sr-users September 2025

sr-users@lists.kamailio.org

42 participants
40 discussions

How to append_branch() after location lookup without affecting looked up branches?
by Benoit Panizzon 09 Sep '25

09 Sep '25

Hi List I'm pretty sure I had already accomplished that task a couple of months ago while manually looping through the more than one registered contact and doing append_branch() for each contact, using a complicated way of stacking various AVP to be able to handle each branch correctly in the corresponding branch routes. But that was the complicated way to go :-) Now I hope I do get branches from multiple locations the intended way by using lookup_to_dset, t_load_contacts, t_next_contacts But I would like to be able to add more branches - not from the location database - for the sake of this example just one more branch to a static destination. For example I do a lookup_to_dset and end up with 2 contacts I loaded from the registered AOR. Now I would like to add another branch and fork the call to all three destinations. So additionally to the destination set obtained by lookup_to_ds I would like to add <sip:someuser@someserver> and have the call forked to the IP of someserver So I basically try: lookup_to_dset("location", "$var(lookupuri)"); t_load_contacts(); $du = "<sip:someserver>"; $rU = "someuser"; append_branch(); t_on_branch("BRANCH_XY"); route(RELAY); But no matter how I try to combine $rU or $ru or the complete URI including user in $du this seems to either affect all branches or none. I guess I am missing something. How do I append one branch without affecting all existing branches? 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 ______________________________________________________

3 6

Looking for SIP testing tool
by Fernando Lopes 08 Sep '25

08 Sep '25

Hello Everyone, I’m looking for a SIP testing tool to test my configurations and scenarios, as well as to check security and handle a large number of ongoing calls. Could you please recommend some reliable options? Thank you.

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Kamailio behind NAT
by Martin Nyström 06 Sep '25

06 Sep '25

Hello, I am not successful in my attempts to configure my Kamailio to work behind NAT. The flow of an incoming call is Provider (Internet) -> AWS LoadBalancer -> Kamailio -> Asterisk Both the Kamailio and Asterisk is on the internal network. The issue I am having is that I need to add Record-Route to the traffic sent back towards the provider, but not to the Asterisk. Currently when I add the record_route() the header is sent to Asterisk which makes it reply to the Kamailio advertised external address for ACKs, BYEs etc. I have dumbed down my Kamailio config as much as possible for this, to show what I am currently doing. debug=2 log_stderror=yes fork=yes tcp_accept_no_cl=yes onsend_route_reply=yes pv_buffer_size=2048 enable_tls=1 listen=udp:<LOCAL_IP>:5060 advertise <EXTERNAL_DOMAIN>:5060 listen=tcp:<LOCAL_IP>:5060 advertise <EXTERNAL_DOMAIN>:5060 listen=tls:<LOCAL_IP>:5061 advertise <EXTERNAL_DOMAIN>:5061 local_rport=on mpath="/usr/local/lib/kamailio/modules_k/:/usr/lib/x86_64-linux-gnu/kamailio/modules/" # MODULES loadmodule "..." route { route(FROM_PROVIDER); } route[RELAY] { if(!t_relay()) { sl_reply_error(); } exit; } route[FROM_PROVIDER] { # The Asterisk that should not receive the external dns in the record route header ds_select_dst(100, 4); # INFO: This adds the Record-Route in all directions if(!has_totag()) { record_route(); } route(RELAY); exit; } /M

