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42 participants
37518 discussions

SIP routing per MSISDN
by Anthony Blandin 30 Oct '22

30 Oct '22

Hello, I need to route SIP INVITE towards a specific node based on the MSISDN of the Request URI. I saw that the dispatcher module should be able to make this with the function ds_select_dst and the algorithm "hash over request-URI user" but I don't understand how the routing is done if we can't associate the request-URI user with the destination address in the dispatcher DB. Maybe my understanding of this module is wrong. Could you help me to achieve this? Thanks Anthony

3 2

Kamailio sending response to source IP and port from Via ?
by Chaigneau, Nicolas 27 Oct '22

27 Oct '22

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.

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Issue with debug 3 after upgrade to kamailio 5.6.2+
by Julia Bo 27 Oct '22

27 Oct '22

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.

2 3

About the WITH_VOICEMAIL Subscriber message 407 issue
by SparkleZou 26 Oct '22

26 Oct '22

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

3 4

ims_auth with AKAv1-MD5
by Oleg Belousov 24 Oct '22

24 Oct '22

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

2 2

No REGISTER sent to third party AS from scscf
by Anthony Blandin 24 Oct '22

24 Oct '22

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

2 3

Dialog Module: How to get two different dialogues if a call is looped through kamailo (call forwarding scenario)
by Benoit Panizzon 21 Oct '22

21 Oct '22

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 ______________________________________________________

2 2

relay sip-tls vs sips-tcp
by beer Ll 21 Oct '22

21 Oct '22

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

3 4

After changing instance type for kamailio server in aws log file is corrupted
by Sammy Dominguez 19 Oct '22

19 Oct '22

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 0

Issue with duplicite SIP messages in traffic loop
by Michal Popovič 19 Oct '22

19 Oct '22

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

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