- sr-users - kamailio.org

Kamailio sending response to source IP and port from Via ?

by Chaigneau, Nicolas

Hello, I'm using Kamailio in stateless mode to send redirect responses. I've noticed that the responses are sent to the packet source IP address, and to the port found in the Via header. I'm a bit confused. I was expecting the response to be sent to the address and port found in the Via header. Is it the expected behaviour ? I'm using Kamailio 5.5.2. Thanks for your help ! Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 year, 11 months

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Issue with debug 3 after upgrade to kamailio 5.6.2+

by Julia Bo

Hello, After upgrading kamailio 5.6.1 to 5.6.2 (from source) debug 3 doesn't print messages related to calls to the log file. I only see OPTIONS transactions in debug. Does anyone have the same problem? Best regards, Julia.

1 year, 11 months

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About the WITH_VOICEMAIL Subscriber message 407 issue

by SparkleZou

Hi Sir/Madam, I'm trying to set a Fresswitch as the Voicemail server. I set the #!define WITH_VOICEMAIL in the kamailio.cfg file to enable the default VoiceMail configuration. Set the voicemail server address as following : voicemail.srv_ip = "10.120.100.179" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" And the kamailio server address is 10.120.100.180. Here face the issue, the Subscribe message could NOT be authorized. The SUBSCRIBER message with token could NOT be forworded to Freeswitch. The log is also attached. It's resllay strange. Please help check. Thanks! BR, Sparkle Zou

1 year, 11 months

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ims_auth with AKAv1-MD5

by Oleg Belousov

Hello, I'm wondering if ims_auth is tested with actual SIM-card/AKAv1-MD5 authentication? Bit concerned with that since as per authorize.c source code can see for AUTH_AKAV1_MD5/AUTH_AKAV2_MD5/AUTH_MD5 response is counted in the same way, though algorithms are different. -- obelousov.tel

1 year, 11 months

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No REGISTER sent to third party AS from scscf

by Anthony Blandin

Hi Community, I am using scscf from Kamailio and I need that scscf sent a REGISTER to a third party AS during the registration phase. When scscf received a new registration, no REGISTER is sent to the AS. SAR-SAA is well processed but only 200 OK is sent for the REGISTER to the pcscf. REGISTER is well sent to the AS only for Re-Registration or De-registration. Do you have any idea to handle this case? Regards Anthony

1 year, 11 months

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Dialog Module: How to get two different dialogues if a call is looped through kamailo (call forwarding scenario)

by Benoit Panizzon

Hi Using the dialog module to generate CDR for billing. We have this situation: Kamailio 'routing' instance, where dialog is used and billing happens. Kamailio 'register' instance. IC is connected to the routing instance. Customers register to the register instance. Call originates from 'A' via IC through routing instance to register instance and rings customer 'B'. This times out and 'call forwarding' is engaged by sending the call back to the routing instance with a new RURI containing the call forwarding destination 'C'. An appropriate 'no-answer' Diversion Header is added. Now we need two different CDR for this scenario (add more CDR if the call is forwarded multiple times) One from the A to B leg to bill termination fee to the IC carrier. One from B to C to bill the forwarded call with destination 'C' to customer B. The issue I now face is that I only get the kamailio routing instance to generates one dialogue from B to C despite handling two separate connections. How could that challenge be solved? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

1 year, 11 months

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relay sip-tls vs sips-tcp

by beer Ll

Hi everyone, I'm using Kamailio as TLS gateway/filter for an internal Asterisk server the network schema is : UAC (tls) --- INTERNET --- (tls) KAMAILIO (sip udp) --- LAN --- (sip udp) ASTERISK with kamailio in multi-homed mode WAN network interface for sip tls LAN network interface for sip udp to asterisk server UAC address 80.0.0.1 KAMAILIO Wan address 80.0.0.2 KAMAILIO Lan address 172.16.0.2 ASTERISK Lan address 172.16.0.3 SIP-TLS call example If the UAC use tls(sip) all works good [image: sip-ok-small.jpeg] SIPS call example If the same UAC use his default settings tls(sips) , there are problems with ACK and BYE packet [image: sip-ko-small.jpeg] the SIP OK SDP packet from kamailio to UAC is 2022/10/10 09:28:47.854721 80.0.0.2:5061 -> 80.0.0.1:49992 SIP/2.0 200 OK Via: SIP/2.0/TLS 192.168.0.1:49992 ;rport=49992;received=80.0.0.1;branch=z9hG4bKM01j360VrBdH5VSV Record-Route: <sip:172.16.0.1:5060 ;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Record-Route: <sip:80.0.0.2:5061 ;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7 From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 CSeq: 2 INVITE Contact: <sip:172.16.0.2:5060> P-Asserted-Identity: "xxxxxxxxx" <sips:*43@pbx.voip.com> Content-Type: application/sdp and the UAC send the ACK and BYE from a different tcp port and to: sips:172.16.0.2:5060;transport=tcp 2022/10/10 09:28:48.495365 80.0.0.1:49996 -> 80.0.0.2:5061 ACK sips:172.16.0.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TLS 192.168.0.1:49996;branch=z9hG4bKppftdQze20lnwT41;rport Route: <sip:80.0.0.2:5061 ;transport=tls;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Route: <sip:172.16.0.1:5060;lr;r2=on;ftag=F798336AA08EF9FCFA89D3BDFE0C8C8F> Max-Forwards: 70 To: <sips:*43@pbx.voip.com>;tag=961d0e22-a4f0-453c-9870-6a41578afc96 From: <sips:200@pbx.voip.com>;tag=F798336AA08EF9FCFA89D3BDFE0C8C8F Call-ID: 1EC2AB679C1EA1BAB60FD03B09F878020B12D3E7 CSeq: 2 ACK kamailio error log WARNING: <core> [core/forward.c:229]: get_send_socket2(): protocol/port mismatch (forced udp:172.16.0.2:5060, to tls:172.16.0.3:5060) How can I solve this ? Best Regards Leo

1 year, 11 months

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After changing instance type for kamailio server in aws log file is corrupted

by Sammy Dominguez

Hello guys, after changing instance type from t2 to t3a for kamailio server in aws log(/var/log/kamailio.log) file and some other files are corrupted and I can not touch,delete,create or anything and when kamailio start receiving traffic it crashed. Is there any particular reason for this? Basically on t2, they are under /dev/sda1 and /dev/xvda. Under t3 they are nvme disks (biggest part of the speed increase), so they are under /dev/nvme1n1. Note: once I rollback to t2 everything is working as expected.

1 year, 11 months

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Issue with duplicite SIP messages in traffic loop

by Michal Popovič

Hello, I am trying to configure kamailio to use external antifraud protection using so called "traffic loop" when call is forwarded to external SIP proxy that would response even with same INVITE back or with error message if number is fraud. schema: PBX -> kamailio -> traffic loop -> kamailio -> carrier Kamailio in second and fourth step is the same. Issue is that if kamailio receives same INVITE back to I would like to forward to next carrier it is possible, but kamailio struggles with transaction handling. If ACK message come from PBX, kamailio will correctly forward it to traffic loop but when it comes back, function t_check_trans is failed and I cant relay it. Only workaround is to use send() message, but that causes deny of BYE message from carrier. My question is following: Is it possible to register two different transactions states for same message? If yes how? If no, would it help to send and receive messages on another network port or not? Thank you, Michal

1 year, 11 months

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Recommended openSSL version

by Igor Olhovskiy

Hello! Due to I still experience irregular Kamailio 5.4 crashes (like 1/month) related to SSL (using websockets and SIPS) I'm wondering, could openSSL upgrade change the situation? As of now in CentOS 7 I have 1.0.2k version. Does anyone have experience to fix crash-related to TLS problems with openSSL upgrade? Or maye some tuneup of TCP parameters can help here?My current setup is quite simple: children=4 enable_tls=yes tcp_accept_no_cl=yes tcp_connection_lifetime=600 tcp_max_connections=998976 # 1000000 - 1024, so we're leaving 1k for system reserve tls_max_connections=998976 Number of clients ~ 200 constantly connected to websocket. -- Best regards, Igor

