1530178531.860303] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:38654 (elapsed time 0.000112 sec) [1530178531.969040] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:39374 [1530178531.969069] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:39374 (elapsed time 0.000004 sec) [1530178537.176906] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'delete' from 127.0.0.1:40258 [1530178537.177003] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Scheduling deletion of call branch '9052b05a' (via-branch '') in 30 seconds [1530178537.177026] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'delete' from 127.0.0.1:40258 (elapsed time 0.000084 sec) [1530178537.178368] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Received command 'start recording' from 127.0.0.1:40258 [1530178537.178392] INFO: [MWFhMWM4MmVhYjFmYjY3MzVlZDlmZWMyYjNmNGVhNmY.]: Replying to 'start recording' from 127.0.0.1:40258 (elapsed time 0.000004 sec) ^C[1530178543.752740] INFO: Version 6.4.0.0+0~mr6.4.0.0 git-master-4eb80da shutting down Also when i start RTPEngine, I have FATAL error about module xt_RTPENGINEer in syslog: Jun 28 05:44:58 debian8 systemd[1]: Starting NGCP RTP/media Proxy Daemon... Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: modprobe: FATAL: Module xt_RTPENGINE not found. Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: iptables: No chain/target/match by that name. Jun 28 05:44:58 debian8 ngcp-rtpengine-iptables-setup[10203]: ip6tables: No chain/target/match by that name. Jun 28 05:44:58 debian8 systemd[1]: Started NGCP RTP/media Proxy Daemon. Jun 28 05:44:58 debian8 rtpengine[10219]: [1530179098.989322] INFO: Generating new DTLS certificate Jun 28 05:44:59 debian8 rtpengine[10219]: [1530179099.033499] ERR: FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED J Let me know what is the problem and how can i solve it? -- --Mojtaba Esfandiari.S

4 7

Video call recording and mobile push notifications
by Jeremy Renner 05 Jul '18

05 Jul '18

Hi, I would like to setup up my SIP server / PBX for my business, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Kamailio can works as a PBX ? 2. If yes, does the Kamailio support push notifications and video recording ? 3. Does the Kamailio can works for Multi-tenant ? 4. Does Kamailio support Windows ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards,

3 3

Re-write From URI Domain ($fd) with source IP or relayed message
by Marrold 04 Jul '18

04 Jul '18

Hi All, For interoperability reasons I need to re-write the From URI Domain ($fd) in requests proxied by Kamailio to the source IP (socket) that Kamailio will send them via. I have experimented with onsend_route and can successfully discover the snd_ip, however it seems I am unable to rewrite the $fd, presumably because I am too late in the logic. Is this possible? I can hardcode IPs if required but would rather keep it dynamic if I can. Thanks in advance, Matthew

2 2

Unable to insert impu into table
by eyas barhouk 04 Jul '18

04 Jul '18

Hello community , on kamailio server i'm using the module "ims_usrloc_scsc" with the following configuration : ......................................................... modparam("ims_usrloc_scscf", "db_url", "mysql://eyas:eyas@localhost/scscf") modparam("ims_usrloc_scscf", "db_mode", 1) modparam("ims_usrloc_scscf", "enable_debug_file", 0) modparam("ims_usrloc_scscf", "matching_mode", 0) modparam("ims_registrar_scscf", "error_reply_code", 503) .......................................................... when i'm running kamailio and try to register i found the following : Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [usrloc_db.c:235]: db_insert_impurecord(): Unable to insert impu into table [sip:09876993998148@NET.SYRIATEL] Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [udomain.c:580]: insert_impurecord(): error inserting contact into db Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_usrloc_scscf [impurecord.c:974]: update_impurecord(): Unable to insert new IMPU for <sip:09876993998148@NET.SYRIATEL> Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_registrar_scscf [save.c:809]: update_contacts(): Unable to update impurecord for <sip:09876993998148@NET.SYRIATEL> Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: ims_registrar_scscf [cxdx_sar.c:253]: async_cdp_callback(): Error processing REGISTER Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [-1] Jul 04 16:03:59 S-CSCF [S-CSCF][3796]: ERROR: <script>: SAR failure - error response sent from module Jul 04 16:04:04 S-CSCF [S-CSCF][3697]: INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [-2] Jul 04 16:04:04 S-CSCF [S-CSCF][3814]: INFO: ims_auth [cxdx_avp.c:137]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp (avp_code = 297, vendor_id = 0) Jul 04 16:04:04 S-CSCF [S-CSCF][3814]: INFO: ims_auth [cxdx_mar.c:79]: create_return_code(): created AVP successfully : [maa_return_code] - [1] Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_auth [authorize.c:834]: authenticate(): uri=sip:NET.SYRIATEL;transport=TCP nonce=cae12e7c256b8a052ad9824ebb91fa07 response=ee82f8f8ad6bf4530d210144e Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_auth [authorize.c:886]: authenticate(): UE said: ee82f8f8ad6bf4530d210144eaf3c413 and we expect ee82f8f8ad6bf4530d210144eaf3c413 ha1 fd5e9c83ecc96ee Jul 04 16:04:04 S-CSCF [S-CSCF][3694]: INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [-2] Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [usrloc_db.c:235]: db_insert_impurecord(): Unable to insert impu into table [sip:09876993998048@NET.SYRIATEL] Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [udomain.c:580]: insert_impurecord(): error inserting contact into db Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_usrloc_scscf [impurecord.c:974]: update_impurecord(): Unable to insert new IMPU for <sip:09876993998048@NET.SYRIATEL> Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_registrar_scscf [save.c:809]: update_contacts(): Unable to update impurecord for <sip:09876993998048@NET.SYRIATEL> Jul 04 16:04:04 S-CSCF [S-CSCF][3819]: ERROR: ims_registrar_scscf [cxdx_sar.c:253]: async_cdp_callback(): Error processing REGISTER could any one tell me where was my wrong and why it can't write on the DB & thanks in advance

1 0

Kamailio Anycast
by Mike Little 03 Jul '18

03 Jul '18

Hello, Sorry if this has already been answered but can Kamailio be configured to support clustering and use Anycast? We are looking to use Kamailio as a front end to Asterisk as the registar and presence server. If this is supported, are there any documents that explain how to configure this type of setup? Thanks!

2 1

IMS usage of Kamailio
by Akib Sayyed 03 Jul '18

03 Jul '18

Dear All I am willing to try Kamailio as IMS. before i try i would like to know following points - in Kamailio i see diameter support and authentication I-CSCF, P-CSCF, S-CSCF charging, QOS, ISC implemented but what about TAS do i need TAS seperately - I see that there is HOW-TO wiki , in how to wiki you are using FHoSS as HSS , can we use commercial/any other hss (afaik its using diameter so any HSS will do)? - Is there any configuration online available kamailio as IMS? - Are all IMS modules of Kamailio can be installed on one server or need each module on same machine (i am using IMS only for testing 1 or 2 calls at a time)

1 0

Having trouble getting presence working with Polycom VVX phones
by Justin Foley 03 Jul '18

03 Jul '18

Hey everyone. I've set up a kamailio server on a local dev box and running the phones on the same subnet as it to avoid any NAT issues. Seems to work fine, phones register & can call eachother and hear voice but the issue I'm running into is I can't seem to get presence to work. I've enabled it in the kamailio.cfg file with #!WITH_PRESENCE but if I go into "do not disturb" or try to call on phone1, it's not indicated on phone2. Also running ngrep on port 5060, I'm not seeing the anything regarding presence being sent over the SIP protocol.

2 2

$rs printed null
by Andrew Chen 03 Jul '18

03 Jul '18

Hi community, So I'm running 4.1.5 Kam version and I wanted to print out $rs variable. It printed out <null> right after the 183 session. Should it print something? What our server side sent is this: INVITE 100 183 488 or 487 And this is the output in kam logs: Jun 28 15:05:49 sjomainfnsg51 /usr/local/sbin/*kamailio*[15215]: INFO: <script>: [4975Fz-ses1-414b02f5R68422169(a)192.168.0.22][2][CSeq 1] fuze_for_mac/99.99.99999 Reply status <null> detected... any thoughts? Thanks -- Andy Chen -- *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 4

Multiple aliases in the Request Message
by Borislav Gergov 03 Jul '18

03 Jul '18

Hello, what is the proper way for kamailio nathelper module to handle multiple aliases in the request . For example We have ACK message that looped in the kamailio because is not forward to user phone: ACK sip:xxxxxx@10.10.10.32:38416;alias=192.168.129.135~5060~1;alias=192.168.205.32~38416~1 SIP/2.0 Via: SIP/2.0/UDP 192.168.146.174:46248 ;branch=z9hG4bK-524287-1---5b14eec905adf6ce Max-Forwards: 70 Route: <sip:mo@192.168.129.133;lr=on;ftag=c7d62d68;did=8a1.bcf2;rm=8> Route: <sip:mo@192.168.129.135;lr=on;ftag=c7d62d68;did=8a1.121> Route: <sip:mt@192.168.129.135;lr=on;ftag=c7d62d68;did=8a1.221> Route: <sip:mt@192.168.129.133;lr=on;ftag=c7d62d68;did=8a1.ccf2;rm=7> Contact: <sip:yyyyy@10.153.146.174:46248;transport=UDP> To: <sip:xxxxxx@domain.net;transport=UDP>;tag=dd9aaf1c From: <sip:yyyyy@domain.net;transport=UDP>;tag=c7d62d68 Call-ID: 5Zogf8chhpYlQRdpaZM4CA.. CSeq: 2 ACK User-Agent: Softphone Content-Length: 0

2 1

Auth and default kami config
by yu＠yu-boot.ru 03 Jul '18

03 Jul '18

Hello. Have a question on auth and default kamailio.cfg. Default behavior of kamailio on auth is: Auth initial requests, delete auth headers before forwarding to next hop. In-dialog requests are forwarded "as-is" so all auth related info is propagated to uplinks or just another Kami users. How should it be done correctly or "kamailio-way"? As for now, I just delete auth headers for all in-dialog requests, but I'm not sure it is correct. I looked how it works on softswitches like Yate. Registers and initial Invites are authenticated, ACK is not authenticated, but all other requests like reINVITE and BYE are. How is should be done? Thanks. -- sent from myMail for Android

2 1

[Kamailio]: Parse and Store nosip Message error.
by Zhong, Alec (NSB - CN/Qingdao) 03 Jul '18

03 Jul '18

Hi, We are developing a project for parser and store MEGACO message by kamailio. During the testing, we find that kamailio inserts one message only once, but there are duplicated messages in mysql datebase. Did you meet the similar issue before? Could you please share some information for us? Thank you in advance. [Environment/version] Environment: Debian-Jessie, docker image Version: kamialio 5.0.6 docker image. [Description] Kamailio stores one message by calling mysql function once, but there are two same message in DB randomly. Function flow: core->nosip parser-> nosip store. [Kamailio Log] Jun 26 07:17:49 dnt02-elk-control-worker-edge-01 journal: }]#033[0;39;49m#033[0;33;49m 5(22) CRITICAL: nosip [megaco_capture.c:49]: megaco_error_code(): No error in MEGCO! Jun 26 07:17:49 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: nosip [megaco_capture.c:406]: megaco_capture(): No error code.#033[0;39;49m#033[0;33;49m 5(22) CRITICAL: nosip [megaco_capture.c:269]: megaco_capture_store(): az_megaco_tmstamp(1529997468887999)//debug log for message, this is duplicated message in mysql. Jun 26 07:17:49 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: nosip [megaco_capture.c:305]: megaco_capture_store(): insert into homer table: [megaco_capture_all_20180626] Jun 26 07:17:49 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: nosip [megaco_capture.c:308]: megaco_capture_store(): storing info... Jun 26 07:17:50 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:50.802482 1 event.go:218] Event(v1.ObjectReference{Kind:"PersistentVolumeClaim", Namespace:"default", Name:"homer-data-dashboard", UID:"136b3329-78f1-11e8-8d74-0050568fb61f", APIVersion:"v1", ResourceVersion:"2841229", FieldPath:""}): type: 'Warning' reason: 'ProvisioningFailed' storageclass.storage.k8s.io "glusterfs-storageclass" not found Jun 26 07:17:50 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:50.802660 1 event.go:218] Event(v1.ObjectReference{Kind:"PersistentVolumeClaim", Namespace:"default", Name:"homer-data-semaphore", UID:"1369de0b-78f1-11e8-8d74-0050568fb61f", APIVersion:"v1", ResourceVersion:"2841228", FieldPath:""}): type: 'Warning' reason: 'ProvisioningFailed' storageclass.storage.k8s.io "glusterfs-storageclass" not found Jun 26 07:17:50 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:50.802706 1 event.go:218] Event(v1.ObjectReference{Kind:"PersistentVolumeClaim", Namespace:"default", Name:"homer-data-mysql", UID:"136e804c-78f1-11e8-8d74-0050568fb61f", APIVersion:"v1", ResourceVersion:"2841231", FieldPath:""}): type: 'Warning' reason: 'ProvisioningFailed' storageclass.storage.k8s.io "glusterfs-storageclass" not found Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:51.223950 1 trace.go:76] Trace[1532720894]: "GuaranteedUpdate etcd3: *core.Endpoints" (started: 2018-06-26 07:17:48.964454076 +0000 UTC m=+1830051.184603352) (total time: 2.259414864s): Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Trace[1532720894]: [2.259374311s] [2.258991423s] Transaction committed Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:51.224190 1 trace.go:76] Trace[557711900]: "Update /api/v1/namespaces/kube-system/endpoints/kube-scheduler" (started: 2018-06-26 07:17:48.964209184 +0000 UTC m=+1830051.184358457) (total time: 2.259905913s): Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Trace[557711900]: [2.259791237s] [2.259619876s] Object stored in database Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: 2018-06-26 07:17:51.220413 W | wal: sync duration of 1.446041666s, expected less than 1s Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: I0626 07:17:51.225704 1 trace.go:76] Trace[656020708]: "GuaranteedUpdate etcd3: *core.Node" (started: 2018-06-26 07:17:50.573518495 +0000 UTC m=+1830052.793667844) (total time: 652.126103ms): Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Trace[656020708]: [651.833583ms] [647.806507ms] Transaction committed Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 5(22) CRITICAL: <core> [core/xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/parse_fline.c:244]: parse_first_line(): parse_first_line: bad request first line Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/parse_fline.c:246]: parse_first_line(): at line 0 char 29: Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/parse_fline.c:252]: parse_first_line(): parsed so far: MEGACO/2 [175.15.0.254]:32791 Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 29) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/msg_parser.c:601]: parse_msg(): invalid message Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/parser/msg_parser.c:676]: parse_msg(): ERROR: parse_msg: message=<MEGACO/2 [175.15.0.254]:32791 Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Transaction = 359 { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Context = - { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Notify = ROOT { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: ObservedEvents = 1111 { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: 20180626T15175177: Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: it/ito Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: }> Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/pvapi.c:1377]: pv_printf(): no more space for spec value Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/pvapi.c:1386]: pv_printf(): buffer overflow -- increase the buffer size... Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:344]: megaco_capture(): Dump msg:[MEGACO/2 [175.15.0.254]:32791 Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Transaction = 359 { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Context = - { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: Notify = ROOT { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: ObservedEvents = 1111 { Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: 20180626T15175177: Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: it/ito Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: } Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: }]#033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:49]: megaco_error_code(): No error in MEGCO! Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:406]: megaco_capture(): No error code.#033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:269]: megaco_capture_store(): az_megaco_tmstamp(1529997471897950)//debug log, this is the non-dup message. Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:305]: megaco_capture_store(): insert into homer table: [megaco_capture_all_20180626] Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: nosip [megaco_capture.c:308]: megaco_capture_store(): storing info... Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jun 26 07:17:51 dnt02-elk-control-worker-edge-01 journal: #033[0;39;49m#033[0;33;49m 3(20) CRITICAL: <core> [core/usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Best Regards, Alec Zhong

2 1

Re: [SR-Users] Understanding 5XX stats in Kamailio
by Konstantin Polyakov 03 Jul '18

03 Jul '18

Hello Daniel, Is there information about stats - what is implemented, what is not implemented? I have the similar questions, it would be nice to be informed what stats can be use and what stats should be skipped. Thank you. > > >Message: 8 >Date: Fri, 29 Jun 2018 09:46:21 +0200 >From: Daniel-Constantin Mierla < miconda(a)gmail.com > >To: "Kamailio (SER) - Users Mailing List" >< sr-users(a)lists.kamailio.org >, Joel Serrano < joel(a)textplus.com > >Subject: Re: [SR-Users] Understanding 5XX stats in Kamailio >Message-ID: < cbd84f94-8bb1-ddc3-b901-5743360c1fb9(a)gmail.com > >Content-Type: text/plain; charset=utf-8 > >Hello, > >I haven't implemented those stats, but should be like: > > >On 28.06.18 23:54, Joel Serrano wrote: >> Does anyone know the difference between these metrics?? >> >> core:rcv_replies_5xx > >- the number of 5xx replies received by kamailio > >> sl:5xx_replies > >- the number of 5xx replies generated by kamailio via sl module functions > >> tmx:5xx_transactions > >- the number of transactions that completed with a 5xx response > >> >> Specially between core and tmx? > >Some of received 5xx responses may be due to stateless forwarding, a >matter of config file, thus some of them do not match transactions. > >Cheers, >Daniel > >-- >Daniel-Constantin Mierla -- www.asipto.com >www.twitter.com/miconda -- www.linkedin.com/in/miconda >Kamailio World Conference -- www.kamailioworld.com > > > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >sr-users mailing list >sr-users(a)lists.kamailio.org >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > >------------------------------ > >End of sr-users Digest, Vol 157, Issue 30 >***************************************** С уважением, Константин Поляков.

