- sr-users - kamailio.org

How to reply 484 to uac, and send bye to uas
by 13715209697＠139.com 18 Jun '25

18 Jun '25

Hi team: uac kamailio uas --> invite --> invite <-- 200 ok <-- 484 bye ---> Here is my reply_route: onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } if ($rs == 200) { t_reply("484", "Address Incomplete"); if (!t_cancel_callid("$ci", "$cs", "0", "200")) { xerr("t_cancel_callid failed\n"); } return; } return; } but not woking. Thanks! Han 2025-06-17

3 2

What was working Stop working, Send help
by Fernando Lopes 18 Jun '25

18 Jun '25

Hello everyone, I had this problem before https://kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio.org/t… this was fix, but suddenly with everything the same it stop working, I've tried everything and nothing seems to fix the problem Logs: INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 1048576 on fd 10 INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 1048576 for fd 11 INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 1048576 on fd 11 INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 1048576 for fd 12 INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 1048576 on fd 12 INFO: rtpengine [rtpengine.c:3040]: rtpp_test(): rtpengine instance <udp:10.0.0.4:2223> found, support for> INFO: jsonrpcs [jsonrpcs_sock.c:471]: jsonrpc_dgram_process(): a new child 0/4164 Started kamailio.service - Kamailio - the Open Source SIP Server. WARNING: <core> [core/tcp_main.c:1301]: find_listening_sock_info(): binding to source address 10.0.0.4:506> INFO: <core> [core/tcp_main.c:3097]: tcpconn_1st_send(): quick connect for 0x7ec47d7cfab8 sock 13 When I run this command I get this sudo kamailio -f /etc/kamailio/kamailio.cfg -DD -E -e loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules 0(4778) INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: 10.0.0.4 [10.0.0.4]:5060 advertise <PUBLIC IP>:5060 tcp: 10.0.0.4 [10.0.0.4]:5060 advertise <PUBLIC IP>:5060 Aliases: *: 10.0.0.4:* *: <PUBLIC IP>:* 0(4778) INFO: <core> [core/tcp_main.c:5205]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(4778) INFO: rr [../outbound/api.h:53]: ob_load_api(): unable to import bind_ob - maybe module is not loaded 0(4778) INFO: rr [rr_mod.c:185]: mod_init(): outbound module not available 0(4778) INFO: jsonrpcs [jsonrpcs_sock.c:219]: jsonrpc_dgram_mod_init(): the socket /var/run/kamailio/kamailio_rpc.sock already exists, trying to delete it... 0(4778) INFO: pike [pike.c:93]: pike_init(): PIKE - initializing 0(4778) INFO: pike [ip_tree.c:79]: init_lock_set(): probing 256 set size 0(4778) INFO: <core> [main.c:3189]: main(): processes (at least): 15 - shm size: 67108864 - pkg size: 8388608 0(4778) INFO: <core> [core/udp_server.c:163]: probe_max_receive_buffer(): SO_RCVBUF is initially 1048576 for fd 7 0(4778) INFO: <core> [core/udp_server.c:233]: probe_max_receive_buffer(): SO_RCVBUF is finally 1048576 on fd 7 0(4778) ERROR: <core> [core/tcp_main.c:3260]: tcp_init(): bind(9, 0x74711968d9c8, 16) on 10.0.0.4:5060 : Address already in use 0(4778) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized and have this sudo iptables -L -n -v | grep 5060 12 13188 ACCEPT 17 -- * * 0.0.0.0/0 0.0.0.0/0 udp dpt:5060 5 260 ACCEPT 6 -- * * 0.0.0.0/0 0.0.0.0/0 tcp dpt:5060 1 52 ACCEPT 6 -- !br-aeab3c66b5ee br-aeab3c66b5ee 0.0.0.0/0 172.19.0.5 tcp dpt:5060 0 0 ACCEPT 17 -- !br-aeab3c66b5ee br-aeab3c66b5ee 0.0.0.0/0 172.19.0.5 udp dpt:5060

3 3

RTJson current selected route
by Sergio Charrua 17 Jun '25

17 Jun '25

Hi all! doing some integrations with RTJson module for Kamailio 5.8.4, calling a REST API to return the RTJson's JSON object with 1 or multiple route elements. Kamailio converts the whole SIP headers to JSON using a Javascript function, executed by APP_JDST module, and POSTing the resulting object to the REST API via http_async module. I couldn't find a way to get the current selected route, from the RTJson object, once Kamailio iterates through the JSON object (i.e: using rtjson_push_routes or rtjson_next_route). When the RTJson module iterates through the different routes of the JSON object, is there a way to get the JSON part that defines the route currently being used by RTJson? The goal is to return some custom properties set in each route element of the route array. Once those properties are returned, Kamailio should process them. For example, RTJson supports From and To headers modification and appending new headers with the "extras" element, but there is no way that I know of to modify other headers or remove headers depending on the route. Does anyone know how: 1 - to return current route element from the routes array of the RTJson object ? 2 - to modify / remove headers other than the From and To headers, according to the custom values set in the RTJson returned from a REST API ? Thanks in advance! Atenciosamente / Kind Regards / Cordialement / Un saludo, *Sérgio Charrua*

3 4

usrloc deletes other server's data (suspected bug)
by James Browne 17 Jun '25

17 Jun '25

Hi I've two servers sharing the same postgres database, using kamailio's "server_id" value. One has server_id=184 and the other has server_id=186. I've set these parameters. modparam("usrloc", "server_id_filter", 1) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_ops_ruid", 1) # I've tested this with both 0 and 1. When a user REGISTERs, I see data being written to two tables, as normal, from server 184. update location set expires='2025-06-11 16:31:58' ... server_id=184 ...' where ruid='uloc-b8-684999a9-31e-1' delete from location_attrs where ruid='uloc-b8-684999a9-31e-1' insert into location_attrs (username,ruid,...) values (...'uloc-b8-684999a9-31e-1' ... The problem comes when I start server 186. select contact,... etc. ..., username from location where server_id=186 select username,ruid,aname,atype,avalue from location_attrs delete from location_attrs where ruid='uloc-b8-684999a9-31e-1' What I see is that server 186 always deletes the data from the location_attrs table that belongs to server 184, which I don't think it should do. According to the usrloc documentation, server_id_filter must be set (as I've done) or "Otherwise one instance of proxy cleans records made by another proxy and that breaks its cache." Therefore this looks like a usrloc bug to me. Am I doing something wrong? I'm testing with a commit that was made yesterday (kamailio 6.1.0-dev1). James

2 1

Question about how $rc works
by Joel Serrano 17 Jun '25

17 Jun '25

Hi All, I have a simple doubt: are these snippets the same? *One:* secf_check_country($hdr(X-SRC-IP-CC)); $var(secf_cc_rc) = $rc; if ($var(secf_cc_rc) == -2) { ... *Two:* $var(secf_cc_rc) = secf_check_country($hdr(X-SRC-IP-CC)); if ($var(secf_cc_rc) == -2) { ... They both seem to work, the docs say to use option -one-. In case they do the same, is there a reason I should pick one over the other? Or is this purely cosmetic? Thanks, Joel.