1 year, 11 months

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IMC module

by Giovanni Iamonte

HI, I am facing a problem related to the IMC module. I have installed kamailio 5.2.x and I tried to list the members of a room-qsipchat. In my test environment the kamailio.cfg file it is very basic, the kamailio server listen on tcp=192.168.100.192:15006 the client run on a PC with IP address 192.168.100.156 the imc outbound_proxy is set modparam("imc", "db_url", DBURL) modparam("imc", "imc_cmd_start_char", "#") modparam("imc", "outbound_proxy", "sip:sip-test.quintetto.it:15006;transport=tcp") The client start by sending a #join message to kamailio and looking the log file I can see that the room-qsipchat has been create, and also I received the message 200 OK. By using the tcpdump I can see the join message from the Client to kamailio server and the reply from kamailio to the client. When I send the #list message to the room-qsipchat , I expecting to receive the message relate to room's members and the 200 OK message. When I tcp dumping I saw the following messages ******************************************************************** Request from client to list the members - from client to server ******************************************************************** TX 538 bytes Request msg MESSAGE/cseq=57652 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Content-Type: text/plain Content-Length: 5 #list ******************************************************************** The server ask for the authentication - from server to client ******************************************************************** 2022-10-17 15:45:25.837 4700-10610/it.quintetto.qsipchat D/pjlib(pjsua_1): RX 574 bytes Response msg 407/MESSAGE/cseq=57652 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjRzgOBgY6vfKZ8Fh8GA4CWheCO840Xzud;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.e417 Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57652 MESSAGE Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************* The client send the authenticated request - from client to server ******************************************************************** TX 755 bytes Request msg MESSAGE/cseq=57653 (tdta0x77b26460a8) to TCP 192.168.100.192:15006: MESSAGE sip:room-qsipchat@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.156:48129;rport;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias Max-Forwards: 70 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it> Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Accept: text/plain, application/im-iscomposing+xml Route: <sip:sip-test.quintetto.it:15006;transport=tcp;lr> Proxy-Authorization: Digest username="quintetto00", realm="sip-test.quintetto.it", nonce="Y01doWNNXHVEyRtiAyN28xJdMjNIG8Ju", uri="sip:room-qsipchat@sip-test.quintetto.it", response="d495ec20cb1be8f1b605d4e5b1a4c357" Content-Type: text/plain Content-Length: 5 #list ************************************************************************** The server send the 200 OK message - from server to client ******************************************************************** RX 447 bytes Response msg 200/MESSAGE/cseq=57653 (rdata0x77b3337330) from TCP 192.168.100.192:15006: SIP/2.0 200 ok Via: SIP/2.0/TCP 192.168.100.156:48129;rport=48129;branch=z9hG4bKPjj6uuy25r0kKlnICHKQWEaIweOex-sNyq;alias;received=192.168.100.156 From: <sip:quintetto00@sip-test.quintetto.it>;tag=1is.Ng6ZKJmCT1uAJ9WuJhe3hiQysj9f To: <sip:room-qsipchat@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.365f Call-ID: GwOQMLFfIq7o7jvF4OQlJ-jBb2OBaAKY CSeq: 57653 MESSAGE Server: kamailio (5.2.8 (x86_64/linux)) Content-Length: 0 ********************************************************************************* Till here everything is OK. But regarding the members when a TCP dumping I saw a message with the room's members that go from the kamailio server to the kamailio server ********************************************************************************* MESSAGE sip:quintetto00@sip-test.quintetto.it SIP/2.0 Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it> From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Max-Forwards:.70 Content-Length: 47 User-Agent: kamailio (5.2.8 (x86_64/linux)) Content-Type: text/plain Members: *sip:quintetto00@sip-test.quintetto.it ************************************************************************************** a reply from the kamailio server to the kamailio server, with an authenticated request ************************************************************************************** SIP/2.0.407.Proxy Authentication.Required Via: SIP/2.0/TCP 192.168.100.192:15006;branch=z9hG4bKf20a.be253b92000000000000000000000000.0 To: <sip:quintetto00@sip-test.quintetto.it>;tag=1e5fbdd4b14eff95c400342f4634da91.cfc3 From: <sip:room-qsipchat@sip-test.quintetto.it>;tag=03e4c67b0f95c0aeb1f633479b4aa5cd-72f2 CSeq: 10 MESSAGE Call-ID: 37db44aa523ea24a-6567(a)192.168.100.192 Proxy-Authenticate: Digest realm="sip-test.quintetto.it", nonce="Y01fQmNNXhbsHdM1THmENyJn1MH/VD/8" Server: kamailio (5.2.8 (x86_64/linux)) Content-Length:.0 ************************************************************************************** bottom of line the outbound_proxy request for an authenticated request, but kamailio do not reply to this request. Could you please point me in the right direction? Thanks a lot for any help. Bye -- ------------------------------------------------------------------------ Ing. Giovanni Iamonte Developments and technologies area Quintetto Srl Via Monte Navale, 1 10015 - Ivrea (TO)  mobile: +39 393 9196310  tel: +39 0165 1845290  e-mail: giovanni.iamonte(a)quintetto.it  web: www.quintetto.it

1 year, 11 months

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ClueCon and ... FreeSWITCH 1.10.8/RHEL8 packages!

by Richard Chan

Hi sr-users, Since ClueCon is going on right now, and some Kamailio community members are involved I hope you will forgive a little promo here: I have a RHEL8 build of the just released 1.10.8 on Copr: https://copr.fedorainfracloud.org/coprs/beaveryoga/FreeSWITCH-1.10.8/ Hope you will find it useful in your kamailio + Class 5 scenarios! Cheers Shih-Ping

1 year, 11 months

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Testing dialog module (and acc with dialog cdrs)

by Daniel-Constantin Mierla

Hello, as I am not a user of dialog variables, I am turning to community to ask for help testing the current master branch with configurations that make use of dialog variables and acc dialog-based cdr generation. With a few reports of issues related to dialog modules and unexpected crashes, I looked over the code and noticed that the access of the value for dialog variables was not protected, making them vulnerable of invalid memory access in case of the variable was updated by another process or dialog was terminated. I introduced a couple of new functions to try to cover different use cases of getting the dlg variable values, dialog management code was not affected, but given that these commits need to be backported to stable branch (5.6), I want to get proper feedback from community that things work fine. A previous attempt of a simpler fix was not enough, having side effects to acc module for dialog-based cdr generation, because it was keeping referenced to many dlg variables at the same time. In short, it would be appreciated any feedback on testing dialog and acc with dialog-based cdr generation using git master branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 7-10, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 year, 11 months

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kamailio & rtpengine question

by Gerry Kernan

Hi Looking for advise if anyone could point out where im going wrong. My setup is Kamailio Ver 5.6.0 running in azure , rtpengine Version: git-master-dc395d5, behind kamailio I have asterisk ver 18.12.1 Kamailio conf I have interface set as below listen=udp:10.3.1.6:5060 advertise WANIP:5060 and rtpengine.conf set as below interface = eth0/10.3.1.5 but when I can make a call I don't get any audio, rtpengine.log extract below shows that rtp is going to the private IP rather that the Public IP of the phone that initiated the call, do I need to set rtpengine_manage() to fix this? Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Confirmed peer address as 10.3.1.5:12890 Oct 7 09:38:58 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30278]: [core] Kernelizing media stream: 10.3.1.5:12890 -> 10.3.1.6:30278 | 10.3.1.6:30298 -> 192.168.0.123:40042 Oct 7 09:39:00 kamailio-proxy rtpengine[8127]: INFO: [NmE0MjdhMzA4NjUzYWVjMmRjYzIxNGJjMTY4ZTc2MWE. port 30279]: [core] Confirmed peer address as 10.3.1.5:12891 Kind regards gerry Gerry Kernan | Technical Director Mobile: +353(0)86-170 9790 [Graphical user interface, website Description automatically generated]<https://www.infinityit.ie/> [A picture containing text, clipart Description automatically generated,Icon Description automatically generated]

1 year, 11 months

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tsilo usage more generally

by Alex Balashov

Since on the topic of tsilo: 1. I watched Federico’s presentation on it from 2015, and his examples insinuate that it is necessary to store the index and label of a suspended transaction in some external vehicle, e.g. htable, for use with ts_append_to(). Why would one do this when it is possible to call ts_append() based on R-URI alone? 2. The appropriate pattern of interaction between tsilo and http_async_client — which I imagine to be a rather common use-case — is rather unclear. If I use http_async_query() to call out to a PN service, with $http_req(suspend) = 1, what should happen to the current instance of SIP request processing? That is to say: request_route { . . . t_check_trans(); if(is_method(“REGISTER”)) { . . . save(“location”, “0x06”); if(ts_append(“location”, “$tu”)) { xlog(“success”); } return; } else if(is_method(“INVITE”)) { # PN stuff. ts_store(“$var(uri)”); http_async_query(“http://url”, “PUSH_REPLY”); return; # <— ??? } } route[HTTP_PUSH_REPLY] { . . . if(!t_relay()) sl_reply_error(); } I should add that this proxy front-ends a registrar, but is not in itself a registrar. It contains a shadow registrar - hence the save() calls - in order to be able to take advantage of the ’tsilo’ module, given its dependency on registrar/usrloc. This leads to the difference that the gateway receives requests with RURIs that have already been resolved to a contact binding elsewhere, so there are no lookup() calls. How should this play into the http_async_client workflow? Appreciate any help! — 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/

