- sr-users - kamailio.org

How to get Formatted Current TIME (NOT CACHED)
by Alvaro Araos 31 Jul '18

31 Jul '18

Hi, If i call to $timef multiple times, always returns the same time string (CACHED), how i get formatted current time?? Example: $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS "2018-07-28 18:44:35" .. some seconds later $dlg_var(start)=$timef(%Y-%m-%d %H:%M:%S) // RETURNS THE SAME DATE STRING "2018-07-28 18:44:35" Saludos, ALVARO ARAOS MÉNDEZ Director de Tecnología en FONOIP +56 2 2947 9000 alvaro(a)fonoip.com | www.fonoip.com [1] Moisés Chacón 46, El Monte, RM Links: ------ [1] http://www.fonoip.com

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RTP Proxy
by Alex Tatham 31 Jul '18

31 Jul '18

Hello everyone, I'm in the process of moving approx. 100 Kamailio/Media Proxy (AG Projects) servers to Kamailio/RTPEngine. I have all the functionality in place and I have a handful of servers in production. I'm trying to monitor the RTP engine (using Munin) and I'm graphing the values returned from the RPC command rtpengine.get_hash_total. This value is slowly increasing over time and I'm trying to find out if this is something I should worry about. I want to make sure that sessions are not being left open and therefore the software will eventually run out of ports. To give some values, if I make 100 calls the call will be around 25 after those calls. >From all other tests I have done including a check through the verbose logs I can see the sessions are being closed at the end of the calls. Can anyone help with this at all or does anyone possibly have any better ideas? Thank you, Alex Alex Tatham Technical Director T:01233 220 943 E:alex.tatham@dmcplc.co.uk W:www.dmctechnologies.co.uk DMC Technologies formerly HP Technology Unit 5, Invicta Business Centre | Orbital Park | Ashford | Kent | TN24 0HB DMC Technologies is a trading name of Hobbs Parker 2000 Limited Registered Office: Romney House, Orbital Park, Monument Way, Ashford TN24 0HB Registered in England No.04027755

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Is anyone having issues with dialog metrics when DMQ is enabled?
by Joel Serrano 30 Jul '18

30 Jul '18

Hi guys, Is anyone having issues gathering active/early dialog metrics from kamailio when DMQ is enabled? I have an open ticket in GH ( https://github.com/kamailio/kamailio/issues/1591) and I'm wondering if this could be happening to anyone else? So far the theory is that the counter goes below 0 and that creates the problem. I'd love to hear from you if you are experiencing this too. If any of you are using Kamailio 5.1 + DMQ and gather active/early dialog metrics using the stats.fetch or stats.get_statistics rpc commands, do you ever get values close to 64BIT ULONG_MAX? (18446744073709551615)?? Thanks! Joel.

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Kamailio 5.0.X RPM packages/repo broken
by Patrick Wakano 27 Jul '18

27 Jul '18

Hello list, Hope you all doing fine! I've been trying to install Kamailio packages from the opensuse repo (as per https://www.kamailio.org/wiki/packages/rpms) but the installation is failing with this message: Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-ims-5.0.7-1.el7.cento FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno 14] HTTP Error 416 - Requested Range Not Satisfiable --:--:-- ETA Trying other mirror. kamailio-5.0.7-1.el7.centos.x8 FAILED 77% [============================================================- ] 1.0 MB/s | 4.9 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.0.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.0.x-rpms clean metadata Trying other mirror. I've successfully installed the 5.0.7 version on Friday (06/07) but I've noticed the packages have changed last Saturday (07/07/2018).... Given this error message looks like the repository is broken somehow.... Is anyone aware of such issue? I haven't tried the 5.1.X repo because my script was not migrated to 5.1.X yet.... Thanks, Kind regards, Patrick Wakano

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NGN and NNI functionalities
by Duarte Rocha 27 Jul '18

27 Jul '18

Greetings, Does Kamailio has any module that deals with NGN and NNI ? Also, is there any variable or transformation that gives me access to a parameter in the User Part of an URI ? Best Regards, Duarte Rocha

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No cdr after Kamailio 5.0.4 and 5.2 dev restart
by Kordován Szabolcs 27 Jul '18

27 Jul '18

Hi All, In March there was a discus about this issue. We tested the new 5.2 dev and 5.0.4 version of Kamailio and we experience that If we restart Kamailio during the call the cdr won't be stored in table acc_cdrs but the data will be inserted to table acc and data will be deleted from table dialog and dialog_vars. Did I omit the fix for it? Thanks for your help. Best regards, Szabolcs

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Package hashes for Kamailio 5.1.4 rpms are wrong.
by Skip Morse 26 Jul '18