2 2

Re: dns_hash_put(): unlinked item errors
by JR Richardson 17 Jun '25

17 Jun '25

> > On Jun 10, 2025, at 4:00 PM, JR Richardson via sr-users <sr-users(a)lists.kamailio.org> wrote: > > > > Hey Folks, > > > > Running kamailio 5.8.4, haven't needed or used DNS functions > > previously but recently completed clearip integration, which requires > > sip request to use URL instead of IP and resolve where to send > > request. Within hours of adding URL lookups I started getting dns_hash > > unlinked errors. I correlated the log errors to when the URL changed > > IP. I assume the caching is working because kamailio sends to the same > > IP for a few minutes, then switches IP, sends to new IP for a few > > minutes, then switches IP again and this activity continues. > > > > Here are some log errors: > > > > switch IP > > Jun 10 14:14:17 siprouter1-pve186 /usr/local/sbin/kamailio[8111]: > > WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item > > 0x7fa0f9b9fcc0 > > Jun 10 14:14:19 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: > > WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): > > unlinked item 0x7fa0f9b9fcc0 > > > > switch IP > > Jun 10 14:15:00 siprouter1-pve186 /usr/local/sbin/kamailio[8111]: > > WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item > > 0x7fa0f9b9ee00 > > Jun 10 14:15:03 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: > > WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): > > unlinked item 0x7fa0f9b9ee00 > > > > switch IP > > Jun 10 14:15:33 siprouter1-pve186 /usr/local/sbin/kamailio[8110]: > > WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item > > 0x7fa0f9b8e640 > > Jun 10 14:15:33 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: > > WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): > > unlinked item 0x7fa0f9b8e640 > > > > switch IP > > Jun 10 14:20:37 siprouter1-pve186 /usr/local/sbin/kamailio[8109]: > > WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item > > 0x7fa0f9b3eb60 > > > > switch IP > > Jun 10 14:21:12 siprouter1-pve186 /usr/local/sbin/kamailio[8117]: > > WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item > > 0x7fa0f9b830f0 > > Jun 10 14:21:23 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: > > WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): > > unlinked item 0x7fa0f9b830f0 > > > > Here are my DNS config parameters, nothing I've changed made any > > difference. And as far as I can tell I'm not dropping any clearip > > queries. > > dns=no > > rev_dns=no > > use_dns_failover=on > > dns_try_ipv6=no > > > > I did find a patch update for this error back from 2023 and the patch > > is in my /src/core/dns_cache.c as expected but still getting errors. > > > > Any guidance in clearing this error? > > > > Thanks. > > JR > > > Not an answer to your question, but to use failover, use_dns_cache (I believe) needs to be on. > > https://github.com/kamailio/kamailio/blob/master/doc/tutorials/dns.txt > > Fred Posner Hi Fred, I read through the DNS docs and believe I have a good understanding, dns_cache is on my default so you don't have to explicitly define it in the config. After further diagnosis I found the dns errors "dns_hash_put(): unlinked item" happen at the same time the DNS query goes out (around 4min30sec to 5min intervals) but the errors in the log don't happen on every DNS query, they only appear when the URL IP changes. This is what I'm seeing, matching timestamps to DNS queries and SIP Invites to clearip. DNS query clearip.com IP#1 no error DNS query clearip.com IP#1 no error DNS query clearip.com IP#2 error DNS query clearip.com IP#1 error DNS query clearip.com IP#1 no error DNS query clearip.com IP#2 error DNS query clearip.com IP#2 no error DNS query clearip.com IP#2 no error DNS query clearip.com IP#1 error Because I'm not seeing any performance or SIP processing issues, I now believe the log entry is a false positive and can be disregarded, I mean it is a warning, but there is no ill effect as far as I can tell, just a lot of log messages, maybe this message should be downgraded to informational? WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item ...... WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): unlinked item ...... *****(why is this one "shm_unsafe") Thanks. JR

2 1

Kamailio 5.8.6 confusing ip protocol version of $nh(d) - where does siptrace capture the packet?
by Benoit Panizzon 17 Jun '25

17 Jun '25

Hi Gang I have come across an strange issue. To make rtpengine use the same IP protocol as signaling I examine: ip_type($nh(d)) Now I have encountered a very strange case involving Kamailio with topos and Asterisk. Initial INVITE was performed via IP6 from Asterisk to Kamailio. Then two Re-Invites were performed Kamailio => Asterisk to refresh the session. The first one happened via IPv6 and the $nh(d) ip_type was IPv6 as rtpengine got hinted to use IPv6 On the second one, something went wrong: According to what Kamailio sent to Homer via siptrace module, the Invite was sent to the IPv6 address of the Asterisk machine. rtpengine got hinted IP4, so $nh(d) must have contained an ipv4 address! But the R-URI (provided by topos) contains the IPv6 address of the Asterisk machine. The HEP capture on Asterisk did not record that Re-Invite sent from Kamailio. Neiter via IPv4 nor via IPv6. Has anyone else seen such issues? At which level does the siptrace module work? Is it capturing 'on wire' or is it capturing what kamailio believes it is sending which might be altered when sent out on the wire? 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 0

$http_rs shows 401 even when final response is 201 (http_async_query + Basic Auth)
by Joel Serrano 17 Jun '25

17 Jun '25

Hi all, I'm seeing some unexpected behavior from $http_rs when using http_async_query() to send a POST request to an endpoint that requires Basic (username/password) authentication. When I use: ``` $http_req(username) = "someuser"; $http_req(password) = "somepass"; ``` the request goes through and the server responds with a *201 Created*, as expected. The response body ($http_rb) contains the correct result. However, $http_rs still shows *401 Unauthorized*. I believe this happens because curl receives a 401 challenge first, then retries with the provided credentials and gets the correct 201. It seems $http_rs is reflecting the status from the initial challenge rather than the final response. This is on Kamailio v6.0.1. As a workaround, I tried manually setting the *Authorization* header like this: ``` $var(credentials) = "someuser:somepass"; $http_req(hdr) = "Authorization: Basic " + $(var(credentials){s.encode.base64}); ``` In that case, $http_rs correctly shows 201. Is it expected for $http_rs to stay at 401 in the first scenario? Or should it reflect the final status code after curl completes the full request cycle? Thanks, Joel.

2 1

Kamailio server not forwarding acknowledgement of 200-OK response to callee
by smh2017＠zoho.com 16 Jun '25