1 year, 11 months

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tsilo logging

by Alex Balashov

Hi, When there are no stored transactions for an AOR, it logs this error: Oct 13 07:48:18 gw1 /usr/sbin/kamailio[14758]: ERROR: tsilo [ts_append.c:64]: ts_append(): failed to retrieve record for sip:abalashov@sip.evaristesys.com But not having transactions ts_store()’d is a rather normal occurrence, right? It is the expected case the vast majority of the time that someone registers. Does that mean one is simply destined to have this scary-looking error, to the syslog ERROR facility, for a normal runtime condition? Is there not some principle against using attention-grabbing error messages this way? Would a patch to fix be welcome? — 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/

1 year, 11 months

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About kamailio + freeswitch issue on Kamailio v5.6

by SparkleZou

Hi Sir/Mada, I'm trying to integrate kamailio + freeswitch, according to the manual http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc on Kamailio v5.6 modified the kamailio.cfg file, add WITH_FREESWITCH parts, then kamailio could NOT start up. Could you please help check, what should be modified to fit the version 5.6？ Thanks！ BR， Sparkle Zou

1 year, 11 months

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Re: [SR-Users] play_media sounds interrupted

by Stepan Kislyakov

> The duration of the media playback is returned from the play_media() > command (see `media_duration` modparam). You could use some sort of > timer to invoke media unblocking after media playback ends. Sadly my > Kamailio-Fu isn't strong enough to be able to tell you how to run this > sort of timer. :) > > Cheers Thank you for the hint. I was thinking about timers, but rtimer cannot be triggered in an event route. I used a "timer" module, it could trigger another route from the event route but when a new route is triggered unblock_media() is not aware of the call-id and if i assign call-id to a variable in the event route, that variable is unavailable at new route. Hope that make sense. here is my peace of code: loadmodule "timer.so" modparam("timer", "declare_timer", "PB_TIMER=START,5000"); route[START]{ xlog("current call id at event route is $ci, but call-id avp is $avp(call_id)"); unblock_media("all"); timer_enable("PB_TIMER",0); #we need to disable timer or it will be triggered over and over again } event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); #block_media(); play_media("all file=/usr/local/src/myfile.wav"); $avp(call_id)=$ci; timer_enable("PB_TIMER",1); }

1 year, 11 months

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play_media sounds interrupted

by Stepan Kislyakov

Hi, I'm using latest kamailio 5.6.2 and rtpengine. I need to play announcement to both parties when call is answered by callee. Now I run play_media in event route but when audio start announcement sounds like we have a 50% packet loss. After it is completed there is no problem with audio both sides. I figured out that if i do block_media() my announcement play perfectly but i dont understand how to do unblock_audio() after announcement ends. Appriciate any help! here is my code: event_route[dialog:start]{ xlog("L_INFO", "dialog started, received 200OK"); block_media(); play_media("all file=/usr/local/src/myfile.mp3"); #unblock_media(); } -- Best Regards Stepan Kislovich

1 year, 12 months

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IPSec problem with P-CSCF

by Ali Akbar Hemmati

Hi everyone, I am using Kamailio as an IMS server. The UE first sends its SIP REGISTER over TCP and gets a 401 in response. Then the UE starts sending its packets over ESP. These ESP packets reach P-CSCF but are not delivered to its process by the kernel. We can see ESP packets in P-CSCF using wireshark. The ports used are correct (both client and server) based on their prior negotiations. We have also checked xfrm policies using the command `ip xfrm policy list` and seems right regarding prior negotiations. Our Linux kernel version is 5.4 and the OS is Ubuntu 20.04. Has anyone any clue of the root cause of this problem? Any ideas are appreciated. Thanks, Aliakbar

1 year, 12 months

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kamcli folder

by Volker Denneberg

hello, I'm not very familiar with PHP, so this may be a stupid question. I'm following https://github.com/kamailio/kamcli to install kamcli. Now i am at $ pip3 install -r requirements/requirements.txt which i should "run inside kamcli folder". So far, I don't have a kamcli folder, so i assume i have to create it. Any recommendations where ? Thanks in advance, Volker

1 year, 12 months

3
3
0 0

detect tcp connection close using tcpops module

by sthustfo

I am running latest kamailio 5.6.2 and attempting to detect tcp/tls connection being disconnected or closed using tcpops module. As suggested in tcpops documentation at https://www.kamailio.org/docs/modules/devel/modules/tcpops.html, registered event routes as below. event_route[tcp:closed] { xlog("L_INFO", "$proto connection closed ($conid)\n"); } event_route[tcp:timeout] { xlog("L_INFO", "$proto connection timed out ($conid)\n"); } event_route[tcp:reset] { xlog("L_INFO", "$proto connection reset by peer ($conid)\n"); } Now it is observed that the event route is being called but the $conid is not set. The logs show the below log statements and extra data is (nil). 24(4263) DEBUG: <core> [core/tcp_read.c:284]: tcp_read_data(): EOF on connection 0x7ffff2ff8b90 (state: 0, flags: 4018) - FD 7, bytes 0, rd-flags 10000 ([10.8.137.33]:51069 -> [10.8.137.33]:5060)24(4263) DEBUG: <core> [core/tcp_read.c:1500]: tcp_read_req(): EOF 24(4263) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x555555ab6960, 7, -1, 0x10) fd_no=2 called 24(4263) DEBUG: <core> [core/tcp_read.c:1870]: handle_io(): removing from list 0x7ffff2ff8b90 id 1 fd 7, state 2, flags 4018, main fd 51, refcnt 2 ([10.8.137.33]:51069 -> [10.8.137.33]:5060) 24(4263) DEBUG: <core> [core/tcp_read.c:1653]: release_tcpconn(): releasing con 0x7ffff2ff8b90, state -1, fd=7, id=1 ([10.8.137.33]:51069 -> [10.8.137.33]:5060) 24(4263) DEBUG: <core> [core/tcp_read.c:1657]: release_tcpconn(): extra_data (nil) 32(4271) DEBUG: <core> [core/tcp_main.c:3651]: handle_tcp_child(): reader response= 7ffff2ff8b90, -1 from 0 32(4271) DEBUG: <core> [core/tcp_main.c:3576]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) 32(4271) DEBUG: tcpops [tcpops.c:295]: tcpops_handle_tcp_closed(): received TCP closed event 32(4271) DEBUG: tcpops [tcpops.c:249]: tcpops_tcp_closed_run_route(): event reason id: 0 rt: 1 32(4271) DEBUG: <core> [core/parser/parse_fline.c:249]: parse_first_line(): first line type 1 (request) flags 1 32(4271) DEBUG: <core> [core/parser/msg_parser.c:679]: parse_msg(): SIP Request: 32(4271) DEBUG: <core> [core/parser/msg_parser.c:681]: parse_msg(): method: <OPTIONS> 32(4271) DEBUG: <core> [core/parser/msg_parser.c:683]: parse_msg(): uri: <sip:you@kamailio.org> 32(4271) DEBUG: <core> [core/parser/msg_parser.c:685]: parse_msg(): version: <SIP/2.0> 32(4271) DEBUG: <core> [core/parser/parse_hname2.c:301]: parse_sip_header_name(): parsed header name [Via] type 1 32(4271) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=2 32(4271) DEBUG: <core> [core/parser/msg_parser.c:555]: parse_headers(): Via found, flags=2 32(4271) DEBUG: <core> [core/parser/msg_parser.c:557]: parse_headers(): this is the first via 32(4271) exec: *** cfgtrace:dbg_cfg_trace(): unknown_route=[tcp:closed] c=[/usr/local/etc/kamailio/kamailio.cfg] l=976 a=26 n=xlog32(4271) INFO: <script> tcp connection closed (<null>) Can anyone provide any pointers on what I am missing? Thanks in advance.