4 3

kamailio with kannel
by eyas barhouk 02 Jul '18

02 Jul '18

hello dears , i'm trying to use "http_client" module to post SMS from kamailio to kannel using the following configuration : ....... modparam("http_client", "httpcon", "kannel=>http://username:password@127.0.0.1:443") .......... ......... http_connect("kannel","cgi-bin/sendsms?user=username&pass=password&charset=utf-16&coding=2&from=$avp(from)&to=$avp(to)&text=$(avp(text){s.escape.user})","$var(result)"); ....... And this test success for English characters & phrases, but when i'm trying to send Arabic or Chinese characters it only success if the content of the SMS only has one word. in other-word i mean when i'm trying to send more than one Arabic word with a space between each word it reply with HTTP error code 400. so do you have any ideas about this problem ??

2 1

Using Kamailio+rtproxy with freeswitch
by Philippe,Edson H 02 Jul '18

02 Jul '18

Hello. I have been trying to use kamailio+rtpproxy to handle freeswitch. So far, I am able to handle the sip routing via Kamailio, however, I am not able to get the RTP portion to go through my kamailio server. it seems once Kamailio handle the initial sip, there is a 1 to 1 communication between freeswitch and my client (phone) Has anyone solved this problem?

2 1

subst query
by mahesh b 02 Jul '18

02 Jul '18

Hi , Can someone explain me what is \1 and \3 doing in the subst expression ? Input : From: <sip:anonymous@10.211.160.168 >;tag=41008079_nab_FFFF_isp_FFFF_cco_FFFF_igo_FFFF_mgt_78DD subst('/^From:(.*)10.211.160.168(.*)>(.*)/From:\110.211.160.174>\3/ig'); Output : From: <sip:anonymous@10.211.160.174 >;tag=41008079_nab_FFFF_isp_FFFF_cco_FFFF_igo_FFFF_mgt_78DD Regards, Mahesh.b

2 5

Should I ignore Route header in ACK?
by Yuriy Gorlichenko 02 Jul '18

02 Jul '18

Hi I know that this is not question too much close to the kamialio users but mostly losed to the RFC specifiacations but this community looks like pretty much close to it that is why I want to ask this question here, that's why sorry and thanks for help in this question: I have a situation when provider sends me 200 response with Request-Route header and changed contact header: Means response comes from 1.1.1.1:5060 Request-Route contains: 1.1.1.1:5060 But Contact contains: 1.1.1.1:5061 My ACK (handled by kamailio) goes to the 1.1.1.1:5060 as it setted up at the Route Hedaer of ACK (because of Request-Route) but provider says me that i should use Contact for the ACK I was surprised because of https://tools.ietf.org/html/rfc3261#section-12.2.1.1 and https://tools.ietf.org/html/rfc3261#section-8.1.2 Says that I should use Route header for reaching destination But I was surprised second time when tested this scenario with FreeSwitch and another softphone (as UA) because of it both sends ACK to the based on Contact address and ignores Route I just wanna ask if I missed some scenario in the RFC when it is described to ignore Route header for the UA (I know that I use kamailio on my case as proxy server but should understand finally who should make changes with packet handling) Thanks one more time for the resonses and sorry one more time for the goal of this question that belongs to the kamailio just partially

2 24

Is it possible to play earlmedia with kamailio
by vinod mn 02 Jul '18

02 Jul '18

Hi, I am trying to configure kamailio to play early media and i am using rtpproxy_stream2uac(), but the audio file plays only after the call is connected. How to use rtpproxy_stream2uac() to play only the early media. Thank you vinod.M.N

1 1

Kamailio installation problem
by Dipayan Sanyal 01 Jul '18

01 Jul '18

Hi, I have installed kamailio in my system but while starting I am geeting following errors- Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: db_mysql [km_my_con.c:129]: db_mysql_new_connection(): driver error: Access denied for user 'kamailio'@'localhost' (using password: YES) Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [db.c:318]: db_do_init2(): could not add connection to the pool Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: usrloc [dlist.c:644]: register_udomain(): failed to open database connection Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: registrar [registrar.c:651]: domain_fixup(): failed to register domain Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:643 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/rvalue.c:3859]: fix_rval_expr(): failure in cfg at line: 643 col: 22 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/rvalue.c:3859]: fix_rval_expr(): failure in cfg at line: 643 col: 22 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: ERROR: <core> [core/route.c:1153]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:646 Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:85]: tm_shutdown(): start Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:88]: tm_shutdown(): emptying hash table Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:90]: tm_shutdown(): removing semaphores Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:92]: tm_shutdown(): destroying tmcb lists Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: tm [t_funcs.c:95]: tm_shutdown(): done Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/shm.c:260]: shm_destroy_manager(): destroying memory manager: q_malloc Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/q_malloc.c:1169]: qm_shm_lock_destroy(): destroying the shared memory lock Jul 1 00:32:42 coditas-Vostro-3558 /usr/local/kamailio-5.1/sbin/kamailio[4354]: DEBUG: <core> [core/mem/pkg.c:90]: pkg_destroy_manager(): destroying memory manager: q_malloc Please look into the problem. Thanking You, Dip

2 1

Fwd: Video call recording and mobile push notifications
by Jeremy Renner 01 Jul '18

01 Jul '18

Hi, I would like to setup up my SIP server / PBX for my business, now we have some candidates: 1. Open source solution: - Asterisk PBX, - Freeswitch PBX - Kamailio - OpenSIPS 2. Business solution: - Brekeke PBX(https://www.brekeke.com - Vodia PBX(https://www.vodia.com) - 3CX PBX(https://www.3cx.com) - PortSIP PBX(https://www.portsip.com/portsip-pbx) *Below features are mandatory for our project:* - Video call recording (For the finance industry, the video recording is necessary) - Push notifications for mobile app - Multi-tenant support - Both Linux and Windows support (at 1st stage, we would like to run it on Windows server and migrate it to Linux server in the future if users increased), the Linux support is required, the Windows support is preferred. We have some questions: 1. Does the the Kamailio can works as a PBX ? 2. If yes, does the Kamailio support push notifications and video recording ? 3. Does the Kamailio can works for Multi-tenant ? 4. Does Kamailio support Windows ? So far according to our research, with the business solution: - The Vodia PBX, PortSIP PBX and brrekeke all are support Multi-tenant, the 3CX is not. - The 3CX and PortSIP support push notifications, - The PortSIP also provide client SDK, with 3CX we only see the 3CX provide client apps, does 3CX has client SDK provided ? - It's seems all these PBX are support video recording ? - The PortSIP PBX and 3CX both support Linux. Please help me to make the decision, base on your experiences, which one (open source or business solution) is good to us ? I'm really new to VoIP... Thanks in advance. Best regards,

1 0

linphone & presence_xml ERROR
by Pali Rohár 01 Jul '18

01 Jul '18

Hi! I'm using linphone 3.6.1 as SIP client and Kamailio 4.4.4 as proxy server and registrar. And every time I try to change online status (presence) in linphone client, I see following error messages in kamailio server log: ERROR: presence_xml [add_events.c:167]: xml_publ_handl(): bad body format ERROR: presence [publish.c:443]: handle_publish(): in event specific publish handling ERROR: tm [t_reply.c:533]: _reply_light(): ERROR: _reply_light: can't generate 500 reply when a final 415 was sent out ERROR: sl [sl.c:269]: send_reply(): failed to reply stateful (tm) ERROR: presence [utils_func.c:146]: send_error_reply(): sending 500 Server Internal Error reply ERROR: presence [publish.c:492]: handle_publish(): failed to send error reply So it looks like kamailio is not able to parse presence change which is linphone doing. I captured traffic via tcpdump what linphone sends: 21:58:58.704792 IP (tos 0x68, ttl 64, id 64109, offset 0, flags [DF], proto UDP (17), length 32) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP 21:58:58.704919 IP (tos 0x68, ttl 64, id 64110, offset 0, flags [DF], proto UDP (17), length 752) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP, length: 724 PUBLISH sip:USER@HOST SIP/2.0 Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport;branch=z9hG4bK2049418743 From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST> Call-ID: 1435474953 CSeq: 26 PUBLISH Content-Type: application/pidf+xml Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Expires: 600 Event: presence Content-Length: 353 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="User <sip:USER@HOST>"> <tuple id="sg89ae"> <status><basic>closed</basic></status> <contact priority="0.8">User <sip:USER@HOST></contact> </tuple> </presence> 21:58:58.710097 IP (tos 0x10, ttl 53, id 10255, offset 0, flags [none], proto UDP (17), length 473) KAMAILIO_IP_ADDRESS.5060 > LINPHONE_IP_ADDRESS.5060: SIP, length: 445 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport=5060;branch=z9hG4bK2049418743;received=EXTERNAL_IP_ADDRESS From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST>;tag=515cb47c4c34ffa85b598d6b25676122.744d Call-ID: 1435474953 CSeq: 26 PUBLISH Proxy-Authenticate: Digest realm="HOST", nonce="NONCE" Server: kamailio (4.4.4 (x86_64/linux)) Content-Length: 0 21:58:58.718147 IP (tos 0x68, ttl 64, id 64111, offset 0, flags [DF], proto UDP (17), length 940) LINPHONE_IP_ADDRESS.5060 > KAMAILIO_IP_ADDRESS.5060: SIP, length: 912 PUBLISH sip:USER@HOST SIP/2.0 Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport;branch=z9hG4bK1165141043 From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST> Call-ID: 1435474953 CSeq: 27 PUBLISH Proxy-Authorization: Digest username="USER", realm="HOST", nonce="NONCE", uri="sip:USER@HOST", response="RESP", algorithm=MD5 Content-Type: application/pidf+xml Max-Forwards: 70 User-Agent: Linphone/3.6.1 (eXosip2/4.1.0) Expires: 600 Event: presence Content-Length: 353 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="User <sip:USER@HOST>"> <tuple id="sg89ae"> <status><basic>closed</basic></status> <contact priority="0.8">User <sip:USER@HOST></contact> </tuple> </presence> 21:58:58.781668 IP (tos 0x10, ttl 53, id 10259, offset 0, flags [none], proto UDP (17), length 380) KAMAILIO_IP_ADDRESS.5060 > LINPHONE_IP_ADDRESS.5060: SIP, length: 352 SIP/2.0 415 Unsupported media type Via: SIP/2.0/UDP LINPHONE_IP_ADDRESS:5060;rport=5060;branch=z9hG4bK1165141043;received=EXTERNAL_IP_ADDRESS From: User <sip:USER@HOST>;tag=184282924 To: User <sip:USER@HOST>;tag=97d8e785fdf42bf9622a64c13c504961-3901 Call-ID: 1435474953 CSeq: 27 PUBLISH Server: kamailio (4.4.4 (x86_64/linux)) Content-Length: 0 I replaced ip addresses in packets by KAMAILIO_IP_ADDRESS, LINPHONE_IP_ADDRESS and EXTERNAL_IP_ADDRESS strings and also SIP account by USER@HOST. Maybe it helps you. Any idea why kamailio refuse presence update and reports those error into error log? Or is there any special setting which is needed for linphone or other SIP clients for online status / presence support? -- Pali Rohár pali.rohar(a)gmail.com

2 6

dovecot authentication
by Pali Rohár 30 Jun '18

30 Jun '18

Hi! Is there any module or configuration how to use dovecot SASL authentication in Kamailio? Dovecot exports unix socket (or tcp on localhost) for authentication service and lot of server applications can be configured to use it (e.g. postfix, prosody, ...) to have one central place of user and authentication database. It supports also strong authentication schemas like SCRAM-SHA, but also plain text password or DIGEST-MD5. In SIP is used DIGEST-MD5 therefore it could be suitable also for SIP and Kamailio. -- Pali Rohár pali.rohar(a)gmail.com

2 1

Understanding 5XX stats in Kamailio
by Joel Serrano 29 Jun '18

29 Jun '18

Hi, Does anyone know the difference between these metrics?? core:rcv_replies_5xx sl:5xx_replies tmx:5xx_transactions Specially between core and tmx? Thanks, Joel.

2 1

Relevant development links
by Amauri Oliveira 28 Jun '18

28 Jun '18

Hello. I'm new to Kamailio perhaps I've seen it in actions at AirBNB. I'm starting the journey as a new developer who wants to grab Kamailio and build something. I would like to check if you could provide the most relevant links for a beginner also with some samples. I noticed the abundance of modules perhaps I would like to see what people are doing with it. Best Regards, Amauri.

2 1

Fwd: Understanding is_first_hop() and how it handles in-dialog replies
by Joel Serrano 28 Jun '18

28 Jun '18

Hi guys, I'm having an issue which I have narrowed down to is_first_hop(), I can apply a workaround, but I don't know if I'm doing it correctly or if my problem is caused by misconfiguration of anything. Let's say we have the following flow: Client (NAT) TLS -> Kamailio1 (only Public IP) UDP -> Kamailio2 (only Public IP) UDP -> FreeSWITCH (only Public IP). Regarding this setup, let's focus only on Kamailio1. And for the sake of a clear example, I have done a little draw.io diagram: >From the above flow, let's stick to the *200 OK* that Kam1 is going to receive from Kam2. (marked in *blue* in the screenshot). First 200 OK (replying to initial INVITE): SIP/2.0 200 OK Via: SIP/2.0/UDP KAM1_PUB_IP;rport=5060;branch=z9hG4bK1e2b. e1307b519c9b5f0015343e13f35aeace.0;i=3 Via: SIP/2.0/TLS [2607:fb90:489b:6e13:85f8:596b:b86b:c831]:54744; received=172.58.17.149;branch=z9hG4bK.jmXSnh-7x;rpo =43842 Record-Route: <sip:KAM2_PUB_IP;lr=on;ftag=7vidaJ3Hw;did=36e.d441> Record-Route: <sip:KAM1_PUB_IP;r2=on;lr=on;ftag=7vidaJ3Hw;did=36e.a792; nat=yes> Record-Route: <sip:KAM1_PUB_FQDN:443;transport=tls;r2=on;lr=on; ftag=7vidaJ3Hw;did=36e.a792;nat=yes> From: "Joel Test 1" <sip:8bd2a0aba14541789bb7269800646458@MY_DOMAIN >;tag=7vidaJ3Hw To: "Joel Test 2" <sip:e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;tag= vceg6N0m5ypHa Call-ID: 3ezoQGF1kp CSeq: 21 INVITE Contact: <sip:e78f2617b0d345d3bdb7b6780ece903c@FS_PUB_IP:6061;transport=udp> User-Agent: TP MEDIA 2.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 358 Remote-Party-ID: "e78f2617b0d345d3bdb7b6780ece903c" <sip: e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;party=calling; privacy=off;screen=no v=0 o=TP 1529576021 1529576022 IN IP4 FS_PUB_IP s=TP c=IN IP4 FS_PUB_IP t=0 0 m=audio 25484 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1; maxplaybackrate=8000; sprop-maxcapturerate=8000 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:25485 IN IP4 FS_PUB_IP Second 200 OK (replying to in-dialog INVITE with updated SDP): SIP/2.0 200 OK Via: SIP/2.0/UDP KAM1_PUB_IP;rport=5060;branch=z9hG4bKed2b. 0006a6a159e800129a62b4415fdd64e6.0;i=7 Via: SIP/2.0/TLS 192.168.30.63:54752;received=A.B.C.D;branch=z9hG4bK. iKO2iYIgK;rport=27819 From: "Joel Test 1" <sip:8bd2a0aba14541789bb7269800646458@MY_DOMAIN >;tag=7vidaJ3Hw To: "Joel Test 2" <sip:e78f2617b0d345d3bdb7b6780ece903c@MY_DOMAIN>;tag= vceg6N0m5ypHa Call-ID: 3ezoQGF1kp CSeq: 22 INVITE Contact: <sip:e78f2617b0d345d3bdb7b6780ece903c@FS_PUB_IP:6061;transport=udp> User-Agent: TP MEDIA 2.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 358 v=0 o=TP 1529576021 1529576022 IN IP4 FS_PUB_IP s=TP c=IN IP4 FS_PUB_IP t=0 0 m=audio 25484 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1; maxplaybackrate=8000; sprop-maxcapturerate=8000 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=rtcp:25485 IN IP4 FS_PUB_IP Now here comes the problem, I have the following in my kam1 config: route[NATMANAGE] { ... if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { fix_nated_contact(); } } } ... } So, on the first 200 OK, when it reaches that part of the config: 1- is_reply() -> OK 2- isbflagset(FLB_NATB) -> OK (because on the initial request NAT was detected blablabla....) 3- is_first_hop() -> FAIL fix_nated_contact() is NOT applied. (This is the correct and the expected behavior). Now, on the second 200 OK, again in that part of the config: 1- is_reply() -> OK 2- isbflagset(FLB_NATB) -> OK 3-is_first_hop() -> OK fix_nated_contact() is applied, thus the contact will be changed, and the client will send the ACK with incorrect information leading to another set of issues... >From the docs: https://www.kamailio.org/docs/modules/5.1.x/ modules/siputils.html#siputils.f.is_first_hop *4.30. is_first_hop()* *The function returns true if the proxy is first hop after the original sender. For incoming SIP requests, it means there is only one Via header. For incoming SIP replies, it means that top Record-Route URI is 'myself' and source address is not matching it (to avoid detecting in case of local loops). Note that it does not detect spirals, which can have the condition for replies true also in the case of additional SIP reply receival.* So going back to the examples: first 200 OK: 1- "top Record-Route URI is 'myself' -> FAIL So we we are NOT the first hop, we do nothing and forward the reply to the client. second 200 OK: 1- "top Record-Route URI is 'myself' -> No record-route headers are present, so we enter the is_first_hop() condition and modify the contact with fix_nated_contact(). Now to the real topic, I have a workaround as: ... if (is_reply()) { if (isbflagset(FLB_NATB)) { if (is_first_hop()) { if (!ds_is_from_list()) { # <-- Check to see if the reply is coming from our internal servers fix_nated_contact(); } } } } ... And that is working correctly, but I would like to understand the reasons. I hope I have explained myself correctly, otherwise it's impossible to get to the point of my questions: 1- Is it correct for is_first_hop() to detect the second 200 OK as a first hop when it isn't? the behavior matches the documentation, so I don't know. If we stick to the check of the headers etc etc, it's working as described, if we stick to the concept of kam1 being actually being the first hop of that 200OK, then the check would need another condition to exclude the 200 OK (or better, the replies) of an in-dialog INVITE. 2- Do you guys consider the workaround something reasonable? Under my opinion I would like to not have to add that, but I also don't know. 3- Am I missing something super standard that avoids all of this? I'm starting to go crazy trying and comparing different things to get to understand this and I want to make sure it's not just a is_first_hop() bug? Sorry for such a long email, but I think that describing the scenario and flow was required. Any input is more than welcome! Thanks! Joel.