6 14

Kamailio + Asterisk Kubernetes Signaling problem.
by Fernando Lopes 05 Sep '25

05 Sep '25

Hello Everyone, I’m running into a problem with Kamailio and Asterisk inside Kubernetes. Both are exposed to the outside. My Network: External IP: 192.168.1.99:32210 TCP Kamailio Service IP: 10.101.179.164 Kamailio Pod IP: 10.244.10.35 Asterisk Service IP: 10.99.122.141 Asterisk Pod IP: 10.244.10.38 The issue is: When connecting directly to Asterisk, SIP calls work with not problems if I remove the local_net setting, so that Asterisk advertises its external IP. When connecting through Kamailio as a proxy, calls only “works” if I enable local_net, but still client can't send ACK to 200 OK. I believe the problem is that Asterisk sets the Contact header to its pod IP instead of the Service IP and Kamailio is not able to send ACK which leads to call timeout. Any Ideas? My Transport: [transport-tcp-nat] type=transport protocol=tcp bind=0.0.0.0 local_net=10.244.10.0/16 local_net=10.96.0.0/12 external_media_address=192.168.1.99 external_signaling_address=192.168.1.99 external_signaling_port=32210 SIP Logs: <--- Transmitting SIP response (1293 bytes) to UDP:10.244.10.35:5001 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.190:5001;rport=5001;received=10.244.10.35;branch=z9hG4bK683e.ce5a170c8d281e2227ec5a8870044ad9.0;i=4 Via: SIP/2.0/TCP 10.244.10.1:54435;rport=54435;received=10.244.10.1;branch=z9hG4bKPjb54c457d4a7d48959ca03391a56e1136;alias Call-ID: d0e65953da0f406498b95e77aae9f850 From: <sip:6001@192.168.1.99>;tag=d9761b4e52944f068f85ff71d139f660 To: <sip:100@192.168.1.99>;tag=76784095-78d9-4467-88e6-d9fba8bfbcd6 CSeq: 21022 INVITE Server: Asterisk PBX 18.24.3 Contact: <sip:10.244.10.38:5060> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 487 v=0 o=- 3966074837 3966074839 IN IP4 10.244.10.38 s=Asterisk c=IN IP4 10.244.10.38 t=0 0 m=audio 10056 RTP/AVP 9 0 101 a=ice-ufrag:4e2d77c12b938bbc4c7f3c773f3a262a a=ice-pwd:5df2f60432675fb26b988faa6a42c1c1 a=candidate:Hc0a80163 1 UDP 2130706431 192.168.1.99 10056 typ host a=candidate:Hc0a80163 2 UDP 2130706430 192.168.1.99 10057 typ host a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:140 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 Kamailio Log: 17(65) WARNING: <script>: [onsend_route] Packet going to sip:10.99.122.141:5060 17(65) WARNING: <script>: [RTPENGINE] ACK 17(65) WARNING: <script>: [REQUEST]: d0e65953da0f406498b95e77aae9f850: got ACK ACK sip:100@192.168.1.99:31953;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 10.244.10.1:54435;rport;branch=z9hG4bKPj130d3b53570f45afa764dd874a115771;alias Max-Forwards: 70 From: <sip:6001@192.168.1.99>;tag=d9761b4e52944f068f85ff71d139f660 To: <sip:100@192.168.1.99>;tag=z9hG4bK983e.12c423f5a5497fcaafae93af5228672e.0 Call-ID: d0e65953da0f406498b95e77aae9f850 CSeq: 21021 ACK Content-Length: 0 WireShark Log: Session Initiation Protocol (ACK) Request-Line: ACK sip:10.244.10.38:5060 SIP/2.0 Method: ACK Request-URI: sip:10.244.10.38:5060 Request-URI Host Part: 10.244.10.38 Request-URI Host Port: 5060 [Resent Packet: False] [Request Frame: 2849862] [Response Time (ms): 3968] Message Header Via: SIP/2.0/UDP 192.168.1.163:49666;rport;branch=z9hG4bKPje5005f15847d460d81553d1f5a5e0393 Transport: UDP Sent-by Address: 192.168.1.163 Sent-by port: 49666 RPort: rport Branch: z9hG4bKPje5005f15847d460d81553d1f5a5e0393 Max-Forwards: 70 From: <sip:6001@192.168.1.99>;tag=d8b13b005dda4b008eaf52f5d7c1f902 SIP from address: sip:6001@192.168.1.99 SIP from address User Part: 6001 SIP from address Host Part: 192.168.1.99 SIP from tag: d8b13b005dda4b008eaf52f5d7c1f902 To: <sip:100@192.168.1.99>;tag=768c6b9f-afd5-4546-b650-8616bbec7fd7 SIP to address: sip:100@192.168.1.99 SIP to address User Part: 100 SIP to address Host Part: 192.168.1.99 SIP to tag: 768c6b9f-afd5-4546-b650-8616bbec7fd7 Call-ID: 210137ed6d6145bc8f864a6b1d9f0f31 [Generated Call-ID: 210137ed6d6145bc8f864a6b1d9f0f31] CSeq: 8750 ACK Sequence Number: 8750 Method: ACK Content-Length: 0 Thank you.