1 year, 12 months

2
3
0 0

suppress t1 timer retransmissions?

by Brooks Bridges

So I have a very odd situation that requires me to suppress retransmissions when using t_reply() to send a response back to a UAC that is horribly broken but I'm stuck with for the time being. In the previous iteration of this platform it was fine since I was using a stateless reply, so even though it wasn't "right" it still worked out. I need to replicate this behavior in the event route for http_async_client(), which prevents me from being able to use t_release(). Anyone have any clever ideas? Brooks Bridges 1200 S. Rogers Circle, Unit 6 Boca Raton, FL 33487 Office: 916-235-2097 Main: 888-444-1111 email: bbridges(a)call48.com | web: www.call48.com Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

1 year, 12 months

2
4
0 0

cannot register ACC CDR JSON engine when using acc_json module

by Alexis Prodhomme

Hi everyone, I have some troubles using acc_json module. I can't load it and I don't understand why it fails. The configuration is quite basic: Debian 10 and Kamailio (fails in 5.5 and 5.6) installed using APT repository. Here is parts of my config: loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "dialog.so" loadmodule "acc.so" loadmodule "acc_json.so" modparam("acc_json", "cdr_enable", 1) I also tried with those 4 additionnal settings with no success: modparam("acc_json", "cdr_output_syslog", 1) modparam("acc_json", "cdr_log_facility", "LOG_LOCAL3") modparam("acc_json", "cdr_enable", 1) modparam("acc_json", "cdr_expired_dlg_enable", 1) As soon as I set cdr_enable to 1, it displays the "*cannot register ACC CDR JSON engine*". Here is logs: 0(2488) DEBUG: <core> [core/sr_module.c:969]: init_mod(): acc_json 0(2488) DEBUG: <core> [core/sr_module.c:722]: find_mod_export_record(): found export of <bind_acc> in module acc [/usr/lib/x86_64-linux-gnu/kamailio/modules/acc.so] 0(2488) INFO: acc_json [acc_json_mod.c:159]: mod_init(): janson version : 2.12 0(2488) DEBUG: <core> [core/mem/q_malloc.c:374]: qm_malloc(): qm_malloc(0x7f0a6b154010, 56) called from acc: acc_mod.c: acc_register_engine(725) 0(2488) DEBUG: <core> [core/mem/q_malloc.c:419]: qm_malloc(): qm_malloc(0x7f0a6b154010, 56) returns address 0x7f0a6b220b40 frag. 0x7f0a6b220b08 (size=56) on 1 -th hit 0(2488) DEBUG: acc_json [acc_json_mod.c:269]: acc_json_init(): init ... 0(2488) DEBUG: acc [acc_mod.c:741]: acc_register_engine(): new acc engine registered: json 0(2488) DEBUG: <core> [core/sr_module.c:722]: find_mod_export_record(): found export of <load_dlg> in module dialog [/usr/lib/x86_64-linux-gnu/kamailio/modules/dialog.so] 0(2488) ERROR: acc_json [acc_json_mod.c:245]: mod_init(): cannot register ACC CDR JSON engine 0(2488) ERROR: <core> [core/sr_module.c:974]: init_mod(): Error while initializing module acc_json (/usr/lib/x86_64-linux-gnu/kamailio/modules/acc_json.so) ERROR: error while initializing modules May it be related to the Jansson version ? I use the native one from Debian 10, v2.12. Do you have any advice, or some tips to debug my issue ? I don't really want to use acc module and parse CDRs manually from syslog, json is far easiest for this job. Thank you -- Alexis R&D Sewan

1 year, 12 months

2
2
0 0

Re: [SR-Users] Mails received twice in my Inbox

by Ginhoux, Patrick

Apparently, the emails are received on the fr unisys dot com address. So maybe there is a need to clean the sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> and sr-users-bounces(a)lists.kamailio.org <mailto:sr-users-bounces@lists.kamailio.org> list. If this is the case, thanks to keep only the fr unisys dot com. Cordialement. Patrick GINHOUX Unisys | +33 1 46 69 52 12 | +33 6 60 32 24 74 | patrick.ginhoux(a)unisys.com De : Henning Westerholt <hw(a)gilawa.com> Envoyé : lundi 10 octobre 2022 09:32 À : Ginhoux, Patrick <patrick.ginhoux(a)fr.unisys.com> Objet : RE: Mails received twice in my Inbox EXTERNAL EMAIL - Be cautious of all links and attachments. Hello, Ive checked last week directly on the list server. Maybe one subscription is old and and does not send mails anymore, and the mails are doubled from your side? Just guessing obviously, have not checked it yet. You should be able to check the email headers to see which address is actually used and maybe some forwarding is done from your side. If you like I can remove one subscription, let me know which one. Cheers, Henning From: Ginhoux, Patrick <patrick.ginhoux(a)fr.unisys.com <mailto:patrick.ginhoux@fr.unisys.com> > Sent: Monday, October 10, 2022 8:39 AM To: Henning Westerholt <hw(a)gilawa.com <mailto:hw@gilawa.com> > Subject: RE: Mails received twice in my Inbox Im back again. After having removed the address with fr unisys dot com, I received these 2 emails on my last one for approval : And later, still 2 copies of each emails : Same for yesterday : Except you see something wrong for the address, Im going to investigate internally. Cordialement. Patrick GINHOUX De : Ginhoux, Patrick Envoyé : jeudi 6 octobre 2022 14:10 À : Henning Westerholt <hw(a)gilawa.com <mailto:hw@gilawa.com> > Cc : Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > Objet : RE: Mails received twice in my Inbox Thanks for the list. I have now removed the one with fr unisys dot com and will wait for new emails on the other address. But it is still strange this problem happened recently. Cordialement. Patrick GINHOUX Unisys | +33 1 46 69 52 12 | +33 6 60 32 24 74 | <mailto:patrick.ginhoux@unisys.com> patrick.ginhoux(a)unisys.com De : Henning Westerholt <hw(a)gilawa.com <mailto:hw@gilawa.com> > Envoyé : jeudi 6 octobre 2022 13:32 À : Ginhoux, Patrick <patrick.ginhoux(a)fr.unisys.com <mailto:patrick.ginhoux@fr.unisys.com> > Cc : Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > Objet : RE: Mails received twice in my Inbox EXTERNAL EMAIL - Be cautious of all links and attachments. Hello, you are subscribed with two addresses, only list the domains below: at fr unisys dot com at unisys dot com You can change your subscription with the URL: <https://lists.kamailio.org> https://lists.kamailio.org Cheers, Henning From: Ginhoux, Patrick < <mailto:patrick.ginhoux@fr.unisys.com> patrick.ginhoux(a)fr.unisys.com> Sent: Thursday, October 6, 2022 1:25 PM To: Henning Westerholt < <mailto:hw@gilawa.com> hw(a)gilawa.com> Cc: Kamailio (SER) - Users Mailing List < <mailto:sr-users@lists.kamailio.org> sr-users(a)lists.kamailio.org> Subject: RE: Mails received twice in my Inbox Hi, Im sure Im not registered with different account, but who can check that in the mailing list to be sure ? Actually, I registered in 2017 and this issue occurs recently but I cant tell exactly since when. Cordialement. Patrick GINHOUX Unisys | +33 1 46 69 52 12 | +33 6 60 32 24 74 | <mailto:patrick.ginhoux@unisys.com> patrick.ginhoux(a)unisys.com De : Henning Westerholt <hw(a)gilawa.com <mailto:hw@gilawa.com> > Envoyé : jeudi 6 octobre 2022 11:06 À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> > Cc : Ginhoux, Patrick <patrick.ginhoux(a)fr.unisys.com <mailto:patrick.ginhoux@fr.unisys.com> > Objet : RE: Mails received twice in my Inbox EXTERNAL EMAIL - Be cautious of all links and attachments. Hello, Strange issue. Stupid question, but you are not registered two times to the mailing list, maybe with different accounts? Cheers, Henning -- Henning Westerholt <https://skalatan.de/blog/> https://skalatan.de/blog/ Kamailio services <https://gilawa.com/> https://gilawa.com From: sr-users < <mailto:sr-users-bounces@lists.kamailio.org> sr-users-bounces(a)lists.kamailio.org> On Behalf Of Ginhoux, Patrick Sent: Wednesday, October 5, 2022 10:04 AM To: <mailto:sr-users@lists.kamailio.org> sr-users(a)lists.kamailio.org Subject: [SR-Users] Mails received twice in my Inbox Hi, Since several weeks, I received the same mails twice in my Inbox. Examples : Is there anybody else that has the same thing ? Thanks in advance for your help. Cordialement. Patrick GINHOUX

1 year, 12 months

1
0
0 0

Kamailio IMS supports Early IMS Authentication

by PAUL SUN

Hello I have a question on using Kamailio IMS, does it support Early IMS Authentication?, Thanks RBK