1 0

Question on the new Kamailio v5.x Releases
by Ginhoux, Patrick 27 Jun '18

27 Jun '18

Hi, I always read your email regarding new stable releases. I am currently working on a project upgrade of multiples servers to the most recent OS (Windows, RedHat) and middleware. In this project, I have upgraded SIP Proxies server running on old version of Kamailio, to the 5.0.1 release (over RedHat 7.4). Effort to make that upgrade and tests have been done with success in the last months. So considering to upgrade to the 5.0.7 or 5.1.4 would be a subsequent work I need to evaluate. I could read all the release notes, but it is also a big challenge to determine if fixes are interesting for me. Based on your experience, is there particular cases, fixed in the new release I need to pay attention? Any comments are welcome. Regards Patrick GINHOUX

2 1

Compile error on 4.4.7
by Andrew Chen 27 Jun '18

27 Jun '18

Hi all, I'm getting the following error when compiling 4.4.7 from src: mem/pkg.o: In function `pkg_init_manager': /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:63: undefined reference to `fm_malloc_init_pkg_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:68: undefined reference to `qm_malloc_init_pkg_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/pkg.c:72: undefined reference to `tlsf_malloc_init_pkg_manager' mem/shm.o: In function `shm_init_manager': /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:239: undefined reference to `fm_malloc_init_shm_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:244: undefined reference to `qm_malloc_init_shm_manager' /home/prod_fuze/jenkins/workspace/BuildKamailioRELEASE_Trusty/mem/shm.c:248: undefined reference to `tlsf_malloc_init_shm_manager' collect2: error: ld returned 1 exit status make: *** [kamailio] Error 1 Build step 'Execute shell' marked build as failure Sending e-mails to: kivanov(a)fuze.com AChen(a)fuzebox.com Finished: FAILURE any thoughts? Thanks -- Andy Chen -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 3

Planning next IRC devel meeting
by Daniel-Constantin Mierla 26 Jun '18

26 Jun '18

Hello, I thought of start planning the next IRC devel meeting, it is quite some time since the last one with a busy spring for most of us. Among the topics to be approached would be the roadmap to the next major release, then discuss what else we can do for or with the project. My proposal as a date is Tuesday, June 27, at 14:00 UTC -- I created a page for it: - https://www.kamailio.org/wiki/devel/irc-meetings/2018a List yourself if you can participate or you prefer a different date, as well as add new topics that you consider they should be approached. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 3

Siremis Log Information
by Waqar Hameed 26 Jun '18

26 Jun '18

Hi I am one of your user using siremis web interface for manipulating kamailio database. I just want to know if it is possible to disable auto siremis logs present in the directory siremis-4.2.0/siremis/log/debug.log. The file debug.log contains database queries along with user ids and password and I just want to stop that logs for security concerns. If it is possible please reply with some solution of it. Thanks in advance. Regards, Waqar Hameed

2 1

invalid char found in the new uri
by Kjeld Flarup 25 Jun '18

25 Jun '18

I have a system where the calls is not disconnected properly. I get this error in syslog when receiving BYE: Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:23:54 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:645]: restore_uri(): invalid char found in the new uri at pos 30 (#012) [sip:004540294149@127.0.0.1:507#01257441;transport=TCP;ob] Jun 24 08:24:11 scantronpbx /usr/local/sbin/kamailio[31588]: WARNING: uac [replace.c:647]: restore_uri(): this can happen when URI values are altered by end points - skipping the update Kamailio receives this From header: f: <sip:004540294149@194.255.22.44>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob And passes this on, where it is obvious that '%' may give a problem f: <sip:0001@192.168.2%:>;tag=e042cfe1-40f7-410d-95fd-e5bb4f8976d7;transport=TCP;ob The RR headers send forward with INVITE looks like this: Record-Route: <sip:127.0.0.1:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> Record-Route: <sip:192.168.2.9:5069;r2=on;lr=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> And received on BYE Route: <sip:127.0.0.1:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> Route: <sip:192.168.2.9:5069;lr;r2=on;ftag=MYEMZIUn-UWwH-R7g9CktpyvOn1ZFuZu;vsf=AAAAAEUAcAB5AxcDGAkYCgALLjEwMQ--;did=2f6.6cc;vst=AAAAAAAABAR0AQsLGgACAW4DHA4uMC4wLjE6NTA3MA--> RR does not seem to be the problem as mentioned here: https://opensips.org/pipermail/users/2010-October/014999.html I have attached a pcap file with the full trace. The error appears in frame 101 The setup is a bit complex, as I have 8 Kamailio's running, where 6 is identical. The first Kamailio is on port 5069, which receives a call from a door phone. It then branches the call to registrations or GSM targets, which is handled by the other Kamailio's Thus 5069 is only doing a lookup in a database and branching, plus it handles rtpengine. I first saw the error on version 4.4, and now I replicated it on 5.1.4. My hardware is a Raspberry Pi. /usr/local/sbin/kamailio -v version: kamailio 5.1.4 (arm6/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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-NOSMP, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 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 23:45:36 Jun 19 2018 with gcc 4.9.2 -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

[topos] no local address - do record routing for all initial requests
by Abdoul Osséni 25 Jun '18

25 Jun '18

Hello all, I use topos with topos_redis modules. For each call, I got the following messages: Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11867]: {1 20 ACK FKgR7X~stk x.x.x.x 24860}WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests Jun 25 09:57:37 sd-110402 /usr/local/sbin/kamailio[11839]: {1 102 BYE FKgR7X~stk x.x.x.x xxxx}WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests It seems, there is no issue on the calls but I want to understand why I got these warning messages. Thank you. Abdoul

2 1

Kamailio installation
by Danny Roberts 23 Jun '18

23 Jun '18

Good Day, I have been task by my company to install Kamailio, can I get some assistance in installing the application? Also is there a GUI for easy install? DANNY ROBERTS | Solutions Technician | dtroberts(a)clearharbor.com<mailto:dtroberts@clearharbor.com> Frequente Industrial Park | P.O.BOX 1499 | St. George's, Grenada o 678-566-3212 x70778 m 473-417-5676 | www.clearharbor.com [cid:image004.png@01D3F36A.B17ABBD0] [cid:image006.png@01D3F36A.B17ABBD0] [cid:image003.png@01D406F0.8843CF10] [cid:image010.png@01D3F36A.B17ABBD0]

7 19

ims_usrloc_pcscf module hashing_type parameter does not work
by tyd 22 Jun '18

22 Jun '18

Dear sir, I'm working on Kamailio 4.4.0 and 5.1.4. When registering to Kamailio IMS using the same ip and port for 30 user (different contact user part), the pcscf get the same hash number and aor value. The hashing_type = 0 seems not to be work. Can anyone help me?

1 0

kamailio 5.1.4 for centos 7 repo
by Aidar Kamalov 22 Jun '18

22 Jun '18

Hello, http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… In file information block you can see file size is 5643596 bytes. But if you download it - file size is 5642888. And it is not possible to install kamailio 5.1.4 on centos7 from repo. -- Aydar A. Kamalov

2 2

Difference between allow trusted and allow address
by Wilkins, Steve 21 Jun '18

21 Jun '18

Hi All, Can someone explain the difference between allow trusted and allow address. Thank you!

2 1

Newbie: TOPOH in SBC/Proxy scenario regarding To-header
by Kevin Olbrich 21 Jun '18

21 Jun '18

Hi! I set up a proxy in front of an asterisk server to handle registration and similar. This setup needs topology hiding. I enabled and configured TOPOH which is working well. The only header which leaks network details is "To". It lists the proxys internal IP. Sure, a proxy should not alter the To and I am investigating if this can be changed in asterisk but is it even possible? Kind regards Kevin

3 2

Kamailio + FreeSwitch + WebRTC
by Emanuel Gianico 21 Jun '18

21 Jun '18

Hi everybody, I followed this tutorial https://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc#kamail… And it works fantastic! The next step was to add WebRTC support, so I added WebSockets module to enable web clients to register on kamailio. It works flawlessly and webphone clients register OK! (Followed this http://nil.uniza.sk/sip/kamailio/configuring-kamailio-4x-websocket ) Now, when I call from a softphone (eyeBeam) to the web client (jssip) the other party reach okay, rings okay, but when I pickup de call (from the web client), the softphone goes directly to the VoiceMail. >From the logs I see the jssip throw this error: "Failed to set remote offer sdp: Called with SDP without DTLS fingerprint." I would like to avoid RTPEngine, because from what I understand, FreeSwitch can handle the media part. Can somebody please have the kindness to guide me on how to enable webrtc between Kamailio and FreeSwitch? If somebody needs to see the "kamailio.cfg" please let me know, and i would upload the file to a gist. Cheers, Emanuel.

5 10

Kamailio v5.0.7 Released
by Daniel-Constantin Mierla 21 Jun '18

21 Jun '18

Hello, Kamailio SIP Server v5.0.7 stable release is out. This is a maintenance release of the previous stable branch, 5.0, that includes fixes since the release of v5.0.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.7 or v5.1.x. For more details about version 5.0.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/06/kamailio-v5-0-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.1, at this moment with its latest release v5.1.4. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com

1 0

can rtpproxy work with webrtc?
by Narasimhaiah, Vinod Maralenahalli 21 Jun '18

21 Jun '18

Hi, I configured the kamailio with rtpengine and the calls betwwn webrtc-clients and sip-clients working fine, Does rtpproxy supports the webrtc, and is there any additional configuration is required. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

1 0

Releasing v5.0.7
by Daniel-Constantin Mierla 21 Jun '18

21 Jun '18

Hello, I am considering to release Kamailio v5.0.7, out of git branch 5.0, sometime next week, likely to happen on Thursday, June 21. As usual, if there is any issue not reported yet on github, do it a soon as possible to have a chance to investigate. Soon I will review and backport the commits that apply from branch 5.1 back to 5.0, so keep an eye and notify us if a patch you expect to be in 5.0 is not backported. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

1 1

Kamailio 5.1.2 and Module PDB: pdb_query returns always 0
by Stefan Rüetschli 20 Jun '18

20 Jun '18

Hi I have here Kamailio 5.1.2 with PDB Module. I use the documented function from PDB-Module as follows: pdb_query("$rU", "$avp(i:82)"); xlog("L_INFO","PDB result $rU: $avp(i:82)\n"); The problem is, that $avp(i:82) is always 0. If I test the PDB-Server, with "pdbt -r localhost:5574 query <phonenumber>" I get a valid response. I use this already with Kamailio 4.3.6 and there it is working. https://kamailio.org/docs/modules/5.1.x/modules/pdb.html Any ideas? Thanks & regards Stefan

2 8

New blog article: Kamailio as an SBC - five years on
by Alex Balashov 20 Jun '18

20 Jun '18

This is an update to my original 2013 article on this topic: http://www.evaristesys.com/blog/kamailio-as-an-sbc-five-years-on/ -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 6

Kamailio IMS PCSCF database problem
by tyd 20 Jun '18

20 Jun '18

Dear sir, I'm using Kamailio 4.4.0 and 5.1.4 IMS module. During using SIPp 3000 subscriber register testing, after enabling PCSCF DB URL function, the contents of 3000 subscribers on PCSCF location table are the same. Using the same IP address and UDP port of 3000 SIPp REGISTER message seems to be the problem. Anyone can help me? Thanks a lot.

1 0

KEMI routing script Python exceptions with empty traceback
by Michael Loughrey 19 Jun '18

19 Jun '18

Hi All, I'm using app_python with a KEMI Python routing script. I have been getting occasional Python exception error logs : kamailio[2242]: 8(2253) ERROR: app_python [python_support.c:150]: python_handle_exception(): python_exec2: Unhandled exception in the Python code: kamailio[2242]: TypeError: an integer is required The scripts seems to function without an issue but these error logs appear fairly frequently, always with an empty Python stack trace. I am running Kamailio 5.1 on a Debian 9 system and using Python 2.7 : >>:/usr/local/etc/kamailio# python --version Python 2.7.13 >>:/usr/local/etc/kamailio# kamailio -v version: kamailio 5.1.4 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, 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 18:56:43 Jun 8 2018 with gcc 6.3.0 I have also tried using a Kamailio load built from the master branch (commit f8b8cecb) which gives enhanced exception logging identifying the function an exception occurs within and parameters. I see this occurring in different parts of the routing script e.g. from script def ksr_onsend_route(self,msg): KSR.xlog.xdbg("ONSEND_ROUTE fs=[$fs] $mb\n"); return 1; an exception is reported within ksr_onsend_route() even when the log level is info. Changing log to xlog.xinfo, the log is seen to be sent to syslog but again the exception is reported. These exceptions are always "TypeError: an integer is required" but don't seem related to the Python script. Here is an example of the log with a patch applied to to Kamailio 5.1 to show the function the exception is reported within. kamailio[14092]: 6(14103) ERROR: app_python [python_support.c:156]: python_handle_exception(): apy_exec: ksr_reply_route((null)): Unhandled exception in the Python code: kamailio[14092]: TypeError: an integer is required . . . kamailio[14092]: 22(14119) INFO: <script>: ONSEND_ROUTE fs=[udp: 10.0.31.126:5080] ACK sip:10008@media.test.io:5061;transport=TLS SIP/2.0 . . . . kamailio[14092]: 22(14119) ERROR: app_python [python_support.c:156]: python_handle_exception(): apy_exec: ksr_onsend_route((null)): Unhandled exception in the Python code: kamailio[14092]: TypeError: an integer is required Has anyone got an idea what can be causing these exceptions and how to avoid them? Currently I have simply patched the load to drop the level of this log to debug when such an exception occurs with an empty traceback as they seem spurious. Cheers Mike

2 3

Installing RTPengine on CentOS 7
by Francisco Valentin Vinagrero 19 Jun '18

19 Jun '18

Hi, I'm trying to install RTPengine in the same machine as I have Kamailio, running CentOS 7, but there are a few dependencies that yum cannot seem to resolve and I that cannot find anywhere else. As soon as I try to build the daemon it produces these errors: make[1]: Leaving directory `/usr/local/src/rtpengine/daemon' Failed build with ../lib/fix_frame_channel_layout-01.h Trying build with ../lib/fix_frame_channel_layout-02.h make[1]: Entering directory `/usr/local/src/rtpengine/daemon' rm -f "resample.c" echo '/******** GENERATED FILE ********/' > "resample.c" cat ../lib/"resample.c" >> "resample.c" Package libevent_pthreads was not found in the pkg-config search path. Perhaps you should add the directory containing `libevent_pthreads.pc' to the PKG_CONFIG_PATH environment variable No package 'libevent_pthreads' found Package libavcodec was not found in the pkg-config search path. Perhaps you should add the directory containing `libavcodec.pc' to the PKG_CONFIG_PATH environment variable No package 'libavcodec' found Package libavformat was not found in the pkg-config search path. Perhaps you should add the directory containing `libavformat.pc' to the PKG_CONFIG_PATH environment variable No package 'libavformat' found Package libavutil was not found in the pkg-config search path. Perhaps you should add the directory containing `libavutil.pc' to the PKG_CONFIG_PATH environment variable No package 'libavutil' found Package libswresample was not found in the pkg-config search path. Perhaps you should add the directory containing `libswresample.pc' to the PKG_CONFIG_PATH environment variable No package 'libswresample' found Package libavfilter was not found in the pkg-config search path. Perhaps you should add the directory containing `libavfilter.pc' to the PKG_CONFIG_PATH environment variable No package 'libavfilter' found So, in the end, at least 'libevent_pthreads','libavcodec', 'libavformat', 'libavutil', 'libswresample' and 'libavfilter' are missing. Trying to install them one by one with yum fails because it cannot find these packages for CentOs 7. Can anyone point me to a solution for this? Thanks, Francisco.