16 Jun '25

Problem: The Kamailio server sends an ack back to the caller UA [Frame 67 below] that it has received its ACK to callee's SIP-Request but Kamailio does not forward the caller UA response to the callee UA. Callee UA continues to send 200-OKs, waiting for an ACK that never arrives, and eventually hangs up. (Note: Blink UAs substitute number ids for user names during call transactions.) Below is some config info. WOW (192.168.1.13) = Blink caller UA registered with Kamailio server on 5060 AMD (192.168.1.11) = Blink callee UA registered with Kamailio server on 5060 Kamailio server @ (192.168.100.50) Kamailio config snippet listen=tls:192.168.100.50:6050 advertise pub-ip-domain.net:6050 listen=tls:192.168.100.50:5061 advertise pub-ip-domain.net:5061 listen=tcp:192.168.100.50:5060 advertise pub-ip-domain.net:5060 Rtpengine config snippet interface = internal/192.168.100.50;external/pub-ip-domain.net WOW and AMD are in a common LAN, Kamailio is in a separate LAN. Both LANs are behind an edge router internet firewall, with port forwarding for SIP port and designated RTP ports to the Kamailio server. The internal LANs are not firewalled from each other. A domain name is associated with the public ip address. Upon answering a call from WOW, AMD sends a 200-OK, expecting WOW to ACK. Kamailio forwards the 200-OK to WOW. The WOW UA responds to the Kamailio Server, sending its ACK [Frame 62] for forwarding, but that does not happen. [Packet 62: 200-OK ACK sent to Kamailio server for forwarding to AMD, but it is not forwarded] Frame 62: 560 bytes on wire (4480 bits), 560 bytes captured (4480 bits) on interface enp1s0, id 0 Section number: 1 Interface id: 0 (enp1s0) Interface name: enp1s0 Encapsulation type: Ethernet (1) Arrival Time: Jun 8, 2025 14:53:39.351196234 EDT [Time shift for this packet: 0.000000000 seconds] Epoch Time: 1749408819.351196234 seconds [Time delta from previous captured frame: 0.008882710 seconds] [Time delta from previous displayed frame: 0.008882710 seconds] [Time since reference or first frame: 91.952560276 seconds] Frame Number: 62 Frame Length: 560 bytes (4480 bits) Capture Length: 560 bytes (4480 bits) [Frame is marked: False] [Frame is ignored: False] [Protocols in frame: eth:ethertype:ip:tcp:sip] [Coloring Rule Name: TCP] [Coloring Rule String: tcp] .... .... = IG bit: Individual address (unicast) Type: IPv4 (0x0800) Internet Protocol Version 4, Src: 192.168.1.13, Dst: 192.168.100.50 0100 .... = Version: 4 .... 0101 = Header Length: 20 bytes (5) Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT) 0000 00.. = Differentiated Services Codepoint: Default (0) .... ..00 = Explicit Congestion Notification: Not ECN-Capable Transport (0) Total Length: 546 Identification: 0x9af3 (39667) 010. .... = Flags: 0x2, Don't fragment 0... .... = Reserved bit: Not set .1.. .... = Don't fragment: Set ..0. .... = More fragments: Not set ...0 0000 0000 0000 = Fragment Offset: 0 Time to Live: 127 Protocol: TCP (6) Header Checksum: 0x1466 [validation disabled] [Header checksum status: Unverified] Source Address: 192.168.1.13 Destination Address: 192.168.100.50 Transmission Control Protocol, Src Port: 49990, Dst Port: 5060, Seq: 5605, Ack: 5124, Len: 506 Source Port: 49990 Destination Port: 5060 [Stream index: 0] [Conversation completeness: Incomplete, DATA (15)] [TCP Segment Len: 506] Sequence Number: 5605 (relative sequence number) Sequence Number (raw): 1801511342 [Next Sequence Number: 6111 (relative sequence number)] Acknowledgment Number: 5124 (relative ack number) Acknowledgment number (raw): 2050979231 0101 .... = Header Length: 20 bytes (5) Flags: 0x018 (PSH, ACK) 000. .... .... = Reserved: Not set ...0 .... .... = Accurate ECN: Not set .... 0... .... = Congestion Window Reduced: Not set .... .0.. .... = ECN-Echo: Not set .... ..0. .... = Urgent: Not set .... ...1 .... = Acknowledgment: Set .... .... 1... = Push: Set .... .... .0.. = Reset: Not set .... .... ..0. = Syn: Not set .... .... ...0 = Fin: Not set [TCP Flags: Â•Â•Â•Â•Â•Â•Â•APÂ•Â•Â•] Window: 8212 [Calculated window size: 2102272] [Window size scaling factor: 256] Checksum: 0x7522 [unverified] [Checksum Status: Unverified] Urgent Pointer: 0 [Timestamps] [Time since first frame in this TCP stream: 91.952560276 seconds] [Time since previous frame in this TCP stream: 0.051442183 seconds] [SEQ/ACK analysis] [This is an ACK to the segment in frame: 49] [The RTT to ACK the segment was: 0.051442183 seconds] [iRTT: 0.000813889 seconds] [Bytes in flight: 506] [Bytes sent since last PSH flag: 506] TCP payload (506 bytes) Session Initiation Protocol (ACK) Request-Line: ACK sip:19350768@192.168.1.11:64901;transport=tcp SIP/2.0 Method: ACK Request-URI: sip:19350768@192.168.1.11:64901;transport=tcp Request-URI User Part: 19350768 Request-URI Host Part: 192.168.1.11 Request-URI Host Port: 64901 [Resent Packet: False] Message Header Via: SIP/2.0/TCP 192.168.1.13:49990;rport;branch=z9hG4bKPja851a221d11c4c9798e08f721afa637f;alias Transport: TCP Sent-by Address: 192.168.1.13 Sent-by port: 49990 RPort: rport Branch: z9hG4bKPja851a221d11c4c9798e08f721afa637f alias Max-Forwards: 70 From: "WOW" <sip:WOW@pub-ip-domain.net>;tag=7c6e2bc0567345288372495eb190f198 SIP from display info: "WOW" SIP from address: sip:WOW@pub-ip-domain.net SIP from address User Part: WOW SIP from address Host Part: pub-ip-domain.net SIP from tag: 7c6e2bc0567345288372495eb190f198 To: <sip:AMD@pub-ip-domain.net>;tag=347c5038ee5c41d8b92a6d473e80cf24 SIP to address: sip:AMD@pub-ip-domain.net SIP to address User Part: AMD SIP to address Host Part: pub-ip-domain.net SIP to tag: 347c5038ee5c41d8b92a6d473e80cf24 Call-ID: 7f8b4805ba4c4bc784370f99085af625 [Generated Call-ID: 7f8b4805ba4c4bc784370f99085af625] CSeq: 18468 ACK Sequence Number: 18468 Method: ACK Route: <sip:pub-ip-domain.net;transport=tcp;lr;nat=yes> Route URI: sip:pub-ip-domain.net;transport=tcp;lr;nat=yes Route Host Part: pub-ip-domain.net Route URI parameter: transport=tcp Route URI parameter: lr Route URI parameter: nat=yes User-Agent: Blink 6.0.3 (Windows) Content-Length: 0 [Packet 67: Kamailio sends acknowledgement back to WOW that it received the 200-OK ACK, but does NOT forward the ACK to AMD.] Frame 67: 54 bytes on wire (432 bits), 54 bytes captured (432 bits) on interface enp1s0, id 0 Section number: 1 Interface id: 0 (enp1s0) Interface name: enp1s0 Encapsulation type: Ethernet (1) Arrival Time: Jun 8, 2025 14:53:39.394825056 EDT [Time shift for this packet: 0.000000000 seconds] Epoch Time: 1749408819.394825056 seconds [Time delta from previous captured frame: 0.013882451 seconds] [Time delta from previous displayed frame: 0.013882451 seconds] [Time since reference or first frame: 91.996189098 seconds] Frame Number: 67 Frame Length: 54 bytes (432 bits) Capture Length: 54 bytes (432 bits) [Frame is marked: False] [Frame is ignored: False] [Protocols in frame: eth:ethertype:ip:tcp] [Coloring Rule Name: TCP] [Coloring Rule String: tcp] Internet Protocol Version 4, Src: 192.168.100.50, Dst: 192.168.1.13 0100 .... = Version: 4 .... 0101 = Header Length: 20 bytes (5) Differentiated Services Field: 0x10 (DSCP: Unknown, ECN: Not-ECT) Total Length: 40 Identification: 0xacb3 (44211) 010. .... = Flags: 0x2, Don't fragment ...0 0000 0000 0000 = Fragment Offset: 0 Time to Live: 64 Protocol: TCP (6) Header Checksum: 0x4390 [validation disabled] [Header checksum status: Unverified] Source Address: 192.168.100.50 Destination Address: 192.168.1.13 Transmission Control Protocol, Src Port: 5060, Dst Port: 49990, Seq: 5124, Ack: 6111, Len: 0 Source Port: 5060 Destination Port: 49990 [Stream index: 0] [Conversation completeness: Incomplete, DATA (15)] [TCP Segment Len: 0] Sequence Number: 5124 (relative sequence number) Sequence Number (raw): 2050979231 [Next Sequence Number: 5124 (relative sequence number)] Acknowledgment Number: 6111 (relative ack number) Acknowledgment number (raw): 1801511848 0101 .... = Header Length: 20 bytes (5) Flags: 0x010 (ACK) 000. .... .... = Reserved: Not set ...0 .... .... = Accurate ECN: Not set .... 0... .... = Congestion Window Reduced: Not set .... .0.. .... = ECN-Echo: Not set .... ..0. .... = Urgent: Not set .... ...1 .... = Acknowledgment: Set .... .... 0... = Push: Not set .... .... .0.. = Reset: Not set .... .... ..0. = Syn: Not set .... .... ...0 = Fin: Not set [TCP Flags: Â•Â•Â•Â•Â•Â•Â•AÂ•Â•Â•Â•] Window: 501 [Calculated window size: 64128] [Window size scaling factor: 128] Checksum: 0x4a97 [unverified] [Checksum Status: Unverified] Urgent Pointer: 0 [Timestamps] [Time since first frame in this TCP stream: 91.996189098 seconds] [Time since previous frame in this TCP stream: 0.043628822 seconds] [SEQ/ACK analysis] [This is an ACK to the segment in frame: 62] [The RTT to ACK the segment was: 0.043628822 seconds] [iRTT: 0.000813889 seconds] Note: When the call is reversed (AMD-to-WOW), WOW responds with 200-OK, which Kamailio forwards to AMD, but there is no ack-reponse at all from AMD to Kamailio server. WOW eventually hangs up.

1 0

help. execution in event_route takes too much time
by Сергей Приймак 16 Jun '25

16 Jun '25

- kamailio version version: kamailio 6.0.1 (x86_64/linux) fce50d - operating system where server is running: Linux Ubuntu 24 - CPU architecture of the system - 1 CPU - error log messages from syslog - no errors - if there is a coredump file, use gdb to extract the following: - information about when it happens: - - at startup - at shutdown - at runtime: - permanently ============= ISSUE DESCRIPTION =============== In case of event below, script execution takes too much time (40 sec): *event_route[dispatcher:dst-down] {exec_msg("/bin/bash -c '/usr/local/etc/kamailio/ipassign.sh &'");}* In case I run this script from shell it takes ~2 sec. ----------------------------------------------------------------------------------------------- Configs attached.

2 2

IMS to IMS communication
by ashokhs77＠gmail.com 16 Jun '25

16 Jun '25

I want to do an IMS to IMS communication - for subscriber querying. If a number dialed is not in the database of the local IMS, it should query the remote database via remote IMS and connect, if user is successfully found. Has anyone tried this? what all needs to be taken care for this..? Thanks & regards, Ashok