1 year, 12 months

1
0
0 0

Kamailio v5.6.2 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.6.2 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.2. For more details about version 5.6.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2022/10/kamailio-v5-6-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 year, 12 months

4
6
0 0

I am having an error with Kamailio

by Michael Osude

Please see my output and see if you can help me resolve the error ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/usr/lib/systemd/system/kamailio.service; enabled; vendor preset: disabled) Active: failed (Result: start-limit) since Thu 2022-10-06 23:42:33 CDT; 10min ago Process: 9931 ExecStart=/usr/sbin/kamailio -DD -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY (code=exited, status=255) Main PID: 9931 (code=exited, status=255) Oct 06 23:42:32 s181866.wholesaleinternet.net systemd[1]: Unit kamailio.service entered failed state. Oct 06 23:42:32 s181866.wholesaleinternet.net systemd[1]: kamailio.service failed. Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: kamailio.service holdoff time over, scheduling restart. Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: start request repeated too quickly for kamailio.service Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: Unit kamailio.service entered failed state. Oct 06 23:42:33 s181866.wholesaleinternet.net systemd[1]: kamailio.service failed. Thank you

1 year, 12 months

2
1
0 0

Kamailio Developers Meeting, Nov 16-17, 2022, in Dusseldorf

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server project is organizing another meeting of its developers during November 16-17, 2022, hosted again by sipgate.de in Dusseldorf, Germany. The event is intended to facilitate the interaction between Kamailio developers and to offer a convenient environment for working together on several topics of high interest for the project, including writing code for Kamailio and its tools, improving documentation, or discuss about future development. Everyone from the community is welcome to join. Please note we have a limited capacity of seats in the meeting room, the main policy for accepting participants being first come first server. Also, very important to be aware that this is not an event to learn how to use Kamailio. More details about the event, the venue, how to register, are available at: * https://www.kamailio.org/w/developers-meeting/ Looking forward to those two intensive hacking Kamailio days in Dusseldorf! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Oct 10-13, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

1 year, 12 months

1
1
0 0

Mails received twice in my Inbox

by Ginhoux, Patrick

Hi, Since several weeks, I received the same mails twice in my Inbox. Examples : Is there anybody else that has the same thing ? Thanks in advance for your help. Cordialement. Patrick GINHOUX

2 years

2
4
0 0

How to use different version of python3 with KEMI

by Lợi Đặng

Hi all, regarding KEMI with python3, how to specify the python instance we're going to use instead of the system's? i.e: virtual environment or alternative installation of python3. Asking as I'm having existing code running on a version which does not match the system's, and upgrading system's to the specific version is not feasible. I tried specifying the python3 instance in src/modules/app_python3/Makefile but it seems only to help with the build, later the system python is used when kamailio starts. Any help would be appreciated. rgds, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

2 years

2
2
0 0

Releasing v5.6.2

by Daniel-Constantin Mierla

Hello, I am considering to release Kamailio v5.6.2 (out of branch 5.6) later this week (likely on Thursday, Nov 6, 2022). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 7-10, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 years

1
1
0 0

xhttp - default reply

by Ilie Soltanici

Hello, Is there a way to set a default reply message for the xhttp module, or at least drop the TCP connection when 'url_match' parameter is used? For example, if used: modparam("xhttp", "url_match", "^/rpc/") which should handle only HTTP requests to this endpoint: http://127.0.0.1:5060/rpc, now if HTTP requests are sent to another endpoint (http://127.0.0.1:5060/v2) - then the TCP connection just stays established even nobody process it. I know I can ignore this parameter and filter all requests in the event_route, but dropping them before calling event_route will make more sense for me, or at least add another parameter to this module that can control that. Thanks

2 years

2
1
0 0

Error in RTP engine setup

by palany

I have installed RTPENGINE on debian 11 and on starting it I am getting the following errors. Oct 04 12:14:39 ip-172-31-26-106 systemd[1]: Starting NGCP RTP/media Proxy Daemon... Oct 04 12:14:39 ip-172-31-26-106 ngcp-rtpengine-iptables-setup[4655]: /usr/sbin/ngcp-rtpengine-iptables-setup: 15: /usr/libexec/rtpengine/rtpengine-get-table: not found Oct 04 12:14:39 ip-172-31-26-106 rtpengine[4659]: INFO: [crypto] Generating new DTLS certificate Oct 04 12:14:39 ip-172-31-26-106 rtpengine[4659]: ERR: [core] FAILED TO CREATE KERNEL TABLE 0 (No such file or directory), KERNEL FORWARDING DISABLED Oct 04 12:14:39 ip-172-31-26-106 rtpengine[4659]: INFO: [core] Startup complete, version 11.1.0.0+0~mr11.1.0.0 git-master-c53f213f Oct 04 12:14:39 ip-172-31-26-106 systemd[1]: Started NGCP RTP/media Proxy Daemon. Oct 04 12:14:39 ip-172-31-26-106 rtpengine[4659]: INFO: [http] Websocket listener thread running How do I solve this. Best Regards, Palany Chakadya

2 years

3
6
0 0

Kamailio Fail to write logs using syslog

by M Arqum CH

Dear All, Thanks in advance for your help. kamailio installed on debain 9, sometime kamailio failed to generate any log in syslog files i.e messages/ syslog , not even generate a single line against complete dialog or generate partial logs. It's a random behaviour, as i am using syslog with the acc module, kamailio also failed to write acc logs on file which is a very critical matter for me. In a script using xinfo, xnotice, xerr function, no output from these functions in log file. Please suggest a way to fix this issue. modparam("xlog", "log_facility", "LOG_LOCAL1") version: kamailio 5.5.4 (x86_64/linux) 9d039b flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 9d039b -- Regards M Arqum

2 years

1
0
0 0

loose_route() clarification

by Mack Hendricks

Hey All, I always struggle with loose_route() and I implemented workarounds in the past. But, I want to get a better understanding. I have a SIP Endpoint -> Kamailio -> FreeSWITCH The domain and alias is set to the ip address of the Kamailio server (137.184.130.206) When I get the ACK back from the SIP Endpoint it looks like this ACK sip:18889072085@137.184.72.42:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.1.10.140:1052 ;branch=z9hG4bK-524287-1---740db0025fa217ce;rport Max-Forwards: 70 *Route: <sip:137.184.130.206;lr>* Contact: <sip:1000@50.192.97.226:53790;transport=UDP> To: <sip:18889072085@dsiptest.dsiprouter.net>;tag=5gv4etZ8tUUcH From: <sip:1000@dsiptest.dsiprouter.net;transport=UDP>;tag=0af5b751 Call-ID: fWido7VIUKIEI6f6kf4vkQ.. CSeq: 1 ACK User-Agent: Z 5.5.8 v2.10.17.2 Content-Length: 0 Kamailio will then remove the ACK and change the RURI to itself, which causes an ACK loop. This is what is look like *ACK sip:137.184.130.206;lr* SIP/2.0 Via: SIP/2.0/UDP 137.184.130.206;branch=z9hG4bK3fd.0cf34b683f0627ee2455594ea6cb2517.0 Via: SIP/2.0/UDP 10.1.10.140:1052 ;received=50.192.97.226;branch=z9hG4bK-524287-1---740db0025fa217ce;rport=1052 Max-Forwards: 69 Contact: <sip:1000@50.192.97.226:1052;transport=UDP> To: <sip:18889072085@dsiptest.dsiprouter.net>;tag=5gv4etZ8tUUcH From: <sip:1000@dsiptest.dsiprouter.net;transport=UDP>;tag=0af5b751 Call-ID: fWido7VIUKIEI6f6kf4vkQ.. CSeq: 1 ACK User-Agent: Z 5.5.8 v2.10.17.2 Content-Length: 0 *Question:* What am I missing here? We have the alias set and we even set it in the domain module (with the register_myself flag). We have workarounds for this, but I would prefer to figure out how to have Kamailio handle this natively via the loose_route or tm module. Thanks in advance,