3 2

Re: [SR-Users] Textops Regex
by Duarte Rocha 19 Jun '18

19 Jun '18

You're right, it is application/sdp and the back slash works as escape character. However, a new problem came. I'm using this, since someone upstream is adding \r\n to a part of the body : replace_body_all("application\/sdp\r\n\r\n", "application/sdp\r\n"); The regex finds the application/sdp just right, but doesn't replace everything. So, as a final result i get applicationapplication\sdp\r\n. Is this intended? Thanks for your help

1 0

centos 7 repo is broken?
by Aidar Kamalov 19 Jun '18

19 Jun '18

Hello, can't install kamailio from repo (repo was update 16 jun), error: kamailio-5.1.4-1.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C…: [Errno 14] HTTP Error 416 - Requested Range Not Satisfiable/s | 0 B --:--:-- ETA -- Aydar A. Kamalov

3 4

SIP Notify use UDP instead TCP
by denislav.tsonev＠musala.com 19 Jun '18

19 Jun '18

Hello, I'm trying to configure Kamailio 5.1 like "presence" server. Pretty much with default settings of kamailio: # ----- presence params ----- modparam("presence", "db_url", DBURL) ;modparam("presence", "server_address", "sip:10.0.6.226:5060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) And when UDP transport is using, everything works perfect. But when TCP is using (only "listen: tcp:IP:5060" is set) NOTIFY is not send. When "listen: tcp:IP:5060" and "listen: udp:IP:5060" I receive NOTIFY but through UDP. SIP trace is bellow: No. Time Source Destination Protocol Length Info 715 2018-06-18 17:05:13,164685 10.0.8.249 10.0.6.226 SIP 585 Request: SUBSCRIBE sip:71010@10.0.6.226:5060;transport=tcp | Frame 715: 585 bytes on wire (4680 bits), 585 bytes captured (4680 bits) on interface 0 Ethernet II, Src: LcfcHefe_86:92:0a (28:d2:44:86:92:0a), Dst: 8e:5c:a8:56:db:cb (8e:5c:a8:56:db:cb) Internet Protocol Version 4, Src: 10.0.8.249, Dst: 10.0.6.226 Transmission Control Protocol, Src Port: 50636, Dst Port: 5060, Seq: 1, Ack: 1, Len: 531 Session Initiation Protocol (SUBSCRIBE) Request-Line: SUBSCRIBE sip:71010@10.0.6.226:5060;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP 10.0.8.249:5060;transport=tcp;branch=z9hG4bK-10772-1-0 From: <sip:61010@10.0.8.249>;transport=tcp;tag=1 To: <sip:71010@10.0.6.226:5060> Call-ID: 1-10772(a)10.0.8.249 CSeq: 1 SUBSCRIBE Contact: sip:61010@10.0.8.249:5060;transport=tcp Max-Forwards: 10 Event: dialog Expires: 7200 [Expert Info (Warning/Malformed): Header has no colon after the name] User-Agent: SIPp/Win32 Accept: application/dialog-info+xml, multipart/related, application/rlmi+xml Content-Length: 0 No. Time Source Destination Protocol Length Info 870 2018-06-18 17:05:13,259323 10.0.6.226 10.0.8.249 SIP 439 Status: 202 OK | Frame 870: 439 bytes on wire (3512 bits), 439 bytes captured (3512 bits) on interface 0 Ethernet II, Src: 8e:5c:a8:56:db:cb (8e:5c:a8:56:db:cb), Dst: LcfcHefe_86:92:0a (28:d2:44:86:92:0a) Internet Protocol Version 4, Src: 10.0.6.226, Dst: 10.0.8.249 Transmission Control Protocol, Src Port: 5060, Dst Port: 50636, Seq: 1, Ack: 532, Len: 385 Session Initiation Protocol (202) Status-Line: SIP/2.0 202 OK Message Header Via: SIP/2.0/TCP 10.0.8.249:5060;transport=tcp;branch=z9hG4bK-10772-1-0 From: <sip:61010@10.0.8.249>;transport=tcp;tag=1 To: <sip:71010@10.0.6.226:5060>;tag=b3c72a85f6f978c1e325d5994421001b-970f Call-ID: 1-10772(a)10.0.8.249 CSeq: 1 SUBSCRIBE Expires: 3600 Contact: <sip:10.0.6.226:5060;transport=tcp> Server: kamailio (5.1.3 (x86_64/linux)) Content-Length: 0 No. Time Source Destination Protocol Length Info 871 2018-06-18 17:05:13,259815 10.0.8.249 10.0.6.226 SIP 337 Status: 200 OK | Frame 871: 337 bytes on wire (2696 bits), 337 bytes captured (2696 bits) on interface 0 Ethernet II, Src: LcfcHefe_86:92:0a (28:d2:44:86:92:0a), Dst: 8e:5c:a8:56:db:cb (8e:5c:a8:56:db:cb) Internet Protocol Version 4, Src: 10.0.8.249, Dst: 10.0.6.226 Transmission Control Protocol, Src Port: 50636, Dst Port: 5060, Seq: 532, Ack: 386, Len: 283 Session Initiation Protocol (200) Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/TCP 10.0.8.249:5060;transport=tcp;branch=z9hG4bK-10772-1-0 From: <sip:61010@10.0.8.249>;transport=tcp;tag=1 To: <sip:71010@10.0.6.226:5060>;tag=b3c72a85f6f978c1e325d5994421001b-970f Call-ID: 1-10772(a)10.0.8.249 CSeq: 1 SUBSCRIBE Content-Length: 0 No. Time Source Destination Protocol Length Info 1924 2018-06-18 17:05:19,259639 10.0.6.226 10.0.8.249 SIP 1193 Request: NOTIFY sip:61010@10.0.8.249:5060 | Frame 1924: 1193 bytes on wire (9544 bits), 1193 bytes captured (9544 bits) on interface 0 Ethernet II, Src: 8e:5c:a8:56:db:cb (8e:5c:a8:56:db:cb), Dst: LcfcHefe_86:92:0a (28:d2:44:86:92:0a) Internet Protocol Version 4, Src: 10.0.6.226, Dst: 10.0.8.249 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (NOTIFY) Request-Line: NOTIFY sip:61010@10.0.8.249:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.0.6.226;branch=z9hG4bKe05d.273c9ca6000000000000000000000000.0 To: <sip:61010@10.0.8.249>;tag=1 From: <sip:71010@10.0.6.226>;tag=b3c72a85f6f978c1e325d5994421001b-970f CSeq: 2 NOTIFY Call-ID: 1-10772(a)10.0.8.249 Content-Length: 644 User-Agent: kamailio (5.1.3 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.0.6.226:5060;transport=tcp> Subscription-State: active;expires=3594 Content-Type: application/dialog-info+xml Message Body No. Time Source Destination Protocol Length Info 1989 2018-06-18 17:05:19,705774 10.0.6.226 10.0.8.249 SIP 1193 Request: NOTIFY sip:61010@10.0.8.249:5060 | Frame 1989: 1193 bytes on wire (9544 bits), 1193 bytes captured (9544 bits) on interface 0 Ethernet II, Src: 8e:5c:a8:56:db:cb (8e:5c:a8:56:db:cb), Dst: LcfcHefe_86:92:0a (28:d2:44:86:92:0a) Internet Protocol Version 4, Src: 10.0.6.226, Dst: 10.0.8.249 User Datagram Protocol, Src Port: 5060, Dst Port: 5060 Session Initiation Protocol (NOTIFY) Request-Line: NOTIFY sip:61010@10.0.8.249:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.0.6.226;branch=z9hG4bKe05d.273c9ca6000000000000000000000000.0 To: <sip:61010@10.0.8.249>;tag=1 From: <sip:71010@10.0.6.226>;tag=b3c72a85f6f978c1e325d5994421001b-970f CSeq: 2 NOTIFY Call-ID: 1-10772(a)10.0.8.249 Content-Length: 644 User-Agent: kamailio (5.1.3 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:10.0.6.226:5060;transport=tcp> Subscription-State: active;expires=3594 Content-Type: application/dialog-info+xml Message Body Could someone help? Thanks in advance. Best regards, Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

2 1

Timer child process loosing MySQL connections
by Tobias Lindgren 19 Jun '18

19 Jun '18

Hi all! Having an issue with MySQL db connections being dropped in a system running 4.4.7. We're using db_mysql and db_cluster modules setup a cluster connecting two different DB servers. We have two cluster connections, one for acc and one for "other queries". One DB (A) is on the same network, another DB (B) is on another network. The default DB connection is for the remote server B. Auto reconnect is enabled. The specific issue seen is that the "timer" child process looses/drops both connections to DB A and B. Looking at the output from lsof when this happens, the connections usually does not both drop connections for A and B at the same time. Sometimes the connections keep up for ~24h, sometimes for 10 minutes, but normally the problem re-occurs every 6 hours or so. We're seeing this problem on two Kamailio servers, both handling fairly high amount of calls. None of the other Kamailio child processes seems to get their connections dropped, only the "timer" process. To solve this we need to restart Kamailio. Lately I've added the timer.so module to make a simple query on each cluster connection each 10 seconds. This is an example output from when the problem appears and connections are dropped: Jun 15 09:39:12 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on 'xxx' (4) (2003) Jun 15 09:39:12 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:12 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) Jun 15 09:39:12 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on 'xxx' (4) (2003) Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on 'xxx' (4) (2003) Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Can't connect to MySQL server on 'xxx' (4) (2003) Jun 15 09:39:13 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:14 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server during query (2013) Jun 15 09:39:14 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:14 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:14 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:15 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:15 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:16 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:16 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:17 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:17 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:18 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:18 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:27 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:27 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:28 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:28 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:32 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:32 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:32 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:32 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:34 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:34 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:34 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:34 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:35 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:35 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:35 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:35 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:37 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:37 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:38 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:38 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:47 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:47 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:48 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:48 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:50 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:50 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while submitting query Jun 15 09:39:50 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:50 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Jun 15 09:39:57 /usr/sbin/kamailio[10439]: ERROR: db_mysql [km_dbase.c:128]: db_mysql_submit_query(): driver error on query: Lost connection to MySQL server at 'waiting for initial communication packet', system error: 4 (2013) Jun 15 09:39:57 /usr/sbin/kamailio[10439]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Again, it's always and only the timer child process where these error messages are seen. Have been trying to debug this for some time now but I'm running out of ideas. Do you have any ideas on what I could look for? Any idea on why it's just the timer-process? Kind regards, /Tobias

2 5

rtpproxy_stream2uac
by Narasimhaiah, Vinod Maralenahalli 18 Jun '18

18 Jun '18

Hi, I am trying to configure kamailio to play early media and i am using rtpproxy_stream2uac(), but the audio file plays only after the call is connected. How to use rtpproxy_stream2uac() to play only the early media. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

1 0

Textops Regex
by Duarte Rocha 18 Jun '18

18 Jun '18

Greetings, What kind of regex does textops module uses? I'm trying to find the string "application\sdp" but i can't the escape character for the "\". Thanks in advance

2 1

Please Deregister me
by siva chirravuri 18 Jun '18

18 Jun '18

2 1

Registration Failed
by Ashok kumar Marra 18 Jun '18

18 Jun '18

Hi all, Few days back only, I started working on Kamailio IMS. I set up PSCSF, ICSCF, SCSCF and HSS(from open hss). When I try to send Register request from Zoiper, ICSCF is giving request timeout. At SCSCF, MAR failed error is logged. Kindly, let me know where I am doing wrong? >From my observation, At HSS, I can't see MAR request processing, It is only processing UAR from ICSCF. Find the attached log file. -- With Best Regards, Ashok Kumar.

1 0

Register failed at SCSCF
by Ashok kumar Marra 18 Jun '18

18 Jun '18

Hi all, Few days back only, I started working on Kamailio IMS. I set up PSCSF, ICSCF, SCSCF and HSS(from open hss). When I try to send Register request from Zoiper, ICSCF is giving request timeout. At SCSCF, MAR failed error is logged. I am attaching the screenshot of log. Kindly, let me know where I am doing wrong? >From my observation, At HSS, I can't see MAR request processing, It is only processing UAR from ICSCF. -- With Best Regards, Ashok Kumar

1 0

TLS module and IPv6 (using tls.cfg).
by Joel Serrano 17 Jun '18

17 Jun '18

Hi all, I was playing around with the "config" modparam to specify and use a TLS configuration file instead of inline modparams inside kamailio.cfg. I did some testing and when I went to try with IPv6 I run into a little doubt?? tls.cfg config example: [server:1.2.3.4:5061] <----- How do you put an IPv6 in there? method = TLSv1 verify_certificate = yes require_certificate = no private_key = /usr/local/etc/kamailio/tls/local_key.pem certificate = /usr/local/etc/kamailio/tls/local_cert.pem verify_depth = 3 ca_list = local_ca.pem crl = local_crl.pem Applying the logic of the "listen" param, it should be: [server:[AAAA:BBBB:CCCC:DDDD:EEEE:FFFF:GGGG:HHHH]:5061] But it just seems totally wrong and I haven't found anything in the docs. Any suggestions? Thanks, Joel.

2 2

SDPC on SDP
by Duarte Rocha 17 Jun '18

17 Jun '18

Greetings, Please help me understand the reason and source of this behaviour on Kamailio. I receive a 200OK to a RE-INVITE that has : Session Name (s) : MySession Connection Information (c) : IN IP4 xxx.xxx.xxx.xxx Then, Kamailio routes the 200OK to it's destination, and it leaves Kamailio like this : Session Name (s) : MySession sdpc=IN xxx.xxx.xxx.xxx The line Connection Information got deleted and it's info is now on Session Name. How can i disable this ? Thanks

2 1

Alias listen
by kai.ohnacker＠cbc.de 17 Jun '18

17 Jun '18

Hello everybody, I have a question about the function "aliases". With my server no alias entry is registered and there is no DNS server (reverse DNS is not possible) in the network. Nevertheless, the SIP server listens to a defined alias. Is there next to the config file and the kamctlrc file another place where an alias can be entered? Cheers, Kai

2 1

Configuring Kamailio as a Proxy to another SIP trunk
by Ori Popowski 15 Jun '18

15 Jun '18

Hi, I'm making my first steps in Kamailio, and need help for a simple configuration. I am trying to configure Kamailio to forward SIP traffic from (and to) a SIP gateway to a Twilio SIP trunk: SIP gateway ----> Kamailio ----> Twilio SIP trunk How to configure Kamailio for this use case? How the kamailio.cfg file will look like in such case? Thanks.

1 0

add_uri_param with variables
by Patrick Wakano 15 Jun '18

15 Jun '18

Hello list, Hope you all doing well! We've been attempting to add a URI parameter to implement the trunk group (tgrp and trunk-context) but discovered the add_uri_param() function only works with constant string.... we can't use a pseudovar to inform the value to be added. Anyone knows why such limitation? Can it be changed? It is possible to do the same thing via the subst_uri function, but it would be so much more elegant to use the function... Thank you, Kind regards, Patrick Wakano

1 0

outbound flow tokens and kamailio restart
by Pan Christensen 14 Jun '18

14 Jun '18

Hello all. I have created a WebRTC to SIP gateway. I implemented it using the outbound module. If I restart Kamailio during a call, subsequent messages fail to be routed. {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: outbound [outbound_mod.c:261]: decode_flow_token(): flow-token failed validation {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:519]: process_outbound(): failed to decode flow token {1 322 BYE 218565972_26748892(a)x.x.x.x} INFO: rr [loose.c:794]: after_loose(): failed to process outbound flow-token Is it possible to make the flow token survive a restart? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

1 1

Re: [SR-Users] Using a var as timeout value in dlg_set_timeout_by_profile function
by Henning Westerholt 14 Jun '18

14 Jun '18

Am Donnerstag, 14. Juni 2018, 09:30:44 CEST schrieb jenus(a)cyberchaos.nl: > I do not see any errors in the logs, it looks like it just sets the > timer to 0 and disconnects the call. Hello Jan, I just looked quickly in the code, it seems that the timeout value is not interpreted as "pseudo-variable", therefore its not possible to use something like $var in this method right now. It is just read as a numeric value. dialog.c, line 1342: if(dlg_set_timeout_by_profile((struct dlg_profile_table *) profile, &val_s, atoi(timeout_str)) != 0) This can of course changed in the module with a change in the code. Best regards, Henning -- If you like the work that I do in Kamailio, please consider supporting me on Patreon: https://www.patreon.com/henningw

2 1

can't assign values to $fU
by Karthik Srinivasan 14 Jun '18

14 Jun '18

From the online documentation:$fU - From URI username *$fU* - reference to username in URI of 'From' header It is R/W variable (you can assign values to it directly in configuration file) But i am no able to assign values to $fU. Has anyone experienced this? Snippet of code in the configuration file of mine and its respective logs showing that $fU is not getting overwritten. I have tried to hard code values to it assign variables to it, all to no avail. configuration file code snippet: xlog("L_INFO", "[$ci] from header username pre mod: $fU"); $fU = "5555555555"; xlog("L_INFO", "[$ci] from header username post mod: $fU"); Logfile snippet: INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username pre mod: 3224567880 INFO: (s) [7EAB8FECD723E8444E36C659F331068B41EC6BB0] from header username post mod: 3224567880 Thanks, Karthik

4 8

Kamailio as outbound proxy for PBX
by Wilkins, Steve 14 Jun '18

14 Jun '18

Good Morning All! If a PBX(Asterisk) uses an outbound_proxy (such as Kamailio), can Kamailio actually make the SIP call? At some point I would like outbound calls to be controlled by Kamailio so that the outside endpoints never communicate with the PBX. Currently a call goes through Kamailio to Asterisk and then Asterisk communicates with the endpoint, and this is night ideal Thanks ALL

2 2

t_save_lumps() and failure_route
by Julien Chavanton 14 Jun '18

14 Jun '18

Hi, I have a scenario where I am using. t_save_lumps(), lookup() and then I can choose to drop some branches in branch routes. However, it is possible that I endup without any branch left, in this case t_relay() is returning and no failure route is called/created. At this point I need to recover the original message I believe it is not possible to restore/reset the message without passing by failure route ? Else, I could trigger a failure route using t_relay(), is there any "handy/clean" way to trigger failure_route() ? Regards