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Support for TLS server_name extension (aka SNI=server name indication)
by Barry Flanagan 04 Sep '25

04 Sep '25

Hi, Back in Kamailio 1.5, the release notes state: "support for TLS server_name extension (aka SNI=server name indication)" However, I cannot find any indication of this in the current TLS docs, and trying to set tls_server_name or server_name in tls.cfg fails with "unsupported option". Is this actually supported? Thanks. -Barry Flanagan

5 7

Need to install rtpengine seperately?
by Chandramouli P 03 Sep '25

03 Sep '25

Hello, I have installed Kamailio v6.0.2 on my server (RHEL 8). I studied somewhere that rtpengine module is integrated with Kamailio in the recent versions. Do I need to install rtpengine separately or not? Thank you. Best regards, Chandramouli.

3 2

Adding rport to outbound SIP INVITE
by Martin Nyström 03 Sep '25

03 Sep '25

Hi, Is it possible to add rport to the top Via for the outbound INVITE? The Kamailio be behind an AWS NAT Gateway and we’re having the issue with the provider not replying to the source port but instead 5060. From what I have understood if we add the rport; without value in the Via it will tell the receiver to reply to the source ip and port. Here’s the route block for calls sent to the PSTN. route[PSTN] { ds_select_dst(1, 4); force_rport(); record_route(); t_on_failure("FAILURE_PROVIDER"); route(RELAY); exit; } The outbound invite towards PSTN looks like this, missing rport in the top Via INVITE sip:+4670xxxxxx@216.xxx.xxx.xxx SIP/2.0 Record-Route: <sip:10.3.130.39:5060;lr;ftag=b9827f5a-fef3-42f6-b8a4-aded67374bfb;lr;vsf=AAAAAAAAAAAAAAAAAAAAAAABFhMACBBKBhFDQhMBQhgBSggVQQ8SQAEWBk8ACm5uZWN0ZWwuYXBw> Via: SIP/2.0/UDP sbc.coolcompany.com:5060;branch=z9hG4bK350f.eb699686bbd2c208bd77d3d92fd5ed55.0 Via: SIP/2.0/UDP 10.2.8.34:5060;received=10.2.8.34;rport=5060;branch=z9hG4bKPj405acf54-c51d-4681-a7c7-9774ee63e219 From: <sip:+4610xxxxxx@sbc.coolcompany.com:5060>;tag=b9827f5a-fef3-42f6-b8a4-aded67374bfb To: <sip:+4670xxxxxx@xxxxxxxx> Contact: xxxx Call-ID: bba640af-369e-4a6c-b5ec-3974919d98c9 CSeq: 5899 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 69 User-Agent: Voice SBC Content-Type: application/sdp Content-Length: 310 /M

2 2

SDP Issue on NAT environment
by Nathan C 02 Sep '25

02 Sep '25

Hello, I have a kamailio on SITE A using a public IP. I'm trying to setup another kamailio on SITE B with NAT. When I try to make a call from SITE B to SITE A, the call is initiated but when comes the SDP my equipment on SITE B does not reply to SDP sent by kamailio on SITE A. I made some test and use only Public IP between the 2 sites and it works properly. So I think that I have a problem on my NAT configuration or maybe I'm missing something. Can someone help me one this problem ? Do you have an idea on what could be blocking. Thank in advance for your help. Regards, Nathan

3 6

Is a kamailio integer signed or unsigned?
by Benoit Panizzon 02 Sep '25

02 Sep '25

Hi I'm chasing an issue where I have a mysql column of type int NOT NULL In MySQL this is a signed integer. In some cases this column can be negative for example -1 I read this column with sql_xquery I end up with 18446744073709551615 That is the unsigned equivalent of 2^64-1 - so kamailio uses SIGNED int? Is there a way to change this to unsigned or to read have sql_xquery convert the value to a string? 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

[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 02 Sep '25

02 Sep '25

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

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sr-users September 2025