2 years

3
2
0 0

Keepalive dialog with SIP/TLS OPTIONS

by Denys Pozniak

Hello! In order to keep the SIP/TCP connection alive for some specific clients (more precisely, specific firewalls) in long calls, I use periodic SIP OPTIONS probes through the dialog module. I noticed that when subscribers are using the SIP/TLS protocol, such keepalives are not sent (with SIP/UDP and SIP/TCP working well). I did not detect any obvious errors in the debug mode. Are there any ideas why the messages are not being sent for SIP/TLS leg? ---- loadmodule "dialog.so" modparam("dialog", "db_url", "DBURL") modparam("dialog", "db_mode", 1) modparam("dialog", "ka_timer", 5) modparam("dialog", "ka_interval", 30) modparam("dialog", "ka_failed_limit", 2) modparam("dialog", "send_bye", 1) modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "default_timeout", 7205) ----- dlg_manage(); dlg_set_property("ka-src"); dlg_set_property("ka-dst"); -- BR, Denys Pozniak

2 years

3
3
0 0

Kamailio stops responding to OPTIONS for some time

by Muhammad Danish Moosa

Hi, With Kamailio 5.5.3 (with KEMI/Lua) I noticed it apparently had stopped responding to OPTIONS in my test system. It can cause problems if a similar issue appears in production. Only thing i could observe it was apparently responding to INVITE and CANCEL etc randomly , its pretty strange as it was not dead completely. There was no core dump, I restarted and it started functioning again. Any clues please? -- Muhammad Danish Moosa " The core of mans' spirit comes from new experiences. "___ Christopher McCandless

2 years

2
1
0 0

http_async_client "route_name" is what type of route?

by Brooks Bridges

Running into an issue where I can't access certain functions inside the response route or "route_name" for handling the asynchronous reply from an http_async_query() that's being run. Does anyone know what type of route that is considered to be? I know it's not a request route or a failure route based on the errors I'm getting, but I can't find anything that outlines what that route actually is. Thanks! Brooks Bridges 1200 S. Rogers Circle, Unit 6 Boca Raton, FL 33487 Office: 916-235-2097 Main: 888-444-1111 email: bbridges(a)call48.com | web: www.call48.com Confidentiality Notice: This e-mail, and any attachment to it, contains privileged and confidential information intended only for the use of the individual(s) or entity named on the e-mail. If the reader of this e-mail is not the intended recipient, or the employee or agent responsible for delivering it to the intended recipient, you are hereby notified that reading this e-mail is strictly prohibited. If you have received this e-mail in error, please immediately return it to the sender and delete it from your system.

2 years

3
7
0 0

Interactions of dmq_usrloc and usrloc w/db_mode > 0

by Alex Balashov

Hi, Are there any known contraindications for replicating contacts using dmq/dmq_usrloc, and injecting those contacts into a database on one of the nodes using usrloc with `db_mode` 1 or 2? Predictably, this is being done to support the use-case of getting registration status from database. If you think this should be done with JSONRPC, I am in complete agreement with you, but it’s not up to me. :-) I am doing this with `db_mode` 2 now, and finding that, for a small number of AORs, one can find instances where they are consistently stored in memory but not present in the `location` table. The overall proportion of these AORs, out of thousands, seems to be quite small. It was initially somewhat higher, and it went down once I increased usrloc `timer_processes` and increased the sync interval from 30 to 60 seconds. Nevertheless, it is still non-zero, and I am getting intermittent reports. I wonder if there are some prior experiences with this and anything to watch out, or if `db_mode` 1 might be a superior choice. I personally cannot see how that would be; it seems to have all the performance downsides of mode 3. But perhaps if something about it is more “problem-free” vis-a-vis DMQ, it’s worth a shot? — 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 years

5
7
0 0

New feature: #defexp -preprocessor define with expression

by Daniel-Constantin Mierla

Hello, following the recent new feature of #!ifexp based on snexpr (https://www.kamailio.org/w/2022/09/ifexp-conditional-preprocessor-blocks/), another preprocessor directive was added which allows defining IDs with a value computed from the evaluation of an expression, respectively: #!defexp ID EXP The EXP can be any expression supported by snexpr, including the previously defined IDs. For example: #!define IPADDR 1.2.3.4 #!define PORT 5060 #!defexp SIPURI "sip:" + IPADDR + ":" + PORT The result is that SIPURI is defined to sip:1.2.3.4:5060. Another variant is available as #!defexps, which encloses the result of the expression in double quotes, making it suitable to be used further as string value: #!defexps SIPURI "sip:" + IPADDR + ":" + PORT The result is that SIPURI is defined to “sip:1.2.3.4:5060”. More details at: - https://www.kamailio.org/wikidocs/cookbooks/devel/core/#defexp - https://www.kamailio.org/wikidocs/cookbooks/devel/core/#defexps Testing would be appreciated, feedback can be addressed to sr-users mailing list! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online Nov 7-10, 2022 (Europe Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

2 years

1
0
0 0

Question about the Secsipid module

by Володимир Іванець

Hello everyone! I'm testing calls with STIR/SHAKEN with a Secsipid module on a Kamailio version 5.6.1 and ran into a problem. Outbound calls work fine when the function *secsipid_add_identity* with appropriate parameters is called. But as soon as I add function *secsipid_check_identity* to my configuration, Kamilio would not start. Here is an error message I get: *kamailio: CRITICAL: <core> [core/cfg.y:3791]: yyerror_at(): parse error in config file /etc/kamailio/include/registrar.cfg, line 47, column 39: unknown command, missing loadmodule?* *loadmodule "secsipid.so"* is present and is above the function *secsipid_check_identity* call. Both secsipid.so and secsipid_proc.so module files are the correct version and present on the system. Another question is related to the rpm package build. How do you build additional modules? I made adjustment to the pkg/kamailio/Makefile file. In the *cfg* section changed *$(MAKE) -C ../../src cfg* with *$(MAKE) FLAVOUR=kamailio include_modules="secsipid secsipid_proc" -C ../../src cfg*. Required modules appeared in the modules.lst file but their rpm packages were not built. Thank you very much!