1 0

Memory leak in tm with push notifications
by Ivaylo Markov 14 Jun '18

14 Jun '18

Hello, I am trying to set up Kamailio as a push notifications proxy, closely following the example in the "Kamailio in a Mobile World" presentation (https://www.slideshare.net/FedericoCabiddu/kamailioinamobileworld-51617342). I am running Debian 9 and Kamailio 5.1.3 from the official Debian repositories. I believe the main modules involved in the issue below are tm, tmx, and tsilo. Every call passing through the proxy leads to a small memory leak in the tm module - there is a large amount of "delayed free" memory cells from tm's internal hash table. At some point the shared memory runs out and Kamailio restarts. Using the "kamcmd corex.shm_summary" command I was able to see that the top users of shared memory are "tm: h_table.c: build_cell" and "core: core/sip_msg_clone.c: sip_msg_shm_clone" with the same allocation count. I experimented with removing different parts of the configuration and noticed that commenting out the "t_continue(...)" call in the "PUSHJOIN" route (see slide #22) prevents the leak from happening. Maybe something in that function is incrementing the reference counter to the hash table cell, but it is not decrementing the counter when done? I tried looking around the source code of the tm and tmx modules, but saw nothing suspicious. I also tried using gdb with a breakpoint in t_continue_helper (tm/t_suspend.c:166) hoping to see what else is accessing the htable cell, but was unable to find anything of use. Has someone encountered anything like this? Can you provide more directions on debuggin this? I can provide some bits of configuration, but an entire test setup would be rather difficult, unfortunately. Thank you for your time, Ivo

4 9

integrate huawei HSS with kamailio
by eyas barhouk 14 Jun '18

14 Jun '18

hello dears i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS , so is there any one has an experiment with this or tested it before ??? if yes, could you please guide me how to do that ? or where to start ? thanks in advance

2 2

Forbidden - Private identity not found (Authorization: username)
by eyas barhouk 14 Jun '18

14 Jun '18

Hello dears when i was trying to build an IMS platform by using kamailio with sip digest authentication scheme and ims_auth module, i couldn't register and the result was as following : 1-the first register message passed to the s-cscf then S-cscf replaying with the message “401 unauthorized - challenging the ue” 2- the second register pass to the S-cscf with the authorization header then the S-CSCF replying with "Forbidden - Private identity not found (Authorization: username)" So based on that could any one tell me what is the reason of this error and how to fix it ?? Thanks in advance

2 5

Using a var as timeout value in dlg_set_timeout_by_profile function
by jenus＠cyberchaos.nl 13 Jun '18

13 Jun '18

I'm trying to use the dlg_set_timeout_by_profile in the xhhtp route to update the dialog timeout for active calls. But if i use a var as the timeout value the call disconnects immediately. This works fine (call disconnects after 80 sec): dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "80"); This fails (call disconnects immediately: $var(DialogTimeout)= 80; dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "$var(DialogTimeout)"); Are vars supported in the dlg_set_timeout_by_profile function? And is it possible to use it in any route as described in the docs? I'm running kamailio 5.1.3. Thanks, Jan

2 1

Registration of UACs i n a partially encrypted scenario
by mip FKF 13 Jun '18

13 Jun '18

Hello, we are pretty new to SIP and kamailio, we do have some questions regarding the following scenario: We have a number of UACs in a small network which are required to communicate without encryption because the are not able to consume certificates. We want to use kamailio (as a proxy?) to establish an encrypted connection to a backend UAS. 1. Is it possible to directly register the UACs with the UAS eventhough communication between kamailio and the UAS is encrypted ? 2. How do we need to configure kamailio in order to make this scenario work ? Thank you for your suppport

2 1

Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX
by Floimair Florian 13 Jun '18

13 Jun '18

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint (chan_pjsip). With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of "Wilkins, Steve" <swwilkins(a)mitre.org> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 12. June 2018 at 16:27 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can’t be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Asterisk inviting Kamailio
by Wilkins, Steve 13 Jun '18

13 Jun '18

Hello All, I have noticed that sometimes when a call is made from one endpoint to another through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after the call has been established. Sometimes this does not happen. When it does happen, calls drop. Why would an INVITE be sent back to Kamailio? Also, Pan, I am working on the response you requested yesterday. Thanks you All, -Steve

3 2

Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 12 Jun '18

12 Jun '18

Got it working. Thank you everyone. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 3:06 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< Alex, Pan, Daniel,... Could this group => group:BUNDLE audio video in Message Body have anything to do with my Kamailio Video issue. Thank you!! -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 12:14 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Hi Alex, No, I'm not using rtpengine. It must definitely be some sort of Codec issue though, since I can seem to move the problem around and actually finally get video on the softphone. It was a little strange that I lost Audio on the WebRTC client though, since I only disabled VP8. Since Kamailio is only relaying I am so confused why introducing Kamailio is messing up Video. Thank you, -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov Sent: Sunday, June 10, 2018 11:46 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Unless you're using rtpengine, you're just moving the problem around. Kamailio does nothing with SDP unless by way of rtpengine or told in some other way, such as sdpops. On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" <swwilkins(a)mitre.org> wrote: >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I >now get Video on the softphone however, I lost two-way Audio. Kamailio >seems to be doing something with the codecs but still can't put my >finger on it. The WebRTC Client, who is the caller, needs VP8 for >Video, and apparently Audio. > >I'm not sure if I getting closer or just moving the problem around. > >Thanks All! > >From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf >Of Wilkins, Steve >Sent: Saturday, June 9, 2018 12:47 PM >To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when >using Kamailio...works without Kamailio > >>>>Email originates from a non-MITRE system. Use caution.<<< >I am desperately trying to resolve a big issue I have when using >Kamailio 5.2.0 and Asterisk 15.3. > >As soon as I go through Kamailio, I get no Video on either side of the >call; I do get two-way audio and the call stays connected. If I simply >go right through Asterisk, I also get two-way Video. I am having a >tough time determining why Kamailio is messing with the Video portion >of the call. >I have included a full pcap file. As a side note when comparing pcap >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from >Asterisk and Softphone are exactly the same. > >Thank you, >-Steve -- Alex -- Sent via mobile, please forgive typos and brevity. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 2

Question on Contact Header on Kamailio 200 OK to PBX
by Wilkins, Steve 12 Jun '18

12 Jun '18

Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can't be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Facing Problem Registration / Invite pass through one side voice
by Ankit vasava 12 Jun '18

12 Jun '18

Hello , I have installed opensips latest version and have free switch. All user is registered with free switch through opensips like registration pass though you can find out my config file into attachments also pcap log. My free switch IP is *192.168.1.147:5060 <http://192.168.1.147:5060>* Opensip IP is : *192.168.1.12:5069 <http://192.168.1.12:5069>* I m sending registration and invite request to public IP and that public IP is bind to open sips. *Problem:* when external user who is registered through public ip when he is dial any extension number who is registered with local ip at that time we are facing one way voice. I need voice(RTP) could be transfer both the side. -- *Regards,Ankit vasava* Sr. VoIP Developer *Phone: *+91-9879491525 *Skype: *ankit.vasava Http://in.linkedin.com/in/ankitvasava

2 1

unable to Kamailio IMS 5.1.4?
by vinay kumar 12 Jun '18

12 Jun '18

Hi All, Im getting following error when i start kamailio.cfg of icscf. ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed ERROR: <core> [core/sr_module.c:990]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) can anyone help me on this?? Regards, Vinay

1 0

unable to bring up kamailio 5.1.4
by vinay kumar 12 Jun '18

12 Jun '18

Hi, Im facing following issue when I tried to bring up kamailio IMS 5.1.4. ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 354, column 47: Can't set module parameter can anyone share complete installation steps for Kamailio IMS 5.1.4 for volte call?? Regards, Vinay

1 0

Problem with TLS
by Arik Halperin 12 Jun '18

12 Jun '18

Hello, I’m using TLS After receiving 200OK from kamailio: r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> From: "number" <sips:972523391991@kamprod.telemessage.com<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com<mailto:1111111@kamprod.telemessage.com>>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: <sip:1111111@10.168.10.200:5080;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti PJSIP responds with: Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session What is the reason for this? How can I fix this issue? Thanks, Arik Halperin

3 5

Disable basic authentication?
by Skip Morse 11 Jun '18

11 Jun '18

Hi All, I couldn't get a clear answer from the documentation. Based on the description of www_authenticate and proxy_authenticate: " The function verifies credentials according to RFC2617. If the credentials are verified successfully then the function will succeed and mark the credentials as authorized (marked credentials can be later used by some other functions). If the function was unable to verify the credentials for some reason then it will fail and the script should call www_challenge which will challenge the user again." The RFC outlines basic and digest auth. Is there a way to disable the ability for a UA to use basic auth? Or maybe these is disabled already? I'm likely understanding this wrong, it seems to me that it's allowed for a UA to auth with basic even if the challenge is for digest. Any clarification would be helpful. Thanks! -Skip

2 2

SIP/2.0 603 Failed to get local SDP for sip to webrtc (SIPML5)call
by vinod mn 11 Jun '18

11 Jun '18

Hi i configured sipml5 and kamailio , sipml5 clients registers to kamailio when i make a call to kamailio call lands on the sipml5 client but i am getting this error *"SIP/2.0 603 Failed to get local SDP" *on chrome/firefox/opera webbrowsers. kamilio configuration as per http://www.kamailio.org/docs/ modules/4.4.x/modules/websocket.html#idp53411716 -- Thanks and regards Vinod.M.N

2 1

sip digest authentication
by eyas barhouk 11 Jun '18

11 Jun '18

Hello dears I’m trying to setup Kamailio with sip digest authentication scheme by using the following configuration from s-cscf side : modparam("ims_auth", "registration_default_algorithm", "HSS-Selected") and the result was as following : 1. when the register message passed form the s-cscf to the hss and in the diameter trace I found the “SIP-AUTHENTICATION-SCHEME” in the MAR request as “Digest-MD5” 1. After that the HSS send MAA message with “DIAMETER-ERROR-AUTH-SCHEME-NOT-SUPPORTED” 2. S-cscf replay with the message “Forbidden – privet identity not found (Authorization : username)” So based on that could any one tell where was my mistake or how can I setup Kamailio with SIP-DIGEST scheme Thanks

2 4

JSON RPC Error: LUMP_RPL_BODY already added!
by Asgaroth 11 Jun '18

11 Jun '18

Hi All, We are trying to query the json rpc server on our registrar for the contact of an aor, but we are seeing the following error(s) being displayed in the log when we query the location table: ERROR: <core> [core/data_lump_rpl.c:83]: add_lump_rpl2(): LUMP_RPL_BODY already added! ERROR: xhttp [xhttp_mod.c:410]: xhttp_send_reply(): Error while adding reply lump We are getting the aor details back in response from the rpc call, but the error's in the log are a little concerning and I cannot see where we are doing something wrong. As an example, the following curl request returns the correct response, but the above error is displayed in the log, Am I doing something wrong here, I cannot see what the issue is: curl --header 'Content-Type: application/json' --data-binary '{ "jsonrpc": "2.0", "method": "ul.lookup", "params": [ "location", "user(a)example.com" ], "id": 3 }' http://registrar.example.com/rpc A secondary question is, is there an upper limit on the size of the "id" parameter? This kamailio instance is v5.0.6. Any thoughts as to what this error means? Thanks

2 5

(N)DB_FIREBASE module?
by Pan Christensen 11 Jun '18

11 Jun '18

Greetings all. We are currently developing web and mobile apps using firebase: https://firebase.google.com/products/realtime-database/ . We have been discussing the possibility of Kamailio querying this database. I assume that it will be possible to query it using HTTP and JSON modules, but it would probably be easier to use the DB API. Any plans to develop such a module? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

3 3

Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 10 Jun '18

10 Jun '18

I just did a test where I disabled VP8 in Kamailio using SDPOPS and I now get Video on the softphone however, I lost two-way Audio. Kamailio seems to be doing something with the codecs but still can't put my finger on it. The WebRTC Client, who is the caller, needs VP8 for Video, and apparently Audio. I'm not sure if I getting closer or just moving the problem around. Thanks All! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Saturday, June 9, 2018 12:47 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

2 3

No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 09 Jun '18

09 Jun '18

I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

1 0

Combining lookup_branches() and t_(next|load)_contacts()
by Nathan Ward 09 Jun '18

09 Jun '18

Hi, About a year ago I posted this, but haven’t had much chance to follow up on it: https://lists.kamailio.org/pipermail/sr-users/2017-March/096337.html I’ve been doing some work on this issue (we worked around it until now), and have reduced the config right down to see where things are going wonky. The scenario I have, is inbound calls (to let’s say “A@domain") which I have aliased to two or more different numbers (let’s say “B@domain” and “C@domain”). Both B and C are registering, but are not always registered. I resolve the aliases with alias_db_lookup() (in my sample below, I use seturi() and append_branch() to avoid any confusion about where things may be going weird), then look them up in the location table with lookup_branches(). I then need to do t_load_contacts() and t_next_contacts(), as in some call scenarios I add a serial fork before or after the main call (pre-call announcements etc.). Then, I call t_relay(). This works great most of the time - however, if for example B is registered, but C is not registered, lookup_branches() notes that B is "Not found in usrloc”, and then t_load_contacts() seems to not realise that this isn’t usable, and it attempts to do a DNS lookup for the domain and so on, which I don’t really want it to be doing. Should lookup_branches() be removing a branch it can’t resolve? I would expect it to behave like that, anyway. Can I configure it to do that, or do I need to manually loop through them or something to make sure they resolved, and remove the ones that didn’t? I’ve tried swapping it around, so I call lookup_branches() after t_next_contacts() but the behaviour is the same. I imagine I can’t be the only person trying to do this - how have others made it work? Config sample: seturi('sip:a@domain'); append_branch('sip:b@domain'); xlog("L_DBG", "1 ds: [$ds]\n"); lookup_branches('location'); xlog("L_DBG", "2 ds: [$ds]\n"); t_load_contacts(); xlog("L_DBG", "3 ds: [$ds]\n"); t_next_contacts(); xlog("L_DBG", "4 ds: [$ds]\n"); -- Nathan Ward

1 1

Peculiar Kamailio Asterisk behavior on outbound calls
by Wilkins, Steve 09 Jun '18

09 Jun '18

Hi All, Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses incorrect Video RTP Payload Type when sending Video packets. I have a situation where I make a call from Device A to Device B and Device A is Registered in Kamailio. When Device A Calls Device B, Kamailio sends an 'INVITE' to Asterisk, Asterisk then 'INVITES' Device B. I get two-way Audio, the call stays connected, however, when Video packets are sent to Device B, the RTP Payload Type is incorrect. The port is correct, but just not the Payload Type. Here is where I think Kamailio is involved. In the first Invite from Kamailio to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, Device B wants to use '115 H264' and when Asterisk sends out Video packets, it is using '100' instead of '115', and of course I have no Video. I don't know if this is just a coincidence but it sure seems like that is where the issue may lie. Has anyone ever seen this behavior? The Asterisk teams does not think it's an Asterisk issue. Thank you, -Steve

2 12

Randomizing t_load_contacts() with same q value?
by Daniel Tryba 08 Jun '18

08 Jun '18

I'm looking for a way to randomize the contacts for serial forking if they have the same q value (so instead of doing parallel forking to those targets). Without taking the q value into account I could do something with the defined contacts_avp before calling t_next_contacts(). But I have no idea how to randomize and serialize the branches with the same q value after calling t_next_contacts(). Any ideas?