1 0

kamailio tls module not loading
by Timothy Delin 16 Jun '25

16 Jun '25

Hi everyone, I am facing an issue where I cannot load the TLS module in my Kamailio server. I keep receiving this error message "0(1) ERROR: <core> [core/sr_module.c:529]: ksr_locate_module(): could not find module <tls> in </usr/local/lib64/kamailio/modules> 0(1) CRITICAL: <core> [core/cfg.y:4015]: yyerror_at(): parse error in config file /etc/kamailio/modules/tls.cfg, line 1, column 12-19: failed to load module" The Kamailio that I run has this configuration version: kamailio 5.8.6 (x86_64/linux) 6c71d5 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_SEND_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: 6c71d5 compiled on 04:12:34 Jun 6 2025 with gcc 12.2.1 ------------------------------------------------------------------- #global.cfg debug=2 log_stderror=no memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ children=2 /* Defining interfaces */ #!substdef "!PUBLICIP!$env(PUBLICIP)!" #!substdef "!UDPPORT!$env(UDPPORT)!" #!substdef "!PUBLIC_UDP_SOCKET!udp:PUBLICIP:UDPPORT!" /* Need for TLS Support */ #!substdef "!PUBLIC_TLS_SOCKET!TLS:PUBLICIP:5061!" #!substdef "!COREIP!$env(COREIP)!" #!substdef "!UDPPORT!$env(UDPPORT)!" #!substdef "!CORE_UDP_SOCKET!udp:COREIP:UDPPORT!" #!substdefs "!CORESUBNET!$env(CORESUBNET)!" /* listen addresses */ listen=PUBLIC_UDP_SOCKET listen=CORE_UDP_SOCKET /* Need for TLS Support */ listen=tls:0.0.0.0:5061 #!substdef "!DBUSERNAME!$env(DBUSERNAME)!" #!substdef "!DBPASSWORD!$env(DBPASSWORD)!" #!substdef "!DBHOST!$env(DBHOST)!" #!substdef "!DBNAME!$env(DBNAME)!" #!define DBURL "mysql://DBUSERNAME:DBPASSWORD@DBHOST/DBNAME" #!define HTTP_API_ROUTING_ENDPOINT "http://172.16.254.30:5000/api/routing" ---------------------------------------------------------------- #kamailio.cfg ####### Enable TLS ######## enable_tls=yes ####### Core Modules Section ######## include_file "/etc/kamailio/modules-core.cfg" ####### Global Parameters ######### include_file "/etc/kamailio/globals.cfg" ####### Other Modules Section ######## include_file "/etc/kamailio/modules.cfg" ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ ####### Routing Logic ######## include_file "/etc/kamailio/routes.cfg"

3 2

hfl not working as well
by ale Gural 14 Jun '25

14 Jun '25

Does anyone know why using $(hfl(Diversion)[0]) cuts off the last part of the value, while hdr doesn't? Thanks!

4 10

Fwd: RFC3261 rules
by Sergey Safarov 14 Jun '25

14 Jun '25

Could you look at the attached PCAP with OPTIONS message. When OPTIONS message is received, then Kamailio generates error logs like dropping insane message from 10.140.6.38:60083 After checking the PCAP, I see extra %x00 characters after the OPTIONS message in the same UDP frame. Example [image: image.png] I have checked RFC3261 and do not see a definition of "x00" chars. Also in the same RFC, present a reference to the RFC2234 <https://datatracker.ietf.org/doc/html/rfc2234> where is described "space chars", "white space". According to RFC2223, char %x00 cannot be treated as "space" and should be treated as control (CTL) char. Also, RFC3261 does not use "CTL" chars. Using these two RFC, I can conclude, OPTIONS in the attached PCAP breaks RFC requirements, and we can request partner fix this issue on their equipment. Is anything else in the RFC can be used to justify that the given examples of OPTIONS messages violate RFCs? Sergey

5 4

rtpengine error: no rtpp_set_list->rset_first
by disa_369＠msn.com 11 Jun '25

11 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR Now Im trying setup rtpengine for proxy media . Kamailio has some errors by starting service. ERROR: rtpengine [rtpengine.c:3131]: select_rtpp_set(): no rtpp_set_list->rset_first Jun 06 12:07:00 zaks-sbc01-op /usr/sbin/kamailio[635658]: NOTICE: rtpengine [rtpengine.c:1815]: mod_init(): Default rtpp set 0 NOT found Missed out on something in config?

2 5

dns_hash_put(): unlinked item errors
by JR Richardson 10 Jun '25

10 Jun '25

Hey Folks, Running kamailio 5.8.4, haven't needed or used DNS functions previously but recently completed clearip integration, which requires sip request to use URL instead of IP and resolve where to send request. Within hours of adding URL lookups I started getting dns_hash unlinked errors. I correlated the log errors to when the URL changed IP. I assume the caching is working because kamailio sends to the same IP for a few minutes, then switches IP, sends to new IP for a few minutes, then switches IP again and this activity continues. Here are some log errors: switch IP Jun 10 14:14:17 siprouter1-pve186 /usr/local/sbin/kamailio[8111]: WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item 0x7fa0f9b9fcc0 Jun 10 14:14:19 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): unlinked item 0x7fa0f9b9fcc0 switch IP Jun 10 14:15:00 siprouter1-pve186 /usr/local/sbin/kamailio[8111]: WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item 0x7fa0f9b9ee00 Jun 10 14:15:03 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): unlinked item 0x7fa0f9b9ee00 switch IP Jun 10 14:15:33 siprouter1-pve186 /usr/local/sbin/kamailio[8110]: WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item 0x7fa0f9b8e640 Jun 10 14:15:33 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): unlinked item 0x7fa0f9b8e640 switch IP Jun 10 14:20:37 siprouter1-pve186 /usr/local/sbin/kamailio[8109]: WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item 0x7fa0f9b3eb60 switch IP Jun 10 14:21:12 siprouter1-pve186 /usr/local/sbin/kamailio[8117]: WARNING: <core> [core/dns_cache.c:170]: dns_hash_put(): unlinked item 0x7fa0f9b830f0 Jun 10 14:21:23 siprouter1-pve186 /usr/local/sbin/kamailio[8122]: WARNING: <core> [core/dns_cache.c:185]: dns_hash_put_shm_unsafe(): unlinked item 0x7fa0f9b830f0 Here are my DNS config parameters, nothing I've changed made any difference. And as far as I can tell I'm not dropping any clearip queries. dns=no rev_dns=no use_dns_failover=on dns_try_ipv6=no I did find a patch update for this error back from 2023 and the patch is in my /src/core/dns_cache.c as expected but still getting errors. Any guidance in clearing this error? Thanks. JR

2 1

get ip status from dispatcher
by ale Gural 10 Jun '25

10 Jun '25

Hello, There is a way to get the status of specific ip of dispatcher in kamailio.cfg? not using kamcmd dispatcher.list, thanks!

3 2

How to detect originating and terminating?
by James Morrison 09 Jun '25

09 Jun '25

I'm trying to find a way to differentiate between originating and terminating legs. Currently in kex module, in core_stats.c I'm adding new counters for more response codes, I want to have two sets of counters, one for originating and one for terminating. In ims_getters.c file, I found a function called cscf_has_originating, which by description "Finds if the message contains the orig parameter in the first Route header" I'm calling it like the following: static int km_cb_rpl_stats_by_method(struct sip_msg *msg, unsigned int flags, void *param) { int originating = cscf_has_originating(msg); .... However it always returns CSCF_RETURN_FALSE (or -1). I added a log message and increased log level for core module: DEBUG: <core> [ims_getters.c:561]: cscf_has_originating(): I_originating: Header Route not found ERROR: kex [core_stats.c:424]: km_cb_rpl_stats_by_method(): In kex module received rpls, the return value of cscf_has_originating is: -1 That's because cscf_has_originating function is calling parse_headers function, trying to get msg->route but that route property is not set correctly in the previous step: if(parse_headers(msg, HDR_ROUTE_F, 0) < 0) { LM_DBG("I_originating: error parsing headers\n"); return CSCF_RETURN_FALSE; } so it returns -1 after checking msg->route value. Do you know any alternative solution? or do you know why msg->route value is 0 or NULL?

3 6

siptrace module not respecting send_sock after kamailio 5.6.6
by Daniel Donoghue 09 Jun '25

09 Jun '25

Hello, I have been using the siptrace module on a multihomed kamailio setup for some time now, with kamailio 5.5.7 and 5.6.6, without incident. I am listening to my external interface using something like: listen=tls:external_ip:5061 listen=tcp:external_ip:5060 Listen=tcp:internal_ip:5000 name "homer" And for siptrace... modparam("siptrace", "duplicate_uri", "sip:homer_ip:9060;transport=tcp") modparam("siptrace", "send_sock_name", "homer") I have also used send_sock_addr and force_send... This works as expected. I recently deployed this same setup using kamailio 6.0.1 and discovered that the send_sock is not being respected, even though the logs say it has been selected. Instead, tcpdump and careful examination of all the file descriptors for the kamailio processes reveal it is sending with the source set to my external_ip instead of the internal_ip. I have verified this on the same server, ruling out os level issues routing issues and have tried manually connecting using nc with the internal ip address for the source, which works. I tried downgrading, and found that the same is true for kamailio 5.7.6, 5.8.6 and 6.0.1 This issue only appears to affect siptrace as other sockets appear to use the correct source ip to, for example, connect to a database. I had a look at the change logs and documentation and cant see anything obvious that I need to change from a configuration perspective. Could anybody help? I would prefer to be running 6.0+ in my new production environment but wont be able to if this cannot be solved quickly. Many thanks for your help. With every blessing, -- Daniel Donoghue

2 2

TCP pings from dispatcher
by Alex Balashov 07 Jun '25

07 Jun '25

Does anyone know offhand if dispatcher has access to existing TCP connections from the core for its OPTIONS pings to TCP dispatcher gateways, or whether it generates its own proprietary TCP connections for this specific purpose? -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com, https://www.csrpswitch.com Tel: +1-706-510-6800