2 years

2
8
0 0

Failed to create a TLS connection when a 'loose' tls_method is configured

by Leonid Fainshtein

Hi, I have a problem creating a TLS connection when tls_method is configured as one of the "loose" methods: SSLv23, TLSv1+, TLSv1.1+, TLSv1.2+. The problem doesn't happen when the tls_method is defined "strictly": TLSv1, TLSv1.1, or TLSv1.2. The problem is reproducible with the modern Yealink phones and also with sipexer. I have Kamailio v.5.5.4 installed from the Kamailio repo ( http://deb.kamailio.org/kamailio55), on Ubuntu 18.04, OpenSSL ver.1.1.1-1ubuntu2.1~18.04.20 The sipexer output: -------------------------- $ ./sipexer -register -vl 3 -co -com -ex 60 -fuser 2001 -cb -au 2001 -ap "......" -sd -sw 20000 -tk ./id_rsa tls:nnnnnnnnn.com:5061 [debug] [sipexer.go:912] main.main(): parsed socket address argument ({Val:tls:nnnnnnnnn.com:5061 Proto:tls Addr:nnnnnnnnn.com Port:5061 PortNo:5061 AType:8 ProtoId:3}) [debug] [sipexer.go:1800] main.SIPExerSendTLS(): client: 192.168.0.178:41586 connected to: 142.93.233.32:5061 [debug] [sipexer.go:1803] main.SIPExerSendTLS(): [48 130 1 34 48 13 6 9 42 134 72 134 247 13 1 1 1 5 0 3 130 1 15 0 48 130 1 10 2 130 1 1 0 222 21 229 237 77 230 193 157 183 132 186 194 65 98 162 250 243 232 189 223 115 181 41 199 130 221 201 143 254 218 254 201 82 173 233 44 158 103 150 78 210 162 187 138 5 179 11 47 208 0 117 27 186 180 197 229 94 13 63 72 227 27 169 127 197 173 67 52 182 68 140 220 132 20 241 57 24 182 29 234 58 60 214 145 225 29 251 240 190 159 43 233 53 57 121 159 81 4 200 12 193 90 91 158 78 196 25 202 74 111 106 215 164 12 178 120 129 8 81 1 245 181 149 20 47 197 165 74 126 24 249 241 187 197 249 209 192 11 204 254 110 19 248 79 122 11 131 24 101 155 237 110 186 5 214 88 79 255 185 156 52 12 29 222 254 76 71 171 142 28 69 22 34 31 111 105 87 7 108 68 6 23 199 236 134 154 42 208 199 158 190 48 193 192 179 120 189 199 117 71 104 231 127 73 128 91 234 168 175 207 55 6 191 144 209 25 10 182 27 53 182 182 96 248 198 206 220 207 102 236 194 96 67 159 34 112 81 145 2 166 207 86 247 232 117 126 158 214 215 10 226 68 81 235 169 81 2 3 1 0 1] <nil> [debug] [sipexer.go:1804] main.SIPExerSendTLS(): CN=nnnnnnnnn.com [debug] [sipexer.go:1803] main.SIPExerSendTLS(): [48 130 2 34....247 31 2 3 1 0 1] <nil> [debug] [sipexer.go:1804] main.SIPExerSendTLS(): CN=ZeroSSL RSA Domain Secure Site CA,O=ZeroSSL,C=AT [debug] [sipexer.go:1803] main.SIPExerSendTLS(): [48 130 2 34 48 ....130 15 2 3 1 0 1] <nil> [debug] [sipexer.go:1804] main.SIPExerSendTLS(): CN=USERTrust RSA Certification Authority,O=The USERTRUST Network,L=Jersey City,ST=New Jersey,C=US [debug] [sipexer.go:1806] main.SIPExerSendTLS(): client: handshake: true [debug] [sipexer.go:1807] main.SIPExerSendTLS(): client: mutual: true [info] [sipexer.go:1432] main.SIPExerDialogLoop(): local socket address: 192.168.0.178:41586 (tls) [info] [sipexer.go:1433] main.SIPExerDialogLoop(): local via address: 192.168.0.178:41586 [info] [sipexer.go:1434] main.SIPExerDialogLoop(): sending to tls 142.xx.yy.32:5061: [[--- REGISTER sip:nnnnnnnnn.com:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 192.168.0.178:41586 ;rport;branch=z9hG4bKSG.f7a247eb-5440-4e7a-8a5a-4f849684f30d From: <sip:2001@nnnnnnnnn.com>;tag=33e66091-193c-4535-b659-d5e562aca5a4 To: <sip:2001@nnnnnnnnn.com> Call-ID: 4556156d-e8da-4632-82f7-f3ab3c4c73a1 CSeq: 621470 REGISTER Date: Sun, 04 Sep 2022 11:21:33 IDT Contact: <sip:192.168.0.178:41586;transport=tls> Expires: 60 User-Agent: SIPExer v1.0.2 Content-Length: 0 [info] [sipexer.go:1436] main.SIPExerDialogLoop(): ---]] [error] [sipexer.go:1312] main.SIPExerDialogReadBytes(): not receiving after 32000ms (bytes 0 - EOF) [debug] [sipexer.go:1009] main.SIPExerExit(): return code: -1154 The Kamailio debug log: --------------------------------- Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 207.232.46.53 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 18566, type 3, socket 72 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/tcp_main.c:1507]: tcpconn_add(): hashes: 1963:2250:2856, 16 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): processing io_watch_add(0x5648d2a55e40, 72, 2, 0x7f9dcaf01958) - fd_no=63 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x5648d2a55e40, 72, -1, 0x0) fd_no=64 called Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/tcp_main.c:4471]: handle_tcpconn_ev(): sending to child, events 1 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/tcp_main.c:4141]: send2child(): selected tcp worker idx:7 proc:45 pid:25549 for activity on [tls:142.xx.yy.32:5061], 0x7f9dcaf01958 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_read.c:1737]: handle_io(): received n=8 con=0x7f9dcaf01958, fd=10 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:244]: tls_complete_init(): completing tls connection initialization Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:274]: tls_complete_init(): Using initial TLS domain TLSs<default> (dom 0x7f9dc4c17078 ctx 0x7f9dc4c65da0 sn []) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f9dc4c65da0: (nil) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:778]: sr_ssl_ctx_info_callback(): SSL handshake started Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:979]: tls_server_name_cb(): received server_name (TLS extension): 'tmx.cpbxmt-demo187.xorcom.com' Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:998]: tls_server_name_cb(): TLS cfg domain selected for received server name [tmx.cpbxmt-demo187.xorcom.com]: socket [:0] server name='' - switching SSL CTX to 0x7f9dc4c65da0 dom 0x7f9dc4c17078 (default) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_main.c:2720]: tcpconn_do_send(): sending... Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_main.c:2753]: tcpconn_do_send(): after real write: c= 0x7f9dcaf01958 n=5435 fd=10 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_main.c:2754]: tcpconn_do_send(): buf= Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: #026#003#003 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): processing io_watch_add(0x5648d29e8c80, 10, 2, 0x7f9dcaf01958) - fd_no=1 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:1208]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f9dc4c65da0: (nil) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:778]: sr_ssl_ctx_info_callback(): SSL handshake started Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:782]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:778]: sr_ssl_ctx_info_callback(): SSL handshake started Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:782]: sr_ssl_ctx_info_callback(): SSL renegotiation initiated by client Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_domain.c:790]: sr_ssl_ctx_info_callback(): SSL handshake done Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:470]: tls_accept(): TLS accept successful Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:477]: tls_accept(): tls_accept: new connection from 207.232.46.53:18566 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:480]: tls_accept(): tls_accept: local socket: 142.xx.yy.32:5061 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:491]: tls_accept(): tls_accept: client did not present a certificate Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: tls [tls_server.c:1245]: tls_h_read_f(): Reading on a renegotiation of connection (n:-1) (2) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_read.c:1503]: tcp_read_req(): EOF Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x5648d29e8c80, 10, -1, 0x10) fd_no=2 called Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_read.c:1872]: handle_io(): removing from list 0x7f9dcaf01958 id 16 fd 10, state 2, flags 4018, main fd 72, refcnt 2 ([207.232.46.53]:18566 -> [207.232.46.53]:5061) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_read.c:1656]: release_tcpconn(): releasing con 0x7f9dcaf01958, state -1, fd=10, id=16 ([207.232.46.53]:18566 -> [207.232.46.53]:5061) Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 45(25549) DEBUG: <core> [core/tcp_read.c:1660]: release_tcpconn(): extra_data 0x7f9dcaefe628 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: <core> [core/tcp_main.c:3573]: handle_tcp_child(): reader response= 7f9dcaf01958, -1 from 7 Sep 4 08:21:33 cpbxmt-demo187 kamailio[23878]: 46(25550) DEBUG: tls [tls_server.c:729]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7f9dcaefe628 As far as I understood the log, the certificate was accepted but something wrong happened later. Any suggestion? Best regards, Leonid Fainshtein

2 years

2
3
0 0

No CDR is written on failover

by Björn Klasen

Hi Community, I'm experimenting with dmq to do dialog sync between two instances of Kamailio. This works great so far. When I did my first experiments I even could create a CDR when there was a failover. Since some Versions (we are now on Version 5.6.1) I just get a acc-request in log but no acc-cdr. In former versions this worked. Both server have the same configuration and both server writes cdrs when theres no failover. In case of a failover everything is fine, the dialogs are clean and thanks to dmq all needed variables are replicated between the server. There's just no cdr. I found nothing on the issue-page on github but maybe somebody knows something about it. BR Björn -- Björn Klasen, Senior Specialist (VoIP) TNG Stadtnetz GmbH, TNG-Technik Projensdorfer Str. 324 24106 Kiel・Deutschland T +49 431 7097-10 F +49 431 7097-555 bklasen(a)tng.de www.tng.de <http://www.tng.de> Executive board (Geschäftsführer): Dr. Sven Willert (CEO/Vorsitz), Helmut Gertz, Moritz Meis, Gunnar Peter, Sven Schade, Carsten Tolkmit Amtsgericht Kiel HRB 6002 KI USt-ID: DE225201428 Die Information über die Verarbeitung Ihrer Daten gemäß Artikel 12 DSGVO können Siehier <https://tng.de/privatkunden/datenschutz/>abrufen.