3 2

Monitoring Dispatcher
by Tim Bowyer 08 Jun '18

08 Jun '18

Hi All, Currently trying to work out the best way to keep track of how many calls are being sent to any given dispatcher destination. I remember being able to do this years ago with the load balancer module in OpenSIPS: opensipsctl fifo lb_list. Dispatching algorithm '10' seems to be the best fit for my requirements, as I'm essentially wanting to send calls to a large quantity of FreeSwitch docker containers acting as fax receiving nodes. Equal weight, just filling them in some balanced fashion. Any hints or ideas would be much appreciated! Kind regards, Tim Bowyer

3 6

removing not update packaging scripts
by Sergey Safarov 07 Jun '18

07 Jun '18

Hello I created PR to remove not updated CentOS packaging files ( https://github.com/kamailio/kamailio/pull/1555/files) Now CentOS packaging is located at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/obs Is there anyone considers it necessary to leave a legacy packaging scripts for CentOS Sergey

2 1

hot switch debug mode
by Sergey Safarov 06 Jun '18

06 Jun '18

I experienced issue with kamailio after some time of uptime. Could you suggest way to enable debug output (equivalent of -ddd options for kamailio executable) for running daemon? May be using kamcmd? Sergey

3 3

RFC: dispatcher refactoring to use xavp and options for attributes
by Daniel-Constantin Mierla 06 Jun '18

06 Jun '18

Hello, I am starting a discussion here about my plans to do a bit of refactoring to dispatcher module and gather some feedback if people think of other improvements. 1) First is about moving from using avps to xavps. It should simplify at least the config in terms of modparams, but also makes it more coherent in storing the details of destination record for serial forking. Practically, from a bunch of avp parameters, there will be one to set the root xavp name, and the old "avps" will be fields inside the xavp. 2) The second is about the attributes field. I am considering to add the option to load additional attributes from dedicated columns when using database. So besides the attrs column, one can specify via a mod param additional columns to be loaded and added to attributes, like: modparam("dispatcher", "xattrs", "weight,socket,maxload") The attributes string will be the value of "attrs" column concatenated with "weight=val1;socket=val2;maxload=val3" (the valX being the corresponding values in those columns). This should make easier the management of the database records. For the option with a text file, no changes will be done. 3) Anything else that one thinks would be good to add to dispatcher? Cheers, Daniel PS. I wrote to both sr-dev and sr-users, to get the attention of the two communities. However, you can reply only to sr-users (larger community) if you are not on sr-dev, because this discussion is about features of the module, not technical aspects of coding. -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

2 2

Re: [SR-Users] Install IMS Modules over KAMAILIO 5.1
by John M 06 Jun '18

06 Jun '18

First compile Kamailio 5.1 binaries and start the services with Kamailio 4.x IMS configuration files. You need to remove few of the deprecated modules. Then compare the working configuration with 5.X IMS cfg files. You can use OpenIMS HSS application. Configuration is straight forward. Calls can be tested using Boghe IMS client. Regards, Uts Hello, not much into IMS myself, and I see you somehow sorted out the Kamailio part, but in case it helps, you can find sample configuration files in the source code tree at misc/examples/ims/ As for FHOSS, I am not sure if that project is active/maintained. Maybe other guys here using IMS extensions can comment more. Cheers, Daniel On 05.04.18 17:37, Roi Arias Salve wrote: >>* Hi all. *>> >>* I would like to thank you all the documents you have to install a *>* Kamailio server. It’s quite easy and fast. *>> >>* Currently I’m working on Azure, witha Debian 9 OS, and I’d followed *>* the instructions to install the Kamailio OPENSER, and everything Works *>* perfect. *>> >>* But now I’m stuck on my next step. I would like to install all the IMS *>* components, P-cscf/I-cscf/S-cscf/Fhoss following this document *>* https://www.kamailio.org/wiki/tutorials/ims/installation-howto <https://www.kamailio.org/wiki/tutorials/ims/installation-howto> *>> >>* But, I have no success. Maybe I’ve a mistake on the selection for the *>* Debian OS installed on the server. I say this because of this sentence: *>>* “ *Note: This document is work in progress and yet unfinished!* *>>* This is a short installation howto for Debian Wheezy (stable). *>>* “ *>> >>* So, I asume that maybe here I have a problem with the OS version, I’ll *>* try to summarize where I have problems: *>> >>* 1.- I’ve installed the , P-cscf/I-cscf/S-cscf not a problem. *>> >>* 2.- I am not able to follow this steps related to the repository: *>> >>* First, install the Key for our repository: *>>* wget -O - http://repository.ng-voice.com/PublicKey <http://repository.ng-voice.com/PublicKey> | apt-key add - *>>* Then add the repository to your “/etc/apt/sources.list”: *>>* echo "deb http://repository.ng-voice.com <http://repository.ng-voice.com/> jessie ims rtpproxy" >> *>* /etc/apt/sources.list *>> >>* and also I have problems with this action over the repository *>> >>* Note: You will need to add the “non-free” packages to your repository: *>>* deb http://ftp.de.debian.org/debian <http://ftp.de.debian.org/debian> lenny main non-free *>* deb http://security.debian.org/ <http://security.debian.org/> lenny/updates main non-free *>> >> >> >>* 3.- I decided to install directly the modules from an apt-get *>* install action, because I followed the instructions on the Log created *>* on the Kamailio starting box, where you can find all the steps related *>* to IMS modules, and the suggestion there it is to compile your own *>* modules. On this step I detected a problem related with the dialog_ng *>* module, according to the .log document on the Starting box, you have *>* to include all the modules before you execute the make all instruction: *>> >>* modify modules.lst for included modules and excluded modules (use *>* saved file) *>>* - include: ims_auth ims_charging ims_icscf ims_isc *>* ims_qos ims_registrar_pcscf ims_registrar_scscf ims_usrloc_pcscf *>* ims_usrloc_scscf dialog_ng cdp cdp_avp *>> >>* 4.- After this issue, I tried to install FHOSS, but here I can’t go *>* further, I am not able to compile it, neither doing my own compilation *>* following this step *>> >>* ant compile deploy | tee ant_compile_deploy.txt *>> >>* nor following *>> >>* apt-get install openimscore-fhoss *>> >>* I was no table to get connected to the repository, so I can’t go further. *>> >>* I’m sorry for all this information, can anyone help? Any suggestion? *>> >>* Regards *>>* Roi *>> >>>* <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai… <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>> *>* Libre de virus. www.avast.com <http://www.avast.com> *>* <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai… <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>> *>>>>>* _______________________________________________ *>* Kamailio (SER) - Users Mailing List *>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Re: [SR-Users] ASN.1 HI2- IRI-Parameters
by Mojtaba 06 Jun '18

06 Jun '18

Any idea in this regards? I understood it's structure of HI2Operations as ASN.1 IRI-HI2, I attached it. I think i should convert SIP signalling flow to this structure for transmission. Is it right way that i am doing? With Regards.Mojtaba On Thu, Apr 26, 2018 at 12:50 PM, Mojtaba <mespio(a)gmail.com> wrote: > Hello, > The declaration of details are in 3GPP TS 101-671 Annex D.5 (ASN.1 > description of IRI (HI2 interface)). > I need a sample packet of HI2Operations for underestanding, so that > i'll develop this HI in kamailio. > With Regards.Mojtaba > > > On Thu, Apr 26, 2018 at 11:20 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> can you provide more details or references (web links) about the operations >> you want to do? Like what sip message comes to kamailio and what you need to >> change to it. >> >> Cheers, >> Daniel >> >> >> On 25.04.18 06:00, Mojtaba wrote: >> >> Hello. >> I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose >> anybody have experience in this regards? >> Thanks. >> Mojtaba >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com > > > > -- > --Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S

2 2

TOPOS module errors
by Enrico Bandiera 06 Jun '18

06 Jun '18

Hi, I'm trying to use the TOPOS module to solve some MTU issues we have (we have multiple kamailio servers in our architecture and route/via headers are becoming a problem) The module seems to work but I'm getting these errors on the log: 0(94) ERROR: topos [tps_storage.c:480]: tps_db_insert_dialog(): No database handle - misconfiguration? 0(94) ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store I get these errors for some time after running kamailio and then they stop appearing. Data is inserted on topos_d and topos_t on the db, so I'm not sure what the error is referring to. Any idea? Thanks, Enrico.

4 14

problems to parse Cseq 0 value
by David Escartin 05 Jun '18

05 Jun '18

hello all we have seen that when an OPTIONS is sent with CSep: 0 OPTIONS, message is not parsed. Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/parse_cseq.c:61]: parse_cseq(): no method found Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [CSeq: 0 Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: WARNING: <core> [core/receive.c:198]: receive_msg(): parsing relevant headers failed as far i see in rfc3261 "The sequence number value MUST be expressible as a 32-bit unsigned integer and MUST be less than 2**31." i don't know if you already discussed about this, but do you think it would be something to fix? best regards and thanks a lot david escartin

2 4

Kamailio v5.1.4 Released
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, Kamailio SIP Server v5.1.4 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.4. For more details about version 5.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/06/kamailio-v5-1-4-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.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address
by Minh Phan 05 Jun '18

05 Jun '18

Hi List, I have this warning message in kamailio log when Kamailio handle an OPTIONS message with topos module enabled (I am using Kamailio 5.1.2): WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests In my config file, OPTIONS message is handled this way: if ( is_method("OPTIONS") ) { t_newtran(); record_route(); t_reply("200","OK"); exit; } I have found this issue on Github but it is closed and I am not sure what was the solution: #1483 <https://github.com/kamailio/kamailio/issues/1483> My server receives 4 OPTIONS messages every 10 seconds that give as many warning in my log file. What could I do to get rid of them? Thanks, Minh

3 5

Releasing Kamailio v5.1.4
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, I am considering to release a new minor version out of branch 5.1, respectively the v5.1.4, sometime next week, likely on Tuesday, June 5, 2018. As usual, if there is any issue not yet reported on github tracker, do it as soon as possible to give it a chance to be fixed. Soon I am going to do the backports of commits with fixes in master that apply in branch 5.1, so keep a close eye and if someone is missing, let me know to push it in the stable branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

3 3

Transaction flags and branch_routes
by Alex Balashov 05 Jun '18

05 Jun '18

Hi, Given: flags PROXY_MEDIA_SET : 10, SIPPING : 11, DUMMY2 : 12 And: request_route { ... route(XYZ); } route[XYZ] { setflag(PROXY_MEDIA_SET); t_on_branch("BRANCH_ROUTE"); if(!t_relay()) sl_reply_error(); } branch_route[BRANCH_ROUTE] { if(isflagset(PROXY_MEDIA_SET)) { ... } ... } Is PROXY_MEDIA_SET meant to be visible inside the branch_route? Because it doesn't seem to be. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

ACK to 200 OK without to tag
by Kjeld Flarup 04 Jun '18

04 Jun '18

We have a SIP doorstation Grandstream GDS3710. When I connect it via kamailio, it does not setup call correct. Kamailio sends a 200 OK with this to field: t: <sip:0001@192.168.1.150:5069>;tag=e33a13d3-c18c-4df1-b68f-446e42957de6 Grandstream answers with: To: <sip:0001@192.168.1.150:5069> As the code follows this example: https://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.…, it seems obvious that this will fail, as there is not route[WITHINDLG] { if (has_totag()) { There simply is not to tag. Is this a bug in the grandstream? If so how could that go unnoticed, or is it me who misses something? I have attached the full sip text for the 200OK and the ACK -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 3

Jumping on new TCP connection during a SIP session?
by Abdoul Osséni 04 Jun '18

04 Jun '18

Hello group, The call flow is: useragent (TLS) -> Kamailio -> Sems Server. During the establishment of a call (before the 200OK or during a retransmission of the 200OK issued by Sems Server via Kamailio), if a user agent (TCP) loses its connection (the user agent is on a mobile network for example or the network is not reliable.) and sends a "register" to Kamailio (same source IP address but a different source TCP port), is it possible for Kamailio to take into account this new TCP port with the function t_relay () ? Abdoul

4 4

Module sanity - memory leaks
by Konstantin Polyakov 04 Jun '18

04 Jun '18

Hello, We had memory leak on a production server and to find it we removed optional modules one by one. As result we found that problem is in the sanity module. Unfortunately I don't have time to fix it since I am going to vacation. We used the following checks: ruri_sip_version (1), required_headers (4), cseq_method (32), cseq_value (64), cotent_length (128), expires_value (256), proxy_require (512), parse uri's (1024). So be aware that this module is not ready to use in production. Best regards, Konstantin

1 0

configuring websockts in kamilio
by vinod mn 04 Jun '18

04 Jun '18

hi configured websockets on kamilio 5.1.3 I am trying to register the client based on webrtc client ,(sip.js) the registration is fine but i am not getting the correct address if i run "*kamctl ul show"* * In the address filed I want the sip:<id>@<ip>:<port>ex: sip:12345@127.0.0.1:12345 <http://sip:12345@127.0.0.1:12345> * output shows: { "jsonrpc": "2.0", "result": { "Domains": [{ "Domain": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "1xxxxxxxxx", "HashID": 1xxxxx, "Contacts": [{ "Contact": { *"Address": "sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, "Expires": 397, "Q": 1, "Call-ID": "ma4feu2i43bnu6ophe4k78", "CSeq": 86, "User-Agent": "SIP.js/0.5.0-devel BB", "Received": "sip:X.X.X.X:56087;transport=ws", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "tcp:X.X.X.X:8080", "Methods": 783, "Ruid": "uloc-5b116457-1dc1-03", "Instance": "<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", "Reg-Id": 1, "Server-Id": 0, "Tcpconn-Id": 259, "Keepalive": 1, "Last-Keepalive": 1527876355, "Last-Modified": 1527876355 } }] } } ], "Stats": { "Records": 1, "Max-Slots": 1 } } }] }, "id": 18958 } In the address filed I want the sip:<id>@<ip>:<port> ex: sip:12345@127.0.0.1:12345 Please give me any suggestions on this, which webrtc client can i use to register to kamailio and able to receive on the client and to make outbound calls from the client. -- Thanks and regards Vinod.M.N

2 1

Kamailio Behind TCP Load Balancer
by Colin Morelli 04 Jun '18

04 Jun '18

Hey all, I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware. My application is deployed in AWS and I'm tying to place Kamailio behind an ELB. For the most part, everything is fine. For my specific implementation I'm disabling UDP as a signaling transport and using only TLS. This enables me to not have to worry about a SIP-aware LB at the edge because replies to an incoming request will be sent over the existing established TCP socket (avoiding any crazy routing requirements). However - this poses an issue with source addresses. Does Kamailio support anything like the proxy protocol ( http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP stream information from a load balancer? Do I need to go back to exposing it directly to the world so that I can get source addresses? Thanks, Colin

3 4

Re: [SR-Users] configuring websockts in kamilio
by Konstantin Polyakov 02 Jun '18

02 Jun '18

Hello Vinod, We used Doubango sipML5 web phone https://www.doubango.org/sipml5/call.htm?svn=252 It was quite good for debugging and testing webrtc calls. Most of work in kamailio config I made by using that client. Best regards. Konstantin > >Message: 8 >Date: Fri, 1 Jun 2018 23:50:50 +0530 >From: vinod mn < vinodmn.dbit(a)gmail.com > >To: sr-users(a)lists.kamailio.org >Subject: [SR-Users] configuring websockts in kamilio >Message-ID: >< CAHeWXy7h4Tcp4Sg9_eM3teCHBdJ5bwHa9TYSJy+7P4e+bm1hqQ(a)mail.gmail.com > >Content-Type: text/plain; charset="utf-8" > >hi configured websockets on kamilio 5.1.3 >I am trying to register the client based on webrtc client ,(sip.js) > the registration is fine but i am not getting the correct address if i >run "*kamctl ul show"* > > >* In the address filed I want the sip:<id>@<ip>:<port>ex: >sip:12345@127.0.0.1:12345 < http://sip:12345@127.0.0.1:12345 > * >output shows: >{ > "jsonrpc": "2.0", > "result": { > "Domains": [{ > "Domain": { > "Domain": "location", > "Size": 1024, > "AoRs": [{ > "Info": { > "AoR": "1xxxxxxxxx", > "HashID": 1xxxxx, > "Contacts": [{ > "Contact": { > *"Address": >"sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, > "Expires": 397, > "Q": 1, > "Call-ID": "ma4feu2i43bnu6ophe4k78", > "CSeq": 86, > "User-Agent": "SIP.js/0.5.0-devel BB", > "Received": "sip:X.X.X.X:56087;transport=ws", > "Path": "[not set]", > "State": "CS_NEW", > "Flags": 0, > "CFlags": 192, > "Socket": "tcp:X.X.X.X:8080", > "Methods": 783, > "Ruid": "uloc-5b116457-1dc1-03", > "Instance": >"<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", > "Reg-Id": 1, > "Server-Id": 0, > "Tcpconn-Id": 259, > "Keepalive": 1, > "Last-Keepalive": 1527876355, > "Last-Modified": 1527876355 > } > }] > } > } > ], > "Stats": { > "Records": 1, > "Max-Slots": 1 > } > } > }] > }, > "id": 18958 >} > >In the address filed I want the sip:<id>@<ip>:<port> >ex: sip:12345@127.0.0.1:12345 > >Please give me any suggestions on this, which webrtc client can i use to >register to kamailio and able to receive on the client and to make outbound >calls from the client. > >-- >Thanks and regards >Vinod.M.N >

1 0

CDRs
by David Villasmil 02 Jun '18

02 Jun '18

Hello guys, I setup the CDRs according to the instructions on https://kb.asipto.com/siremis:install40x:accounting I get CDRs nicely, but the cdrs, even though they have a src_ip, there's no dest_ip, I mean I need the IP where the call was forwarded to... What do i need to do to get that in the cdr? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Upgrade issue from 4.4.7 to 5.1.3
by Kevin Greene 01 Jun '18

01 Jun '18

Hello List users: I am trying to upgrade to the current version of Kamailio from the older 4.4 version to the current version. The 4.4 is functional but there are some features of 5.1 that we wanted to test with. It seems that my version of Linux (RHEL 6.9 or Oracle 6.9) has an issue with a library file. Any help would be useful. Thank you. Kevin Notes: I used the "yum update kamailio" function. (edited for clarity - it showed a long list of modules to be updated) "--> Finished Dependency Resolution Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: libkmi.so.1()(64bit) Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) libkmi.so.1()(64bit) Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) Not found Error: Package: kamailio-xml-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest" However the "missing" library is there in /usr/lib64. [root@sipserver tmp]# find / -name libkmi.so.1 /usr/lib64/kamailio/libkmi.so.1 /usr/lib64/libkmi.so.1 /lib64/libkmi.so.1 To provide additional help, See: [root@sipserver tmp]# kamailio -version version: kamailio 4.4.7 (x86_64/linux) 35d8ab flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 35d8ab compiled on 15:38:14 Feb 26 2018 with gcc 4.4.7

2 1

configuring the websocket in kamailio 5.1
by vinod mn 01 Jun '18

01 Jun '18

Hi, I configured the websocket in kamailio and it is running properly, which is the best webrtc client (with proper code as I dont't want to spend more time in writing the codes based on the example snippets ) can I use for registering, and making/ receiving the calls on the webrtc-client . requirement: webrtc-client(??)->register-> kamailio(already configured to use websocket) webrtc-client -> makecall to kamailio webrtc-client receives call from another client(sip not webrtc-client) -- Thanks and regards Vinod.M.N

1 0

Thank You
by Wilkins, Steve 01 Jun '18

01 Jun '18

Hi All, I normally get on this board to ask questions but today I just wanted to take a minute and say - Thank you to all of you who have taken the time to help me. Some of you have spent a lot of time schooling me on Kamailio, its Interactions with a PBX and different clients. I have learned tons from this board and just want to say again, Thank You All! -Steve

2 1

Multiple CDR rate groups using Siremis sql procedures
by Sergiu Pojoga 01 Jun '18

01 Jun '18

Hi, The SQL procedure takes a param, rgroup, which is default (by default). That works fine. In a real case scenario, having multiple rate groups, I wonder what's the best approach to rate CDRs based on the rate group a client belongs to? Do you store the rate_group in the ACC table and then adjust 'kamailio_cdrs' and 'kamailio_rating' procedures to take into account the rate_group of a CDR? If so, does anybody have a working example? Thanks.