26 Jul '18

Hi, The package hashes for Kamailio 5.1.4 rpms are wrong. I actually saw this happen with 5.1.3 and then somehow I think it got corrected. Repo details: http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.1.x-rpms/C… Filename: kamailio-5.1.4-2.el7.centos.x86_64.rpm Path: /repositories/home:/kamailio:/v5.1.x-rpms/CentOS_7/x86_64/kamailio-5.1.4-2.el7.centos.x86_64.rpm Size: 5.4M (5645948 bytes) Last modified: Thu, 19 Jul 2018 02:53:15 GMT (Unix time: 1531968795) SHA-256 Hash: aed812fa3ccf502634fe6de848eca2ce167ee70f6ace31aa1541a5787502667d SHA-1 Hash: ae0977ca32e6b5ba2a627e58948098f386639582 MD5 Hash: 68784949f33d6259420b50de5e70a4ec BitTorrent Information Hash: f743a7c45d50065084d40a314bd80e3ec143c384 What I get locally: $ wget http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… $ sha256sum kamailio-5.1.4-2.el7.centos.x86_64.rpm a21d2b6c98da041af85ebc0f8b6fe8525e8e247e84317c32db021e90ec43b209 kamailio-5.1.4-2.el7.centos.x86_64.rpm $ sha1sum kamailio-5.1.4-2.el7.centos.x86_64.rpm 9cbf6de153fb8ffc4913f0582d6e85535b44975f kamailio-5.1.4-2.el7.centos.x86_64.rpm $ md5sum kamailio-5.1.4-2.el7.centos.x86_64.rpm d36149f4bd91d0879c239b8e81264fb3 kamailio-5.1.4-2.el7.centos.x86_64.rpm Let me know if I should report this elsewhere. Thanks! -Skip

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DMQ with dialog: KDMQ messages being looped back to the source
by Patrick Murphy 26 Jul '18

26 Jul '18

Hi, I am using Kamailio 5.1.1 with DMQ with the following config (relevant parts) on 2 nodes. Node A: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.a.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) Node B: modparam("dialog", "enable_dmq", 1) modparam("dmq", "server_address", "sip:no.de.b.ip:5060") modparam("dmq", "notification_address", "sip:myvoip.local") modparam("dmq", "multi_notify", 1) "myvoip.local" is a DNS A record that resolves to both the IPs of A and B. I have the dmq_handle_message(); called if method is of type "KDMQ" in the request_route. Now, in order to test it, I only sent a few calls only on node A. KDMQ messages flowed to node B and dialog state replicated. But I noticed, node B was randomly sending one of the KDMQ dialog messages back to A and receiving a 500 "Server Internal Error" back from node A... (for a subset of those calls, not all). Why would this be the case when the nodes correctly recognize their local status by the "local: 1" set as seen from dmq.list_nodes Anyone else experienced this? What could I be doing wrong? Thanks, Pat.

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FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service, ClueCon Presentation Slides
by Giovanni Maruzzelli 25 Jul '18

25 Jul '18

Hello fellow RTCers, just finished presenting about scaling videoconferencing, chat and moderation/direction controls in SIP, with web clients, deskphones, smartphone apps, server push (google fcm) via cordova and SIP.js at www.cluecon.com. Here's the slides of my presentation: http://178.79.181.125/Maruzzelli_FreeSWITCH_OpenSIPS_WebRTC_Pure_SIP_Video_… or, shorter, http://178.79.181.125/Maruzzelli_ClueCon_2018.pdf >From ClueCon blurb: FreeSWITCH and OpenSIPS for a Pure SIP Video, Chat and Conference Service On top of traditional SIP deskphones and softphones, WebRTC let us extend the reach of SIP to all browsers, and to smartphone apps that wake up with a server push. OpenSIPS has superb SIP and WebRTC support and can protect, balance and scale FreeSWITCH insuperable conferencing, video MCU, media switching/mixing, SIP SIMPLE message processing capabilities. We will see how to build and implement such a complete platform, touching both on servers and on clients. Hint: SIMPLE, the SIP Instant Messaging Protocol, can be leveraged to transport statuses, commands and chats between participants and administrators of SIP audio/video calls/conferences. Giovanni MaruzzelliPrincipal at OpenTelecom.IT ------------------------------ Giovanni is a consultant for the Telco sector, developing software and training courses for FreeSWITCH, SIP, WebRTC and Kamailio. He is heavily engaged with FreeSWITCH, of which he wrote the interface with Skype and with cellular phones. An Internet tech pioneer, in 1996 Giovanni was cofounder of Italia Online, the most popular Italian portal and consumer ISP, and architect of its Internet technologies – www.italiaonline.it Then supervisor of Internet operations and architect of the first engine for paid access to www.ilsole24ore.com , the most read financial newspaper in Italy and to its databases (migrated from mainframe). After that, he was CEO of venture capital funded Matrice, developing Telemail unified messaging and multi language phone access to email (Text To Speech), and CTO of incubator funded Open4, an open source managed applications provider. Then he was for two years in Serbia as Internet and Telecommunication Investment Expert for World Bank – IFC. Since 2005 he’s based in Italy, and serves ICT and Telco companies worldwide. -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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sipdump excluding dmq exchanges
by Patrick Murphy 24 Jul '18

24 Jul '18

Hi Can we have sipdump only log SIP messages and not KDMQ? Thanks, Pat.

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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.

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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.

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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

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