2 years

2
2
0 0

Kamailio + RTPEngine SIP Proxy...

by David Walsh

Hi Jerry, I just joined so am unsure if you are good to go now but thought I would post my config as I also have a Freeswitch environment which I have placed Kamailio in front of to act as a load-balancing proxy. I also managed to get RTPEngine proxying through it too. Here is the config, which I have created from Daniel’s default kamailio config. I have omitted the top portions of loading the modules and their settings. My dispatcher list is just a text file. I have two groups. One for registrations and outbound calls, and one for calls inbound coming from the FreeSwitch server. In the NATMANAGE route, I currently have RTP proxying disabled as my free switch servers have external IPs and I let them handle the rtp stream. On Freeswitch, I made the following changes to accomodate user/pass and IPAuth reg: Changes to the Freeswitch servers listed in the dispatcher file Edit /usr/local/freeswitch/conf/autoload_configs/acl.conf <list name="proxy" default="deny"> <node type="allow" cidr=“IP address of Kamailio/32"/> </list> Edit /usr/local/freeswitch/conf/sip-profiles/internal-private.xml (or the sip-profile you need to edit in your instance) <param name="apply-proxy-acl" value="proxy"/> #Add this line above the one below. This tells Freeswitch to look at the X-Auth-IP header which was inserted by Kamailio. <param name="apply-inbound-acl" value="domains"/> Then, restart freeswitch (with no active calls), service freeswitch restart # or rescan the profile / reloadxml I am not proxying RTP, however, I can un-comment out the stanza in the NATMANAGE route to get it to work. The biggest gotcha for me was I was trying to use fix_nated_register when handling NAT for registrations instead of: Fix_nated_contact() or the better set_contact_alias(). Using add_path_received(); in the Dispatcher just before sending to Freeswitch ensured all traffic returned to Kamailio too. I am unsure of the SUBSCRIBE/PRESENCE stuff for you but the route should go out via the Kamailio proxy with this. ---------------------------- # main request routing logic request_route { xlog("L_WARN", "--- Going to <$ru>. src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si\n"); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); if(ds_is_from_list("33")) { setflag(FLT_FS); #xlog("L_WARN", "This source $si is from the dispatcher list.\n"); } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_WARN", "--- Going to <$ru>. src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si\n"); xlog("L_ALERT", "Marker 1\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if (isflagset(FLT_FS)) { ##t_on_reply("EXTERNAL_REPLY"); route(FROM_FS); exit; } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); return; } route[FROM_FS] { #record_route(); #loose_route_preloaded(); route(DLGURI); route(RELAY); exit; } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } #reply_route { # if(status == 403) { # xlog("L_WARN", "Forbidden from $si:$sp blah...\n"); # xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); # } #} onreply_route[LOGRPL] { if(status == 403) { xlog("L_ALERT", "$rs response. Consider banning AVP \"ipbancandidate\" is $avp(ipbancandidate) "); } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies # Enable tracing for SNGREP ##sip_trace(); ##setflag(24); # set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly|scanner|sipcli|sipvicious|VaxSIPUserAgent|pplsip|Hello\ SIP\ Automated") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); xlog("User agent is: $ua\n"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO", "Routing before DLGURI\n"); route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; xlog("L_WARN", "--- Going to <$ru>. src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si\n"); add_path_received(); $avp(ipbancandidate) = $si; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } route(DISPATCH); if (!save("location")) { sl_reply_error(); } exit; } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { #fix_nated_register(); Disabled as we proxy registrations #fix_nated_contact(); set_contact_alias(); ##add_contact_alias(); #fix_nated_sdp("3"); xlog("L_ALERT", "Nat Registration nat fixed.\n"); } else { if(is_first_hop()) { set_contact_alias(); xlog("L_ALERT", "Contact NAT fixed\n"); } } setflag(FLT_NATS); xlog("L_ALERT", "FLT_NATS flag set\n"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (nat_uac_test("3")) { #fix_nated_contact(); set_contact_alias(); ##add_contact_alias(); force_rport(); } if (has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("11"); } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; ###!ifdef WITH_RTPENGINE ## xlog("RTPEngine enabled\n"); ## if(nat_uac_test("8")) { ## xlog("RTP 1\n"); ## rtpengine_manage("SIP-source-address replace-origin replace-session-connection"); ## } else { ## rtpengine_manage("replace-origin replace-session-connection"); ## } ###!else ## if(nat_uac_test("8")) { ## rtpproxy_manage("co"); ## } else { ## rtpproxy_manage("cor"); ## } ###!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); xlog("Nat notation added!\n"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) #fix_nated_contact(); set_contact_alias(); #add_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); xlog("L_INFO", "Routing in-dialog $rm from $fu to $du\n"); } #!endif return; } # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' remove_hf_re("Subject|P-.*|X-.*"); append_hf("X-Auth-IP: $si\r\n"); #add_rr_param(";nat=yes"); t_on_reply("LOGRPL"); if(!ds_select_dst("1", "0")) { send_reply("404", "No destination"); exit; } #xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); xlog("L_WARN", "--- Going to <$ru>. src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si; $RAi; $Ri;\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); xlog("L_ALERT", "Marker 20\n"); route(NATMANAGE); return; } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } return; } # Manage incoming replies in transaction context onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); xlog("L_ALERT", "Marker 21\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } # if (has_body("application/sdp")) { # rtpengine_manage(); # } return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { xlog("L_ALERT", "Marker 23\n"); route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif return; }

2 years

1
0
0 0

MANAGE_REPLY not getting replies to PRACK?

by Benoît Panizzon

Hi all On calls with privacy enabled to customers, I want to ensure no information is disclosed. Not even if one would capture packets. A PRACK and the 200 OK to a PRACK therefore may need the To and From Header changed. Unfortunately I seem not to be able to catch the 200 OK REPLY sent to a PRACK with t_on_reply. It looks like they somehow pass completely unprocessed through Kamailio (not completely Via Header are removed accordingly) Am I missing something? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 years

1
0
0 0

Guide to set up Kamailio as an SBC

by Unai Rodriguez

Dear List, I’m new to Kamailio and would like to set it up as a standalone general purpose SBC. I’m having a hard time finding a guide specific for this. What can you recommend? Thank you so much With best wishes, Unai Rodriguez

2 years

2
1
0 0

app_python3 crash woes

by Alex Balashov

Hi, Some exceptions generated in Python script cause a harmless traceback in the logs, while others cause Kamailio to crash: Sep 22 06:28:34 proxy /usr/sbin/kamailio[3984]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 11 Sep 22 06:28:34 proxy /usr/sbin/kamailio[3957]: ALERT: <core> [main.c:783]: handle_sigs(): child process 3965 exited by a signal 11 Sep 22 06:28:34 proxy /usr/sbin/kamailio[3957]: ALERT: <core> [main.c:786]: handle_sigs(): core was generated (gdb) where #0 0x00007fdaea729079 in vfprintf () from /lib64/libc.so.6 #1 0x00007fdaea754179 in vsnprintf () from /lib64/libc.so.6 #2 0x00007fdac45e9afd in make_message () from /usr/lib64/kamailio/modules/app_python3.so #3 0x00007fdac45e5bc9 in python_handle_exception () from /usr/lib64/kamailio/modules/app_python3.so #4 0x00007fdac45ebfde in apy_exec () from /usr/lib64/kamailio/modules/app_python3.so #5 0x00007fdac45ecef1 in python_exec2 () from /usr/lib64/kamailio/modules/app_python3.so #6 0x0000000000474ee0 in do_action () #7 0x000000000048217d in run_actions () #8 0x0000000000474d39 in do_action () #9 0x000000000048217d in run_actions () #10 0x0000000000474d39 in do_action () #11 0x000000000048217d in run_actions () #12 0x0000000000474d87 in do_action () #13 0x000000000048217d in run_actions () #14 0x00000000004828c1 in run_top_route () #15 0x00000000005e8044 in receive_msg () #16 0x00000000004df772 in udp_rcv_loop () #17 0x0000000000429f5b in main_loop () #18 0x000000000043423e in main () Is there a reasonable way to try to prevent this type of mid-air explosion? — 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 years

2
2
0 0

SIPp uas(SIPp)->client reINVITE late offer

by David Villasmil

Hello all, Anyone got a SIPp scenario for reINVITE with late offer? I.e.: UAT--->INVITE (sdp)--->SIPp UAT<--- 100 <---SIPp UAT<--- 183 <---SIPp UAT<--- 200 OK <---SIPp UAT----> ACK --->SIPp UAT<--- INVITE (NO sdp) <---SIPp UAT---> 100 --->SIPp UAT---> 183 --->SIPp UAT---> 200 OK (sdp) -->SIPp UAT<--- ACK (sdp) <---SIPp UAT---> BYE -->SIPp UAT<--- ACK <--SIPp I can’t get past UAT<--- INVITE (NO sdp) <---SIPp Kamailio responds with “not here” Appreciate the help! -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 years

3
4
0 0

Substdef syntax change?

by Fredrik Dahlgren

Hi, I'm looking at a kamailio config where all substdef have the syntax "!regexp!subst!flags", "!DB_URL!db.example.com!g" but i can't find this syntax in the documentation. It's currently running on a fairly modern 5.x version and as far as i know it has never been on 4.x or older. Does anyone know where this syntax is from and if it's possible/recommended to simply replace it with "/regexp/subst"? *Fredrik* *Dahlgren* | Senior SIP/VoIP Engineer fredrik.dahlgren(a)freespee.com | LinkedIn <https://www.linkedin.com/company/freespee/> | Twitter <https://twitter.com/freespee>

2 years

2
1
0 0

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