2 1

Kamailio Deployment in AWS
by gcampos＠momenta.network 06 Jun '25

06 Jun '25

Hi community, I'm not sure if anyone has ever tried that before, but I'm trying to run Kamailio in AWS with RTPengine. I successfully installed both Kamailio and RTPengine on my EC2 instance and I'm able to make my devices register to it. However, when I make a call from one of my devices, the call is muted. As I analyze my kamailio.cfg file, I see that the "route[NATMANAGE]" is likely the cause of the mute call. I think that none of the conditions below are met, therefore, the flags are not set. Because of that, the last condition is met, thus triggering the "return" and exiting from the route[NATMANAGE] prematurely without even reaching the section where I define the RTPengine modifications. Not only that, but if I remove/comment out the "return", then the RTPengine is triggered, but there call doesn't connect. I wonder why this is happening and if there's any known workaround. # RTP relaying management and signaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO", "NATMANAGE path was followed\n"); #!ifdef WITH_NAT xlog("L_INFO", "WITH_NAT Triggered\n"); if (is_request()) { xlog("L_INFO", "is_request path was followed\n"); if (has_totag()) { xlog("L_INFO", "has_totag path was followed\n"); if (check_route_param("nat=yes")) { xlog("L_INFO", "nat=yes param detected, setting FLB_NATB\n"); setbflag(FLB_NATB); } } if (nat_uac_test("3")) { xlog("L_INFO", "nat_uac_test(3) matched, setting FLT_NATS + FLB_NATB\n"); setflag(FLT_NATS); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xlog("L_INFO", "No NAT flags set, skipping NAT handling\n"); return; } #!ifdef WITH_RTPENGINE xlog("L_INFO", "WITH_RTPENGINE Path followed\n"); if (has_body("application/sdp")) { if (is_method("INVITE")) { xlog("L_INFO", "IS_METHOD_INVITE Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } else if (is_reply()) { xlog("L_INFO", "IS_Reply Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } } #!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"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_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; }

3 4

Kamailio Asterisk Realtime integration
by cedric＠mrtn.ovh 06 Jun '25

06 Jun '25

Hello, I'm trying to use Kamailio as proxy for Asterisk. I follow this tutorial https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb I maked some updates to replace old libs and old elements related to db table and fields to use the pjsip table (ps_auths) and fields (username, password). I configured Asterisk Realtime by following the Asterisk documentation (extconfig.conf, sorcery.conf, rec_odbc.conf etc...). I'm lock at this step : The registering is handle correctly by Kamailio and forwarded to Asterisk, but asterisk respond with a "401" error and want to get a digest auth... Bellow, my kamailio config file: ``` #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ASTERISK #!define WITH_NAT #!define WITH_DEBUG # # Kamailio (OpenSER) SIP Server v4.0 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asterisk_password@localhost/asterisk" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.168.1.2:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = "192.168.1.2" desc "Asterisk IP Address" asterisk.bindport = "5080" desc "Asterisk Port" kamailio.bindip = "192.168.1.2" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else #mpath="/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "jsonrpcs.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs fifo params ----- modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "password") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # 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 to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection 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 (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { 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 t_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")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "ps_auths", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif ``` Also, I'm not sure about the configuration to add into asterisk, like a kamailio registration block or endpoint block, and... probably others... Thanks to help. Flywall.

2 1

Disable IPSec in Kamailio PCSCF
by Navjyoty Singh 05 Jun '25

05 Jun '25

Hi, I'm working on a custom IMS deployment using the pcscf module in Kamailio. For our setup, we do not want Kamailio to handle IPsec or generate Security-Server headers in 401/407 responses. Is there a clean way to disable IPsec functionality entirely in the pcscf module? Specifically, I would like to: Prevent the module from requiring IPsec setup during UE registration Avoid any modification of SIP messages related to IPsec (e.g., removing Security-Verify, Security-Client, or Security-Server headers) Use only TLS or UDP without IPsec involvement Has anyone achieved this via configuration or by disabling specific features? Any guidance or examples would be greatly appreciated. *Regards,Navjyoty*

2 1

Replace ip address of source
by disa_369＠msn.com 05 Jun '25

05 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from IPPhone ----- Avaya PBX ---sip----> Kamailio -----sip----> Video System-IVR IVR number is 3000. Avaya Prefix for routing to Kamailio is 1234. When we dial number 12343000 the call goes to Video system with Header "TO" 12343000@ip_of_PBXAvaya and call drops. How to replace ip-address of Avaya to ip-address of Video_System? What to add in kamailio.cfg? ip video_system - *.*.60.3 ip avaya *.*.20.10 My route config ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # 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); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } if ($rU==$null) { sl_send_reply("484", "Address Incomplete"); exit; } $du = "sip:*.*.60.3:5060"; route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); # enforce symmetric signaling # - send back replies to the source address of request force_rport(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); 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 request 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()) { 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; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE #!ifdef WITH_MSGREBUILD # apply changes in case the request headers or body were modified msg_apply_changes(); #!endif if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!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 (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { 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"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_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(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-Hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # JSONRPC over HTTP(S) routing #!ifdef WITH_JSONRPC event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if(src_ip!=127.0.0.1) { xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>"); exit; } if ($hu =~ "^/RPC") { jsonrpc_dispatch(); exit; } xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\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"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { 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;

5 5

Why won't this work
by Fernando Lopes 05 Jun '25

05 Jun '25

Kamailio isn’t working on my Azure VM, even though it’s set up to use port 5060. The network rules on Azure have both inbound and outbound traffic open for all protocols. I also tried using advertised_address, but that didn’t help either. It looks like no traffic is even hitting Kamailio — none of the routing blocks are triggered. Kamailio.cfg alias=<PUBLIC-IP> listen=udp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 listen=tcp:10.0.0.4:5060 advertise <PUBLIC-IP>:5060 Sip Message: 2025/06/04 15:50:37.870679 64.16.250.10:5060 -> 10.0.0.4:5060 INVITE sip:+351200600020@<PUBLIC-IP>:5060 SIP/2.0 Record-Route: <sip:64.16.250.10;r2=on;lr;ftag=DpHvv09240egQ> Record-Route: <sip:10.255.0.2;r2=on;lr;ftag=DpHvv09240egQ> Via: SIP/2.0/UDP 64.16.250.10;branch=z9hG4bKd59a.fb67bf395ea20488a7a5ebf76b9b5685.0 v:SIP/2.0/UDP 10.239.196.24:6000;received=10.239.196.24;rport=6000;branch=z9hG4bKy9ZvBZcgretvr Max-Forwards:48 f:"+351908020050"<sip:+351908020050@sip.telnyx.com>;tag=DpHvv09240egQ t:<sip:+351200600020@<PUBLIC-IP>:5060> i:122d71bb-4f3d-473f-8853-9137023f1f14 CSeq:99972326 INVITE m:<sip:mod_sofia@10.239.196.24:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:202 P-Asserted-Identity:"+351908020050"<sip:+351908020050@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1749019739 1749019740 IN IP4 50.114.147.4 s=Telnyx c=IN IP4 50.114.147.4 t=0 0 m=audio 32498 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:32499 IN IP4 50.114.147.4 a=ptime:20

6 15

Kamailio Deployment in AWS
by gcampos＠momenta.network 04 Jun '25

04 Jun '25

Hi community, I'm not sure if anyone has ever tried that before, but I'm trying to run Kamailio in AWS with RTPengine. I successfully installed both Kamailio and RTPengine on my EC2 instance and I'm able to make my devices register to it. However, when I make a call from one of my devices, the call is muted. As I analyze my kamailio.cfg file, I see that the "route[NATMANAGE]" is likely the cause of the mute call. I think that none of the conditions below are met, therefore, the flags are not set. Because of that, the last condition is met, thus triggering the "return" and exiting from the route[NATMANAGE] prematurely without even reaching the section where I define the RTPengine modifications. Not only that, but if I remove/comment out the "return", then the RTPengine is triggered, but there call doesn't connect. I wonder why this is happening and if there's any known workaround. # RTP relaying management and signaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO", "NATMANAGE path was followed\n"); #!ifdef WITH_NAT xlog("L_INFO", "WITH_NAT Triggered\n"); if (is_request()) { xlog("L_INFO", "is_request path was followed\n"); if (has_totag()) { xlog("L_INFO", "has_totag path was followed\n"); if (check_route_param("nat=yes")) { xlog("L_INFO", "nat=yes param detected, setting FLB_NATB\n"); setbflag(FLB_NATB); } } if (nat_uac_test("3")) { xlog("L_INFO", "nat_uac_test(3) matched, setting FLT_NATS + FLB_NATB\n"); setflag(FLT_NATS); setbflag(FLB_NATB); } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xlog("L_INFO", "No NAT flags set, skipping NAT handling\n"); return; } #!ifdef WITH_RTPENGINE xlog("L_INFO", "WITH_RTPENGINE Path followed\n"); if (has_body("application/sdp")) { if (is_method("INVITE")) { xlog("L_INFO", "IS_METHOD_INVITE Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } else if (is_reply()) { xlog("L_INFO", "IS_Reply Path Followed\n"); rtpengine_manage("replace-origin replace-session-connection trust-address direction=internal"); } } #!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"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_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; }

1 0

Will kamailio handle sip messages if Request contains public Ip
by Fernando Lopes 04 Jun '25

04 Jun '25

I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. The SIP message received looks like this: INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0 Will Kamailio handle this SIP message, or will it ignore it? Right now, Kamailio is not doing anything with these messages. If it’s supposed to handle them, can someone help me fix this issue?