1 0

Anybody using 2 instances with topos communicating with each other?
by Daniel Tryba 01 Jun '18

01 Jun '18

Scenario: endpoint (bob) <-> kamailio proxy (5.1.3) <-> kamailio backend (5.1.2) Both kamailio instances are running the topos module (mysql backend). Call setup is fine, but forwarding BYEs fail. With a call from endpoint to kamailio, topos on the proxy scrambles Contact to kamailio backend. But in responses from backend contact remains untouched. In the ACK to the 200, topos on the proxy does nothing. If the caller hangsup, the BYE will loop internally till max hops is exceeded (483 which will be send to caller eventually). When the backend initiates the call to the endpoint, topos appears to work, but the BYE from the endpoint on hangup get a 404 Not here from the proxy (route[WITHINDLG]). Disabling topos on the proxy will result in a functioning proxy (config attached). Is there something wrong with my config? Or is this an unforseen scenario that just doesn't work with topos? * Endpoint to backend scenario endpoint:5060 -> proxy:5060 INVITE sip:callee@sip.pocos.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy:5060;lr> Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 INVITE sip:callee@sip.itco.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... CSeq: 29848 INVITE Route: <sip:proxy:5060;lr> Contact: <sip:btpsh-5af5972d-bf5e-2@proxy> backend:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:atpsh-5af19bf0-9f94-7@backend> proxy:5060 -> endpoint:5060 SIP/2.0 200 OK Contact: <sip:atpsh-5af19bf0-9f94-7@backend> Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Route: <sip:proxy;lr> proxy:5060 -> backend:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... endpoint:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 69 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP proxy;rport=5060;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 68 Route: <sip:proxy;lr> [not going to mention the 50+ BYEs adding Vias and decrementing Max-Forwards. * Backend to endpoint scenario backend:5060 -> proxy:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP backend;branch=... Route: <sip:loadbalancer@proxy;lr;received=sip:endpoint:5060> Contact: <sip:btpsh-5af07a27-72b-502c1@backend> proxy:5060 -> endpoint:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:btpsh-5af5972d-bf5d-5@proxy> endpoint:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;received=proxy;branch=... Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 SIP/2.0 200 OK Contact: <sip:bob@endpoint:5060;ob;alias=endpoint~5060~1> Via: SIP/2.0/UDP backend;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 BYE sip:btpsh-5af5972d-bf5d-5@proxy SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... proxy:5060 -> endpoint:5060 SIP/2.0 404 Not here Via: SIP/2.0/UDP endpoint:5060;rport=5060;branch=...

2 2

Dispatcher module ERROR
by Aleksandar Sosic 01 Jun '18

01 Jun '18

Hi everyone, We have an error using the dispatcher module. In the logs: ```0(107) ERROR: dispatcher [dispatcher.c:815]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled``` We are running 5.2.0~dev5+0~20180523010152.1118+xenial The configuration file parts regarding dispatcher are as follows: ``` loadmodule "dispatcher.so" ... # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)# ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)``` What are we doing wrong? What are we missing? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

2 2

Authenticate against outbound carrier
by Krzysztof Grobelak 01 Jun '18

01 Jun '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

1 0

Acc module and PostgreSQL
by Javohir Maxmarajabov 01 Jun '18

01 Jun '18

Is it possible to enable accounting module with PostgreSQL? All I found is documentation and examples that showing accounting module is used with MySQL module. -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

2 1

Strange behavior with $shtinc
by Brooks Bridges 01 Jun '18

01 Jun '18

Let me preface this with the statement that as best I can tell from docs and code, this function is supposed to be 100% atomic and this shouldn't be able to happen. I also have not found anything in the module's github commit logs that would indicate this behavior has changed from previous versions. That being said, I appear to have run into an oddity where intermittently I'm getting 2 slots being put in the same entry, which results in trying to parse the output for graphing and reporting not working quite right. The mechanism I'm using is a very simple counter mechanism to record statistics about calls. As an example, I am using the following mechanism: $var(z) = $shtinc(callstats=>$avp(realm),cps_exceeded); or $var(z) = $shtinc(callstats=>$avp(realm),sessions_exceeded); This typically results in something like this: { entry: 4335 size: 1 slot: { { name: realm1,total_requests value: 2365034 type: int } } } { entry: 4532 size: 1 slot: { { name: realm2,cps_exceeded value: 30 type: int } } } And so on... However, one or more of them sometimes end up like this: { entry: 4646 size: 2 slot: { { name: realm1,total_requests value: 15958026 type: int } { name: realm2,cps_exceeded value: 6432103 type: int } } } Thanks in advance for your help and/or suggestions! Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 5

Authentication against outbound carrier trunk
by Krzysztof Grobelak 31 May '18

31 May '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

1 0

error in tls configuration in kamailio 5.1.3
by Narasimhaiah, Vinod Maralenahalli 31 May '18

31 May '18

Hi, I upgraded kamailio 4.4.2 to kamailio5.1.3 and i am getting this error *"Unable to load certificate file '/etc/kamailio/kamailio-selfsigned.pem",* kamilio 4.4.2 does not need the ssl file, but kamailio 5.1.3 need ssl file, what changes required in kamailio 5.1.3 to run with tls. Please help me in resolving this issue. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

4 4

Upgrade issue with old Kamailio 3.0.3 in table rls_presentity
by Kristijan Vrban 31 May '18

31 May '18

Hi, i try a upgrade from old 3.0.3 Kamailio mysql db. Got stuck with update the structure : ALTER TABLE rls_presentity MODIFY callid VARCHAR(255) NOT NULL;ALTER TABLE rls_presentity MODIFY contact VARCHAR(128) NOT NULL; found on: https://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0 But this two table does not exist? See also table definition at: https://kamailio.org/docs/db-tables/kamailio-db-3.1.x.html#GEN-DB-RLS-PRESE… I reckon it is safe to skip this to tables? Kristijan

2 4

Register on kamailio.org wiki page not working
by Kristijan Vrban 31 May '18

31 May '18

Hi, https://www.kamailio.org/wiki/start?do=register Is not working. You get a empty page here. Kristijan

2 2

Sharing static variables in external library across child proceses
by User 721 31 May '18

31 May '18

Hi, I am adding a kamailio module that references an external C library that maintains a hashtable in global scope. I am wondering what practice I should follow so that: - This hashtable (which I don't control directly) is shared by all child processes - The hashtable is changed by one child process at a time Thanks in advance

2 1

PIKE module
by Robert Barrera 31 May '18

31 May '18

Good morning! We operate a VoIP platform and are interested in integrating the PIKE module. Is it possible for someone to assist us to discuss how this can be done and our options? Thank you logo_nop Robert Barrera 305-663-8989 305-600-4786 fax info(a)myrealmobile.com Complete Business PBX phone systems and Nationwide wireless! REAL Mobile does it!

2 1

sip_msg structure and memory free
by Konstantin Polyakov 31 May '18

31 May '18

Hello All, There is a structure sip_msg which is passed into a module and represents a received sip message. It contains number of pointers to hdr_field members. In its turn hdr_field contains member void* parsed In the sanity module we assign to parsed member a pointer to newly allocated memory and we don't free that memory at the end. Do we need to free explicitly or this memory will be deallocated later together with sip_msg ? In other words is it memory leak or not? Best regards, Konstantin

2 1

General Kamailio with Asterisk (or another PBX) Opinion
by Wilkins, Steve 30 May '18

30 May '18

HI All, I am curious, for those of you who use Kamailio with Asterisk or another PBX. Who forwards Registrations to Asterisk or PBX, and who lets Kamailio maintain Registrations? Thanks All!

4 3

send CANCEL to caller after hitting fr_inv_timeout
by Sergiu Pojoga 29 May '18

29 May '18

TM's noisy_ctimer=1 seems to take care of sending a CANCEL to the callee, but it eludes me which tm param makes a CANCEL to be sent to caller as well? Caller gets a '408 Request Timeout' only, which in my case doesn't effectively terminate a call for the calling party. Merci.

1 0

INFO: stun_parse_header: incomplete header of STUN message
by Pan Christensen 29 May '18

29 May '18

Hello! After proxying a REGISTER from a WebRTC client to our SIP backend, the backend starts sending keepalives. Every 30 seconds I get a log entry in Kamailio saying: "INFO: stun [kam_stun.c:169]: stun_parse_header(): INFO: stun_parse_header: incomplete header of STUN message". Here's one such packet: No. Time Source Destination Protocol Length Info 1 0.000000 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy UDP 60 5060 ? 15060 Len=4 Frame 1: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: 02:81:64:58:cf:d1 (02:81:64:58:cf:d1) Internet Protocol Version 4, Src: xxx.xxx.xxx.xxx, Dst: yyy.yyy.yyy.yyy 0100 .... = Version: 4 .... 0101 = Header Length: 20 bytes (5) Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT) Total Length: 32 Identification: 0x5bf5 (23541) Flags: 0x4000, Don't fragment Time to live: 56 Protocol: UDP (17) Header checksum: 0x2fc3 [validation disabled] [Header checksum status: Unverified] Source: xxx.xxx.xxx.xxx Destination: yyy.yyy.yyy.yyy User Datagram Protocol, Src Port: 5060, Dst Port: 15060 Source Port: 5060 Destination Port: 15060 Length: 12 Checksum: 0xfa28 [unverified] [Checksum Status: Unverified] [Stream index: 0] Data (4 bytes) 0000 00 00 00 00 .... Data: 00000000 [Length: 4] I found this in kam_stun.c: static int stun_parse_header(struct stun_msg* req, USHORT_T* error_code) { if (sizeof(req->hdr) > req->msg.buf.len) { /* the received message does not contain whole header */ LOG(L_INFO, "INFO: stun_parse_header: incomplete header of STUN message\n"); /* Any better solution? IMHO it's not possible to send error response * because the transaction ID is not available. */ return FATAL_ERROR; } ... Could someone please explain what's wrong? Med vennlig hilsen Pan B. Christensen Utvikler Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-post: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobil: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

2 3

Dialplan module for load balancing
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I'm maybe confused on how to use dialplan module. I have a dialplan setup for a long time now and I'd like to add a new rule with the same match_exp than an existing one but with a different repl_exp. If I set the same priority, the latest entry is used systematically. If I set priority +1, the new rule is never used. The goal is having two rules and alternatively reply with one or the other repl_exp. Is it possible? Regards, Igor.

1 0

Pkg memory stats
by Konstantin Polyakov 29 May '18

29 May '18

Hello, Could somebody explain what do numbers mean for command "kamcmd mod.stats all pkg"? Is there somewhere documentation about that? I can see the following information. " Module: xlog { mod_init(213): 4104 xlog_fixup_helper(533): 3176 xdbg_fixup_helper(499): 632 Total: 7912 } " I guess that numbers after colon are bytes. But what do numbers in (213) mean? When I use command "kamcmd pkg.stats", I got absolutely different numbers which don't match previous statistics? What is wrong? What is the best way to analyze pkg memory usage? Thank you Best regards Konstantin

1 0

Realtime Extensions SQLITE
by Benjamin Marty 29 May '18

29 May '18

I compiled a fresh Asterisk 13.18 certified with res_config_sqlite. I configured following: extconfig.conf [settings] sippeers => sqlite,asterisk,ast_config res_config_sqlite.conf [general] dbfile => /home/myuser/database.db On Bootup it gives the following error: [May 29 17:07:14] WARNING[32444]: res_config_sqlite.c:1206 realtime_multi_handler: no such table: ast_config Is the sqlite module deprecated or is my configuration wrong? Couldn't really find any documentation on that topic. The table ast_config is definitely present in the sqlite file.

2 1

Serial forking
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I know that the default behaviour of two aliases in kamailio.aliases result in parallel forking. I'm wondering if its possible to do serial forking instead? For example, in case that the first INVITE has no 180/183, or neither 100 Trying, then send the INVITE to the second entry? Regards, Igor.

2 2

Switching form UDP to TCP causes authentication errors
by Wilkins, Steve 29 May '18

29 May '18

Hello All, When I switched from UDP to TCP I started getting Authentication Errors, Asterisk responds to an INVITE via Kamailio with a '401 Unauthorized', but Kamailio does nothing with it. Processing just stops near WITH_BLOCK401407. Shouldn't the 401 be relayed so a new INVITE can be sent? Thank you.

3 4

ims_isc: 3rd party Registration (S-CSCF) and P-Associated-Uri header
by Valentin Christoph 29 May '18

29 May '18

Hi, The function build_p_associated_uri() in module ims_isc filters for TEL URIs. Why are not ALL unbarred IMPUs added, but only TEL URIs? Is this specified in some 3GPP spec? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

jsonrpc http interface cant recognize method
by Yuriy Gorlichenko 29 May '18

29 May '18

Hi i trying to get htable.dump mytable via jsonrpc request {jsonrpc: "2.0", method: "htable.dump mytable"} { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" } } But via kamcmd it wroks perfect Aslo i see that if i will make request withont param (core.version for example) IT works fime How to make request with param?

1 1

Carrierroute Wholesale termination issue
by riaz hasan 29 May '18

29 May '18

Hello, I am facing an issue with kamailio carrierroute.I have a setup where multiple asterisk is behind a kamailio server.I am able to dispatch the load by using carrierroute. Its working perfectly for retail users.But problem is its not working for wholesale users.I am attaching the dump for wholesale client. here is a little descrption for the shark:- 192.151.149.102 is the wholesale client ip. 46.166.139.60:4848 is kamailio server. 46.166.139.60:3628 is asterisk server. when 192.151.149.102 sends invite to 46.166.139.60:4848 [Kamailio] we are getting 401. [dump2.pcap] when 192.151.149.102 sends invite to 46.166.139.60:3628 [Direct Asterisk ] we are getting 200. [dump.pcap] here is the code i am using for load balancing:- route[Carrer] { #!ifdef WITH_CARRIERROUTE if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); exit; } #!endif $avp(s:host)= $rd+":"+$rp; t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { revert_uri(); #!ifdef WITH_CARRIERROUTE if (!cr_next_domain("default", "default", "$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) { xlog("L_ERR", "cr_next_domain failed\n"); exit; } if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); } t_on_failure("1"); route(NASIR); exit; #!endif $avp(s:host)= $rd+":"+$rp; if (!t_relay()) { xlog("L_ERR", "t_relay failed-custom\n"); exit; }; }

1 0

S-SCSF not forwarding SIP packets to AS
by John M 29 May '18

29 May '18

Hi All, My setup consists of Kamailio P-CSCF,I-SCSF , S-CSCF and OpenIMS HSS nodes and I have configured application server details and trigger point on HSS. I am using Boghe IMS client SIP validation. Whenever the client sends a REGISTER, messages are not reaching AS. In S-SCSF log files, I am seeing " isc_match_filter_reg(): No REGISTER match found in filter criteria" and " isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group" error message. Kmailio version: version: kamailio 5.1.3 (x86_64/linux) HSS XML The UserData XML document which is sent to the S-CSCF: <?xml version="1.0" encoding="UTF-8"?><IMSSubscription><PrivateID> alice(a)ims.mnc001.mcc001.3gppnetwork.org </PrivateID><ServiceProfile><PublicIdentity><Identity> sip:alice@ims.mnc001.mcc001.3gppnetwork.org </Identity><Extension><IdentityType>0</IdentityType></Extension></PublicIdentity><InitialFilterCriteria><Priority>0</Priority><TriggerPoint><ConditionTypeCNF>0</ConditionTypeCNF><SPT><ConditionNegated>0</ConditionNegated><Group>0</Group><Method>REGISTER</Method><Extension><RegistrationType>0</RegistrationType></Extension></SPT></TriggerPoint><ApplicationServer><ServerName>sip:192.168.20.10:5060 ;transport=TCP</ServerName><DefaultHandling>0</DefaultHandling></ApplicationServer></InitialFilterCriteria></ServiceProfile></IMSSubscription> SCSF Log-1 0(1541) DEBUG: ims_isc [ims_isc_mod.c:428]: isc_match_filter_reg(): Checking triggers 0(1541) DEBUG: ims_isc [ims_isc_mod.c:432]: isc_match_filter_reg(): Starting triggering 0(1541) DEBUG: ims_isc [ims_isc_mod.c:446]: isc_match_filter_reg(): Orig User before clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [ims_isc_mod.c:448]: isc_match_filter_reg(): Orig User after clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [checker.c:416]: isc_checker_find(): isc_checker_find: starting search 0(1541) DEBUG: ims_isc [checker.c:438]: isc_checker_find(): isc_checker_find(): got a r_public for the user sip:alice@ims.mnc001.mcc001.3gppnetwork.org 0(1541) DEBUG: ims_isc [checker.c:454]: isc_checker_find(): About to try p->s->service_profiles_cnt!! #profiles is 1 0(1541) DEBUG: ims_isc [checker.c:456]: isc_checker_find(): About to try p->s->service_profiles[a].filter_criterai_cnt 0(1541) DEBUG: ims_isc [checker.c:467]: isc_checker_find(): DEBUG ISC: SECOND TIME About to try p->s->service_profiles_cnt!! 0(1541) DEBUG: ims_isc [checker.c:471]: isc_checker_find(): DEBUG ISC : About to try p->s->service_profiles 0(1541) DEBUG: ims_isc [checker.c:474]: isc_checker_find(): DEBUG ISC : About to try public identities 0(1541) DEBUG: ims_isc [checker.c:477]: isc_checker_find(): DEBUG ISC : About to try WPSI 0(1541) DEBUG: ims_isc [checker.c:319]: isc_check_filter_criteria(): ifc_checker_trigger: Starting expression check: 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 0(1541) DEBUG: ims_isc [ims_isc_mod.c:460]: isc_match_filter_reg(): No REGISTER match found in filter criteria SCSF Log-2 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 1 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 2 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 Regards, Uts

