sr-users October 2022

sr-users@lists.kamailio.org

48 participants
48 discussions

Appending new branches via external means

by Alex Balashov

Hi, Is there a means of appending new branches from config script for a transaction which is already proceeding in the background, without using `tsilo`? That is to say, I have an async transaction which is resumed in an async worker and is t_relay()’d away to some destination. Meanwhile, I have learned about new destinations for it to try via an incoming registration, and would like to append additional parallel forking branches to that transaction, even though it is already proceeding. ts_append() does this for me; the issue I have is that it is mediated through a registrar, but I am not a registrar. My registrar is upstream, and my proxy is just a front-end to it. So, in essence, I require something like `tsilo` at a more general level, where contacts can be populated by some other mechanism rather than being learned via callbacks out of `usrloc`. Is there a way to accomplish this? Cheers, — 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

consume_credentials not working on PRACK?

by Benoit Panizzon

Hi List I noticed, that one of our CPE copies the Proxy-Authorization HF in almost all messages sent. As PRACK were not authenticated, those headers were potentially sent on to the destination disclosing the authentication username and realm. So assuming, if credentials are present, the client wishes them to be validated, I added: if (has_credentials("$fd")) { xlog("L_INFO", "$cfg(route): got $rm with credentials. Validate them!\n"); route(AUTH); } and in route[AUTH] I call: pv_auth_check() which returns 1 thus success upon which I use: if(!is_method("REGISTER|PUBLISH")) consume_credentials(); If the method is INVITE: Proxy-Authorization HF is removed by consume_credentials() if the method is PRACK: Proxy-Authorization HF is still present on the outbound leg. 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 years

SIP routing per MSISDN

by Anthony Blandin

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

2 years

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.

2 years

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.

2 years

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

2 years

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

2 years, 1 month

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

2 years, 1 month

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 ______________________________________________________

2 years, 1 month

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

2 years, 1 month

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sr-users October 2022