2 6

Re: Will kamailio handle sip messages if Request contains public Ip
by airsay＠duck.com 04 Jun '25

04 Jun '25

It should. You need to add a listen and alias in the kamailio.cfg file listen=udp:1x5.2x6.1x8.1x3:5060 alias=1x5.2x6.1x8.1x3 BR, airsay > On 4 Jun 2025, at 10:46, Fernando Lopes via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > I have an Azure VM with a public IP of 1x5.2x6.1x8.1x3 and a local IP of 10.0.0.4. > The SIP message received looks like this: > INVITE sip:750046850780334@1x5.2x6.1x8.1x3:5060 SIP/2.0

2 1

adding codecs to rtpengine
by Jeff Brower 03 Jun '25

03 Jun '25

Hi all, Is there a documented procedure for adding codecs to rtpengine ? As I understand it so far, we would adapt rtpengine encoder and decoder interfaces to the voplib API (documented at https://github.com/signalogic/SigSRF_SDK/blob/master/codecs_readme.md), allowing use of voplib shared object libs. I've been searching (e.g. adding codecs to rtpengine site:kamailio.org) but have not found a step-by-step procedure yet. Thanks for any advice / guidance. -Jeff

3 2

event_route[xhttp:request] is not executed
by Володимир Іванець 03 Jun '25

03 Jun '25

Hello! I'm doing some testing with http requests but got stuck at the beginning. event_route[xhttp:request] does not seem to be executed. I'm getting the following in the log with the most basic configuration. Am I missing something? Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): tcpconn_new: new tcp connection: 127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1238]: tcpconn_new(): on port 46328, type 2, socket 23 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:1630]: tcpconn_add(): hashes: 1391:1425:2615, 2 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:367]: io_watch_add(): processing io_watch_add(0xb22d80, 23, 2, 0x7f7a63877f30) - fd_no=17 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xb22d80, 23, -1, 0x0) fd_no=18 called Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4671]: handle_tcpconn_ev(): sending to child, events 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:4335]: send2child(): selected tcp worker idx:1 proc:10 pid:9158 for activity on [tcp:127.0.0.1:8089], 0x7f7a63877f30 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1782]: handle_io(): received n=8 con=0x7f7a63877f30, fd=6 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:627]: tcp_read_headers(): no clen, p=A Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) ERROR: <core> [core/tcp_read.c:1555]: tcp_read_req(): bad request, state=7, error=4 buf: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: parsed: Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: GET /TEST HTTP/1.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Host: 127.0.0.1:8089 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: User-Agent: curl/7.43.0 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: Accept: */* Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1562]: tcp_read_req(): received from: port 46328 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/ip_addr.c:577]: print_ip(): received from: ip127.0.0.1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1704]: release_tcpconn(): releasing con 0x7f7a63877f30, state -2, fd=6, id=2 ([127.0.0.1]:46328 -> [127.0.0.1]:8089) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 10(9158) DEBUG: <core> [core/tcp_read.c:1705]: release_tcpconn(): extra_data (nil) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3745]: handle_tcp_child(): reader response= 7f7a63877f30, -2 from 1 Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3668]: tcp_emit_closed_event(): TCP closed event creation triggered (reason: 0) Jun 3 20:50:35 kamailio-dev-2 kamailio[9148]: 13(9161) DEBUG: <core> [core/tcp_main.c:3676]: tcp_emit_closed_event(): no callback registering for handling TCP closed event #!KAMAILIO children = 4 tcp_children = 4 debug=3 log_stderror=yes memdbg=5 memlog=5 fork=yes #!substdef "!LOCAL_UDP!udp:127.0.0.1:5060!g" #!substdef "!LOCAL_HTTP!tcp:127.0.0.1:8089!g" listen=LOCAL_UDP listen=LOCAL_HTTP loadmodule "xlog.so" loadmodule "pv.so" loadmodule "ctl.so" loadmodule "sl.so" loadmodule "xhttp.so" event_route[xhttp:request] { xhttp_reply("200", "OK", "text/html", "<html><body>OK - [$si:$sp]</body></html>"); } Thank you very much!

2 1

Kamailio Dispatcher for Docker Asterisk
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello everyone, I have Kamailio running on port 5060, and I'm running Asterisk inside a Docker container. I'm trying to connect to Asterisk via port 5061. Could someone please advise what I should add to my dispatcher list? Currently, I have the dispatcher.list entry: 1 sip:10.0.0.5:5061;transport=tcp 0 Here’s the network info for my Kamailio host (eth0 interface): inet 10.0.0.4/24 brd 10.0.0.255 scope global eth0 And here are the relevant port mappings from Docker: 5038/tcp -> 0.0.0.0:5038 5060/tcp -> 0.0.0.0:5061 5060/udp -> 0.0.0.0:5061 ... My main question: Is the dispatcher entry sip:10.0.0.5:5061;transport=tcp correct if Asterisk is listening on port 5061 in the container? Do I need to adjust anything on the Kamailio side or change the dispatcher entry format? Thanks in advance for any help!

3 5

Small Question
by Fernando Lopes 03 Jun '25

03 Jun '25

Hello Everyone, I have a cloud vm with public ip: 135.2xx.1xx.1xx:5060 and locally 10.0.0.4 When I receive a SIP message the invite looks like this, it gets stuck here. SIP Invite: 2025/06/03 16:24:58.249284 192.76.120.10:5060 -> 10.0.0.4:5060 INVITE sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060 SIP/2.0 Record-Route: <sip:192.76.120.10;r2=on;lr;ftag=765jrrKarBe3F> Record-Route: <sip:10.255.0.1;r2=on;lr;ftag=765jrrKarBe3F> Via: SIP/2.0/UDP 192.76.120.10;branch=z9hG4bKbb37.28332dbc5901e729308403cd5e049713.0 v:SIP/2.0/UDP 10.239.44.224:6000;received=10.239.44.224;rport=6000;branch=z9hG4bKD4UtjN52Qrm9D Max-Forwards:55 f:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.com>;tag=765jrrKarBe3F t:<sip:+35122xxxxxxxx@135.2xx.1xx.1xx:5060> i:c64314db-75b9-4029-8147-be3f3d77da2a CSeq:99930157 INVITE m:<sip:mod_sofia@10.239.44.224:6000> Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REFER,NOTIFY k:timer,path u:talk,hold,conference,refer Privacy:none c:application/sdp Content-Disposition:session l:205 P-Asserted-Identity:"+3519xx78xx1x"<sip:+3519xx78xx1x@sip.telnyx.eu;verstat=No-TN-Validation> v=0 o=Telnyx 1748940128 1748940129 IN IP4 50.114.150.25 s=Telnyx c=IN IP4 50.114.150.25 t=0 0 m=audio 27770 RTP/AVP 9 a=rtpmap:9 G722/8000 a=sendrecv a=rtcp:27771 IN IP4 50.114.150.25 a=ptime:20 Kamailio Code: #!define DS_LIST "/etc/kamailio/dispatcher.list" #!define LISTEN_UDP_PUBLIC udp:135.2xx.1xx.1xx:5060 #!define LISTEN_UDP_PRIVATE udp:10.0.0.4:5060 #!define LISTEN_TCP_PUBLIC tcp:135.2xx.1xx.1xx:5060 #!define LISTEN_TCP_PRIVATE tcp:10.0.0.4:5060 #!define EXTERNAL_IP "10.0.0.4" #!define KUBERNETES_NETWORK "172.19." #!define FLAG_TELNYX_CALL 12 #set module path mpath = "/usr/lib/x86_64-linux-gnu/kamailio/modules" ####### Global Parameters ######### #mhomed=1 # doesn't work with virtual IPs #!ifdef WITH_DEBUG debug=4 #!else debug=2 #!endif log_stderror=no memdbg=5 memlog=5 #log_facility=LOG_LOCAL6 log_facility=LOG_LOCAL0 fork=yes children=4 auto_aliases=no #listen=LISTEN_UDP_PUBLIC listen=LISTEN_UDP_PRIVATE listen=LISTEN_TCP_PRIVATE ####### FLAGS ###### #!define FLAG_FROM_ASTERISK 10 #!define FLAG_FROM_PEER 11 sip_warning=no

1 0

Kamailio as Proxy only
by disa_369＠msn.com 03 Jun '25

03 Jun '25

Hello to everyone. I deployed Kamailio on Debian system. Install was successfull. I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. Outgoing calls only from Avaya PBX -------> Kamailio ----> Video System. We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. I got questions. 1. Do we need install maria_db for that? 2. Do we need create subscriber\peer on Kamailio? 3. Do we need add modules for that? And main question what to add in kamailio.cfg is? Thx!!!!!!!!!