1 0

Kamailio 5.1.x - Dialog event route not fired
by Loic Chabert 28 May '18

28 May '18

Hi list, I'm facing a strange behaviour with dialog module and stop event route. When a BYE request has been receved by Kamailio, no stop event route has been fired but dialog state go to 5. My config is following this guide: http://kb.asipto.com/kamailio:presence:k43-blf Using dialog_manage on RELAY route, as describe on this guide. If i set more debug information, i can see dialog has been found and dialog status incremented : 1(7) DEBUG: <core> [core/udp_server.c:491]: udp_rcv_loop(): received on udp socket: (106/100/818) [[BYE sip:172.16.74.201:5060;prxroute=1 SIP/2.0 0D 0A Record-Route: <sip:172.16.74.24;r2=on;lr=on;ftag=1;ng]] 1(7) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <BYE> 1(7) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:172.16.74.201:5060;prxroute=1> 1(7) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKc2c.939fa953ea648059586a7f684fd56b5a.0>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <5060>; state=6 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-69-1-10>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=78 1(7) DEBUG: <core> [core/parser/msg_parser.c:505]: parse_headers(): this is the second via 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=f651e13a-0aac-40d5-b877-1c24497ff87c 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [87]; uri=[sip:+33975181257@sip.testinfratrunk.vox.fr] 1(7) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body [<sip:+33975181257@sip.testinfratrunk.vox.fr>] 1(7) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <3> <BYE> 1(7) DEBUG: <core> [core/receive.c:232]: receive_msg(): --- received sip message - request - call-id: [1-69(a)60.0.0.30] - cseq: [3 BYE] 1(7) DEBUG: <core> [core/receive.c:287]: receive_msg(): preparing to run routing scripts... 1(7) DEBUG: maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 69 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7) DEBUG: sanity [sanity_mod.c:256]: w_sanity_check(): sanity checks result: 1 1(7) NOTICE: <script>: {1 3 BYE 1-69(a)60.0.0.30} Initialisation des variables. 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=1 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:476]: t_lookup_request(): start searching: hash=3116, isACK=0 1(7) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7) DEBUG: tm [t_lookup.c:1087]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T end=(nil) 1(7) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(7) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found 1(7) DEBUG: rr [loose.c:960]: loose_route(): There is no Route HF 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:1094]: t_check_msg(): T previously sought and not found for msg (0x7ffa420727d8) 1(7) DEBUG: dialog [dlg_hash.c:768]: internal_get_dlg(): ref dlg 0x7ffa327dd588 with 1 -> 3 1(7) DEBUG: dialog [dlg_hash.c:770]: internal_get_dlg(): dialog callid=' 1-69(a)60.0.0.30' found on entry 3063, dir=1 1(7) DEBUG: dialog [dlg_profile.c:536]: set_current_dialog(): setting current dialog [3063:9173] 1(7) DEBUG: dialog [dlg_hash.c:1151]: next_state_dlg(): dialog 0x7ffa327dd588 changed from state 4 to state 5, due event 7 (ref 3) 1(7) DEBUG: dialog [dlg_hash.c:932]: dlg_ref_helper(): ref op on 0x7ffa327dd588 with 1 from dlg_handlers.c:1389 1(7) DEBUG: dialog [dlg_hash.c:936]: dlg_ref_helper(): ref dlg 0x7ffa327dd588 with 1 -> 4 1(7) DEBUG: dialog [dlg_handlers.c:1400]: dlg_onroute(): BYE successfully processed Do you have any hints concerning this issue ? Is it a bug or a config part is missing ? Thanks in advance, Best regards.

2 1

Loop Detect - 482 Merged Request
by Amar Tinawi 28 May '18

28 May '18

Hello All i'm facing strange behavior with Zoiper Client only in an IMS Kamailio based system. in IMS call scenario (ims to ims) a lot of Loop Detect Msgs Appeared in the sip log. while other clients like Acrobits did not occurred i'm using TCP protocol and Wireshark shows a lot of (TCP Re-transmission and TCP ACKed unseen Segment) Any Suggestion Thanks !

1 0

Link problem between kamailio and MariaDB
by Ginhoux, Patrick 28 May '18

28 May '18

Hi, At the moment, I'm trying to replace MySQL by MariaDB on my Kamailio server (5.0.1) . In previous post I have been told that Kamailio supports MariaDB and there is no special module to load. This work is done on a new server running RHEL 7.4 and MariaDB 10.2.14 that is installed under /opt/mysql/na/10.2.14 folder. I encounter different problem: - Without changing anything in the kamailio script, kamailio failed to find the libmysqlclient.so.18 library. - So I thought that I could get rid of the above error by changing the MODULEPATH in the Kamailio-local.cfg as follows : MODULEPATH="/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/" But with this, I get 2 problem: The kamailio.log that was written in a /opt/log/Kamailio folder is no longer updated. I can only find the errors in the /var/log/messages file: May 23 11:19:19 vm-siprouter74 systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... May 23 11:19:19 vm-siprouter74 kamailio: Not starting : invalid configuration file! May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [DEFINE_LOG_FACILITY] value [LOG_LOCAL0] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN] value [udp:129.227.82.109:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN_2] value [udp:127.0.0.1:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [MODULEPATH] value ["/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/"] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1632]: yyparse(): loading modules under /usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/ May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module db_mysql.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/db_mysql.so> May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/lib64/kamailio/modules/db_mysql.so>: libmysqlclient.so.18: cannot open shared object file: No such file or directory May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) CRITICAL: <core> [core/cfg.y:3401]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 191, column 12-24: failed to load module May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module jsonrpcs.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/jsonrpcs.so> To work around the problem we have created 2 symbol links: <<< ln -s /opt/mysql/na/10.2.14/lib64/libmysqlclient.so.18.0.0 /usr/lib64/libmysqlclient.so.18 >>> the libmysqlclient error <<< ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf >>> to fix the issue to open the db It seems that kamailio is always looking for the my.cnf in the /etc/ directory. My concern are to share this job with people who have much more experience with Kamailio. I would like to know - If the different workaround (ln) makes sense to you or is there better change to do - How to explain the logging issue when changing the modulepath variable - Is it true that kamailio is always looking for the my.cnf in the /etc/ directory? Any thoughts are welcome. Thanks in advance. Cordialement. Patrick GINHOUX

2 2

Developer Guide
by Igor Olhovskiy 27 May '18

27 May '18

Hi! Is http://www.asipto.com/pub/kamailio-devel-guide/ still holds actual info? Cause it’s mostly for 3.3 version, and now is 5.1 And where to read, except source code of course, about how to develop modules for Kamailio? Regards, Igor

2 1

Compact Headers
by Duarte Rocha 26 May '18

26 May '18

Greetings, Is there any method on Kamailio that automatically rewrites the headers in its compact form ? I know how i can do it by re-writing each of the eligible headers, but i was wondering if there is any way to do it automatically. Thanks

2 1

UAC Module Registration Table
by Greg Adams 25 May '18

25 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg ________________________________

1 0

kamailio.org server maintenance work
by Oriol Capsada 25 May '18

25 May '18

Hello, A kamailio.org server maintenance work will be done today at 18:00 CEST. There will be some minutes of downtime (should be less than 5') of the following main services: website (including the wiki) and mailing lists. Regards, Oriol.

1 0

Kamailio does not receive anything after exit route
by Andrey Deykunov 25 May '18

25 May '18

Hi there, I've added a simple black list that rejects incoming requests (REGISTER and INVITE) containing certain text in the message body. Source IP addresses of the rejected requests are going to be banned and added to the ban list: route > { > ... > > > $var(ban_list_key) = $si + ":" + $sp; > if ($sht(ip_ban_list=>$var(ban_list_key)) != $null) { > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm request > from: $fU, si: $si, sp: $sp was rejected. The source host is temporarily > banned."); > sl_send_reply("400", "Bad request"); > exit; > } > > ... > > if (method=="REGISTER" || method=="INVITE") { > sht_iterator_start("i1", "black_list"); > while(sht_iterator_next("i1")) { > $var(term_is_in_wl) = 0; > $var(collision_idx) = 0; > while($sht(white_list=>$fU[$var(collision_idx)]) != $null > ) { > if($shtitval(i1) == > $sht(white_list=>$fU[$var(collision_idx)])) { > $var(term_is_in_wl) = 1; > break; > } > $var(collision_idx) = $var(collision_idx) + 1; > } > if ($var(term_is_in_wl) == 0) { > if ( $mb =~ $shtitval(i1) ) { > # found a term from the black list. access denied. > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm > request from: $fU, si: $si, sp: $sp was rejected. SIP message body contains > a term from the black list - $shtitval(i1)"); > # add device to banlist > $var(ban_info) = "" + $Ts + "_" + $shtitval(i1); > $sht(ip_ban_list=>$var(ban_list_key)) = > $var(ban_info); > xlog("L_INFO", "$var(log_fmt_tag) The source host > $si:$sp was banned for 1 hour."); > sht_iterator_end("i1"); > sl_send_reply("400", "Bad request"); > exit; > } > } > } > sht_iterator_end("i1"); > } > ... > } All works fine with UPD, but using TCP, kamailio ceases to accept any requests. Please take a look, what do I wrong. Thanks, Andrey

2 1

Performance Testing Kamailio 5.1 as SIP300 LCR on AWS - Recommendation on Tools and Methods
by KamDev Essa 24 May '18

24 May '18

I am currently setup with Kamailio 5.1 on an AWS t2.micro as a SIP300 LCR solution with a http_query call to an external resource (c# .NetCore app for LRN dip and MySQL Route choices) to get the contacts string, tag it to the Contact header an send 300. Its been a while since I have performance tested on SIP and It was SIPp. - Is http_query a bad choice for performance. Is KEMI better? - Are there any better free apps than SIPp? - Has anyone got cps stats for a SIP300 only Kamailio instance on AWS - Is AWS a bad choice for prime time LCR. Any gory stories? KD

2 3

SIP Server Hardware or VM ??
by Pinter, Gerd. 24 May '18

24 May '18

Hallo there, we are a Broadcasting Network with about 50 Stations and try to establish our own SIP Network for high quality Audio over IP with Bitrates of 192-256 kbit for transmitting an accurate stereo signal. Since we encountered problems with connections via mobile Networks we set up our SIP Server as RTP Proxy which resolved all those multiple NAT issues. But on the other hand we pay for that with a higher jitter rate. The question is, if anyone has experience if this issue maybe can be solved by running the Sip Server on a standalone hardware plugged into network Port on our Hardware Firewall, instead of a virtual Machine running on VM with virtual switches and so on. The current SIP Server has a Public IP Address, but all the Traffic runs thru Firewall, VPN, Hardware Switches and Virtual Switches. What you think? Thanks a Lot Nice regards Gerhard Pinter

4 5

count of registered users or count of tls connections
by Karthik Srinivasan 24 May '18

24 May '18

Hello, Via the routing file i am trying to find access two items:

2 5

Core Value : myself
by Duarte Rocha 23 May '18

23 May '18

Greetings, I'm doing some work with the AUTH routes and modules, and sometimes i get a "403 Not relaying" answer when the INVITE has a parameter in the R-URI. That parameter was not included in the REGISTER. I think it's this part of the code that throws this reply : # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } Does a different or new parameter in the R-URI trigger the "uri!=myself" ? if so, can I set it up to check R-URI but not check the parameters? Thanks

2 1

Any modules with LRN dipping capacity?
by KamDev Essa 22 May '18

22 May '18

Are there any modules with LRN dipping capacity?. If not, whats the best way we can implement it as a bolt on. KD

3 14

Best way to implement custom LCR
by KamDev Essa 22 May '18

22 May '18

I apologize for asking unit questions today and decided to just present the big picture and address what I am trying to get @ holistically. I want to implement a custom LCR only solution on kamailio. INVITE in 300 out. Nothing else. I expect awesome cps stats that makes we lean towards kamailio rather than throwing up a home grown SIP server and worrying about performance. I have the guts of the LCR logic down pat, but todays its manual. In my process I do the following : - Check LRN of the requested user. I use dig doing a ENUM call. - Use the requesting user and LRN of requested user to mediate call type. Local or Interstate using LRN check via ENUM call (above) and LCAD data lookup. I use sql query. - Mediate call type Intrastate or International if not Local or Interstate using LERG lookup. SQL query too. - dip into database with requesting user and call type and get N lists by costs ascending. SQL query. - Construct Contact info in db or kamailio. - Send 300 reply will contact having the various routes. I can automate 2 through 6 using Kamailio. Basically xquery with supplied data and deal with the Contact header with data received. 404 or default route if no data received. I cannot handle the ENUM in kamailio or mysql. The provider does not provide SIP redirect. What are my options, besides changing LRN providers? KD

3 11

300 Multiple Choices response engine
by KamDev Essa 22 May '18

22 May '18

Whats the best route to implement a basic 300 Multiple Choices response engine. INVITE in 300 Multiple Choices out with choices in the response. I have a logic to generate my list choices, so a basic process would be better. H

2 10

2 pull requests: issue #1168 and another small issue
by Valentin Christoph 22 May '18

22 May '18

Hi all, Issued two pull requests. Any comment welcome, in particular, whether it will be taken for master branch...... https://github.com/kamailio/kamailio/pull/1538 https://github.com/kamailio/kamailio/pull/1539 Kr Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk
by Wilkins, Steve 22 May '18

22 May '18

Hello All, When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP Address that Kamailio placed in the Record-Routes so... Is there a way to force Kamailio to set the Record-Route to its Private IP address of the first Record-Route in messages forwarded to Asterisk? Thank you?

2 8

rtpp_test() in kamailio log
by Giovanni Tommasini - evosip 22 May '18

22 May '18

Hi guys, I've a question on using rtpengine with kamailio *scenario*: kamailio istance with rtpengine enabled and configured in localhost (but with a remote rtpengine is the same) modparam("rtpengine", "rtpengine_sock","udp:localhost:22222") kamailio version 4.4.6 + rtpengine version 6.3.0.0 running on Ubuntu 16.04 when I start kamailio I see a lot of INFO log rows like that May 22 12:13:12 my-dev /usr/sbin/kamailio[3388]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled May 22 12:13:12 my-dev /usr/sbin/kamailio[3397]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled ... more to the point. I see always 24 rows for each rtpengine_sock configured i n rtpengine.c I find the rtpp_tes function and that's fine, but why 24 times? thanks a lot -- Giovanni Tommasini | evosip web: evosip.cloud

2 2

Kamailio sends SIP messages to old websocket after reconnection during call
by Idris AVCI 22 May '18

22 May '18

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the dropped one but kamailio tries to send to old websocket port which is not active anymore. We use GRUU and configured all the related parameters in registrar, dialog, tm, websocket, tls, xhttp and userloc modules both for versions 4.4 and 5.1 but couldn't find a solution yet. For a typical call below are the ordered events: 1. UA register to kamailio over wss (Code: WS1) 2. Kamailio send an INVITE to the UA (Code: Call1) 3. UA answer and media starts 4. UA drops WS1 to kamailio and reconnect (Code: WS2) 5. UA has an active websocket (WS2) to kamailio 6. Kamailio tries to send BYE message of Call1 to WS1 which is no longer active and complains that websocket could not be found 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2 8. UA answer and media starts 9. Kamailio send BYE message of Call2 to UA over WS2 and everything work as expected (No websocket drop) Error log: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found I think we need a mechanism that will update the dialog information of Call1 to use WS2 after reconnection. Any help greatly appreciated. Thanks.

2 2

htable - unable to update entry when event_route[htable:expired:<table>] is executed
by David Heath 22 May '18

22 May '18

Hi, Looking for some help/ advice on updating the expiration timer/ values in a hash table when when [htable:expired:<table>] occurs. On startup my script successfully reads in a set of entries from a DB; I also know from testing/ tracing that I can add new entries to the hash table in request_route. My issue occurs when the expiration timer kicks in - my goal was to re-read from the DB to ensure either the entry still exists or hasn’t changed and either delete, update or refresh the value until the next time the expiration event occurs. This way entries in the DB could be modified overtime. At the moment all that’s occurring is that in the event_route[htable:expired:cli] block the entries are being drained one-by-one. I re-read the row from the from the DB and I reset the value using $sht(cli=>$shtrecord(key)) = $xavp(cli=>MAPPED_CLI); I’ve also tried updating the expiration timer against the key. I’ve tried both $shtex(cli=>$shtrecord(key)) = 60 and $shtex(cli=>$shtrecord(key)) = $TS+60 [the latter based on the values shown from sht_print()] but the entries still get deleted. I’ve tried this against both 5.1.0 & 5.1.2. Any help appreciated. Thanks David Heath

1 0

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