2 2

Kamailio Regex
by Timothy Delin 03 Jun '25

03 Jun '25

Hi everyone, I need your advice. I am extracting a SIP Body message from a User Agent Client using the regex as such "$var(calling_party)= $(var(body){re.subst,/.*Calling-party:\s*sip:(\+[0-9]+)@.*/\1/});" And I tried to log it as "xlog("L_INFO", "Extracted number 1: $var(calling_party) <><><>\n");" However, the output in my logs is only showing as below, where I cannot see any value after my variables. "1(6) INFO: {1 10 SUBSCRIBE 5017ae21204c72a3} <script>: Extracted number 1: +6512345678" I suspect this is the reason why when I tried to use this variable in a redis_cmd, it returns as null as the variable may have impacted the command input to Redis. I tried to change the $var(calling_party)="+6512345678" and it is working perfectly fine Best regards, Timothy

3 5

Re: Kamailio as Proxy only
by airsay＠duck.com 30 May '25

30 May '25

I'm not very skilled with Kamailio config. But it sounds like dsiprouter (a graphical UI built ontop of Kamailio) would be your easiest solution. On Dsiprouter, You can set up Avaya as a Carrier Group and Video conference as an Endpoint Group. Then route accordingly. Sent from my iPhone > On 30 May 2025, at 12:19, disa_369--- via sr-users <sr-users_at_lists.kamailio.org_airsay(a)duck.com> wrote: > > Hello to everyone. > I deployed Kamailio on Debian system. Install was successfull. > I have not much experience Kamailio and i dont know what need to run Kamailio as Proxy. > I have the task to use Kamailio as proxy between Avaya PBX and Videoconference System. > Outgoing calls only from > Avaya PBX -------> Kamailio ----> Video System. > We dont need Kamailio subscribers\peers, Just to forward a call from avaya to Video system by ip. > Video system has IVR number 3000, all we need to dial number 3000@ip_of_video_system. > I got questions. > 1. Do we need install maria_db for that? > 2. Do we need create subscriber\peer on Kamailio? > 3. Do we need add modules for that? > > And main question what to add in kamailio.cfg is? > > Thx!!!!!!!!! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- sr-users(a)lists.kamailio.org > To unsubscribe send an email to sr-users-leave(a)lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the sender!

1 0

Hyperthreading and high request volumes in modern times
by Brooks Bridges 30 May '25

30 May '25

So, I have a platform that handles in the order of around 2500 or more call attempts per second. There are 2 servers that are built that handle this traffic, "old" and "new". The "old" design is a bit heavy, has multiple redis calls to external IO, does a lot of "work" on each invite before passing it along, etc. The "new" design is extremely lightweight, only a couple of htable comparisons and a dispatcher check on each invite before passing it along. The 2 servers are running different versions of Kamailio, however they're both in the 5.x train and I've found nothing in the changelogs that I believe would explain a significant difference in how child process CPU scheduling is performed, especially a detrimental change. I'm more than happy to be proven wrong about that though! The "old" system is running 5.4.8: This system is running 4 x 8 core CPUs (32 cores) with no hyperthreading (4 x E5-4627 v2 to be exact) and has multiple external IO interactions with things like redis, lots of "work" being done on each invite checking various things before allowing it to proceed. This is currently running 32 child processes and again, has no issues handling the requests whatsoever and has been running at this level and higher for literal years. The "new" system is running 5.8.3: This system is running 2 x 8 core CPUs (16 cores) *with* hyperthreading (2 x E5-2667 v4 to be exact) and has no external IO interactions with anything, and only does a couple of checks against some hash table values and dispatcher before allowing it to proceed. This is also currently running 32 child processes, however I am experiencing spikes in context switching and odd memory access issues (e.g. backtraces show things like "<error: Cannot access memory at address 0x14944ec60>" in udp_rvc_loop) with similar loads. What I'm looking for here is empirical knowledge about things specifically like context switching and if hyperthreading is detrimental in cases where I'm doing very high volume, low latency (e.g. no waiting on external replies, etc) work. Is it possible there are additional issues with the way hyperthreading works and the concepts behind it if the individual calls for CPU resources are so "quick" as to overwhelm the scheduler and cause congestion in the CPU's thread scheduler itself? Have you run into anything like this? Have you discovered that very low latency call processing can incur more hyperthreading overhead than the benefit you get from it? Do you have additional data I may be missing to help me justify to the C suite to change out the system architecture to provide a higher physical core count? Can you share it? Thanks!

6 14

Sca in cluster/load balancer
by mauro.celli76＠gmail.com 29 May '25

29 May '25

We have two kamailio in load balancing, mysql is in cluster mode with galera. We need to use sca, but kamailio db_cluster is not supported (i need to specify connection string to sql not cluster name) and if phone1 is registered on kamailio1 and phone2 on kamailio2, sca not work. It is possibile to implement cluster?

3 5

How to count the first re-register request?
by James Morrison 26 May '25

26 May '25

I want to count the first REGISTER request during an initial register or re-register from UE. In my scscf instance, I have defined att_init_reg and att_rereg counters to count the number of initial registers and re-registers respectively. In my early tests, the following piece of logic worked as expected to filter the very first register request (one that is sent from UE before getting challenged). However, more tests showed that during re-registers, UE sends a REGISTER request, containing an authorization header and non-empty response, in that case the code is unable to distinguish a re-register attempt and the related counter fails to increment. Is there a way to fix this issue? #!ifdef WITH_XHTTP_PROM /* Only count the first register request */ if (!is_present_hf("Authorization") || search_hf("Authorization","response=\"\"","f")) { if ($var(impu_registered) != 1) { prom_counter_inc("att_init_reg", "1"); } else { prom_counter_inc("att_rereg", "1"); } } #!endif

3 5

5.7.7 release, tag and docker build
by Henning Westerholt 23 May '25

23 May '25

Hello, I have noticed that the git tag for release 5.7.7 was missing, I've just set it. The 5.7.7-bullseye docker image is also missing from https://github.com/kamailio/kamailio-docker/pkgs/container/kamailio Not sure if this is done automatically, if not - please can somebody please trigger the workflow to create it? Thank you, Henning

1 0

Help Adding WebRTC to Existing SIP Setup
by Fernando Lopes 22 May '25

22 May '25

Hi everyone, I currently have a working SIP setup, but I'm trying to get WebRTC working alongside it and haven’t had much success so far. Does anyone have tips or best practices for enabling both SIP and WebRTC in the same configuration? Is it possible to handle both within a single Kamailio config? Or would I need some kind of conditional logic (like if/else) to manage how the config handles different types of clients? My setup looks like this: Router → Kamailio → Kubernetes (Asterisk inside the cluster) I was having some issues with the Contact header, but I managed to resolve that. Now I'm stuck figuring out how to make WebRTC work cleanly alongside SIP. Any advice would be appreciated! Thanks!

2 1

Kamailio v5.7.7 Released
by Daniel-Constantin Mierla 21 May '25

21 May '25

Hello, Kamailio SIP Server v5.7.7 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.7. Note that 5.7 is now the third last stable branch, v5.7.7 being the last planned released in 5.7.x series. The latest two stable branch are 5.8 and 6.0, with v6.0.1 being released a while ago. For more details about version 5.7.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2025/05/kamailio-v5-7-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 0

Releasing v5.7.7
by Daniel-Constantin Mierla 21 May '25

21 May '25

Hello, I am considering to release Kamailio v5.7.7 (out of branch 5.7) on Wednesday, May 21, 2025, as the last one from the 5.7.x series, to mark the end of support for it. Currently 5.8.x and 6.0.x are the latest two maintained branches. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com

1 1

Re: stir/shaken integration with clearip redirect method
by JR Richardson 20 May '25

20 May '25

I appreciate all the feedback Fellas, checking the docs, there is no need to change the RURI, they do what some E164 fixup presented without the +. I'll follow up after testing, got a bit to chew one here. Thanks! JR Richardson

1 0

stir/shaken integration with clearip redirect method
by JR Richardson 19 May '25

19 May '25

Hey Folks, I'm integrating STIR/SHAKEN using kamailio, but not using self hosted certs with the secsipid module, just using a redirect from clearip on the SIP Calls. I've got the basic connection to clearip using t_relay_to_tcp for testing, but I'm struggling with the checks and timeouts. I have call progress past the initial request where I need to wait for a response. Should I put the clearip invite in its own route block use return? Has anyone successfully integrated the redirect method with kamailio to clearip and can give me some example route checks in the kamailio configs? Thanks. JR

4 6

Server closing TLS connection
by Ihor Olkhovskyi 16 May '25

16 May '25

Hello, I'm having a strange issue, when Kamailio(?) is closing TLS connection just in the middle of session. Usually I see a sort of timeout ~ 600 secs. But it does not ring a bell for me. With strace I've found, that it's actually not Kamailio, but underlying SSL library is doing this, according to line I've got from strace [2025-05-16 10:00:30] PID=1337952 FD=0 SYSCALL=close -> [No socket info] Raw: sendto(3, "<183>May 16 10:00:30 /usr/local/sbin/kamailio[1337952]: DEBUG: tls [tls_server.c:712]: tls_h_tcpconn_close_f(): Closing SSL connection 0x7f01a1ab1870\n", 150, MSG_NOSIGNAL, NULL, 0) = 150 As I don't see any explicit calls of tls_h_tcpconn_close_f in the source code, I assume it's a callback from the external OpenSSL(?) module. What would be the best way to debug why OpenSSL(?) is terminating connection, cause on Wireshark I just do see usual TCP connection with keepalives [image: image.png] kamailio -v version: kamailio 5.8.3 (x86_64/linux) 6f8a04 flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_SEND_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. openssl --version OpenSSL 3.2.2 4 Jun 2024 (Library: OpenSSL 3.2.2 4 Jun 2024) enable_tls=yes enable_sctp=no tls_threads_mode=2 tcp_accept_no_cl=yes tcp_max_connections=63536 tls_max_connections=63536 tcp_accept_aliases=no tcp_async=yes tcp_connect_timeout=30 tcp_conn_wq_max=63536 tcp_crlf_ping=yes tcp_delayed_ack=yes tcp_fd_cache=yes tcp_keepalive=yes tcp_keepcnt=5 tcp_keepidle=30 tcp_keepintvl=10 tcp_linger2=30 tcp_script_mode=1 tcp_rd_buf_size=80000 tcp_send_timeout=10 tcp_wq_blk_size=2100 tcp_wq_max=10485760 open_files_limit=63536 Many thanks in advance! -- Best regards, Ihor (Igor)

1 1

extract_candidate() errors on 6.0.0
by Arsenijs Kabihno 14 May '25

14 May '25

Hello, I am getting the following errors on kamailio 6.0.0: May 14 07:10:20 hostname.xxxxxx.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' May 14 07:10:20 hostname.xxxxxx.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' May 14 07:10:20 ap07stvp.dyninno.net docker[20678]: 53(59) ERROR: {2 24186 INVITE 7c127e81-45a8-41bb-ab64-7b0e7223f7ff} <core> [core/parser/sdp/sdp_helpr_funcs.c:336]: extract_candidate(): no component in `a=candidate' Candidates in SDP looks valid per RFC: a=candidate:vnyxKH3f79Lw0zSj 1 UDP 2130706431 1x.xxx.xxx.xxx 26010 typ host a=candidate:vnyxKH3f79Lw0zSj 2 UDP 2130706430 1x.xxx.xxx.xxx 26011 typ host a=remote-candidates:1 9x.xx.xxx.xxx 56628 I am using the 6.0.0-alpine version (https://github.com/kamailio/kamailio-ci/pkgs/container/kamailio-ci/34563256…) version: kamailio 6.0.0 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_SEND_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: unknown compiled on 12:27:58 Jan 29 2025 with gcc 14.2.0 It appears that it could have been fixed in 6.0.1 (https://www.kamailio.org/pub/kamailio/6.0.1/ChangeLog commit a5ac9e04b91971da44256373904450501f9a4493) but I don't see 6.0.1-alpine image in https://github.com/kamailio/kamailio-ci/pkgs/container/kamailio-ci. Please advise, is there any plans to publish it? Kind regards, [logo] Arsenijs Kabihno | VoIP SIA "Dynatech", DYNATECH, Jeruzalemes St. 1, Riga, LV-1010, Latvia<https://www.google.com/maps/place/Jeruzalemes+St.+1,+Riga,+LV-1010,+Latvia> a.kabihno(a)dynatech.lv<mailto:a.kabihno@dynatech.lv> | https://dynatech.lv/<https://https://dynatech.lv/> Stay connected: [logo]<https://www.linkedin.com/company/dynatech-latvia/> Confidentiality Notice: This message contains confidential information and is intended only for the named recipient(s). If you are not the addressee you may not copy, distribute or perform any other activities with this information. If you have received this transmission in error, please notify us by e-mail immediately. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

How to remove Reason HF from branched CANCEL?
by Benoit Panizzon 13 May '25

13 May '25

Hi Gang I try to remove the Reason Header from a CANCEL message. I use parallel branching on the registrar. The route below is called very soon in request_route. The cancel matches as I get the log entry just before calling remove_hf("Reason") But the Reason HF is still present when the CANCEL is forked to the registered CPE. The INVITE is routed through a branch_route after forking where headers can be altered per branch. But I don't see the CANCEL in the branch route. Is there a way to alter header in a branched cancel message? # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(FILTER_CANCEL); route(RELAY); } exit; } route[FILTER_CANCEL] { # Buggy iPhones send wrong cancel cause code 200. if (is_present_hf("Reason")) { $var(cause) = $(hdr(Reason){param.value,cause}); $var(text) = $(hdr(Reason){param.value,text}); if (($var(cause) == 200) && ($var(text) == "Call Rejected By User")) { if ($avp(debug) > 0) { xlog("L_INFO", "$cfg(route): Remove wrong iPhone Cancel Reason HF\n"); } remove_hf("Reason"); } } } 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

xhttp module generates an error sip parsing error log
by 崔天浩 13 May '25

13 May '25

### Description I see that when the xhttp module is enabled, kamailio will forge the Via header to process the HTTP request, but the parse sip error will be generated in the log ``` 104(113) ERROR: <core> [core/parser/parse_uri.c:1417]: _parse_ruri(): bad uri </metrics> 104(113) ERROR: <core> [core/parser/parse_uri.c:1455]: parse_orig_ruri(): parse orig ruri failed 104(113) ERROR: pv [pv_core.c:355]: pv_get_ouri_attr(): failed to parse the R-URI ``` ### Troubleshooting #### Debugging Data Set up a Kamailio server with the xhttp module enabled. ``` loadmodule "xhttp.so" event_route[xhttp:request] { $var(xhttp_prom_root) = $(hu{s.substr,0,8}); if ($var(xhttp_prom_root) == "/metrics") { xlogl("L_INFO", " [$utimef(20%y-%d-%m %H:%M:%S)] [$(hdr(Call-ID))]: [bizCall=$xavu(bizCallId)] [bot=$xavu(bizBotId)] Called metrics\n"); prom_dispatch(); } } ``` #### Log Messages ``` 104(113) ERROR: <core> [core/parser/parse_uri.c:1417]: _parse_ruri(): bad uri </metrics> 104(113) ERROR: <core> [core/parser/parse_uri.c:1455]: parse_orig_ruri(): parse orig ruri failed 104(113) ERROR: pv [pv_core.c:355]: pv_get_ouri_attr(): failed to parse the R-URI ``` ### Additional Information ``` root@kamailio-60-bot-67d564c54c-xj97x:/# kamailio -V version: kamailio 6.0.1 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, MEM_JOIN_FREE, 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_SEND_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: unknown compiled with gcc 8.3.0 ``` * **Operating System**: ``` root@kamailio-60-bot-67d564c54c-xj97x:/# uname -a Linux kamailio-60-bot-67d564c54c-xj97x 5.4.119-19.0009.28 #1 SMP Thu May 18 10:37:10 CST 2023 x86_64 GNU/Linux ```

1 0

Is there an mistake in the dialog documentation?
by 13715209697＠139.com 12 May '25

12 May '25

Hi, team https://github.com/kamailio/kamailio/blob/master/src/modules/dialog/doc/dia… `dlg.list_match furi eq sip:alice@test.com 2` ? `dlg.list_match furi eq sip:alice@test.com` ?

2 2

Analysis regarding performance regressions OpenSSL 3.0/3.x
by Henning Westerholt 11 May '25

11 May '25

Hello, If somebody is using Kamailio in a larger environment with TLS, this new technical analysis and performance report from haproxy could be interesting: https://www.haproxy.com/blog/state-of-ssl-stacks The bottom-line - OpenSSL 3.0 will show serious performance regressions for larger TLS services with higher performance requirements. OpenSSL 3.1 and newer versions are a bit better, but still much slower in key operations. You should consider using the tls_wolfssl module or stay on OpenSSL 1.1.1. For the medium- to long-term we probably should observe how other OpenSSL libraries are developing and act accordingly for the tls modules. The haproxy project recommends besides using wolfssl also the aws-lc library. The situation regarding OpenSSL 3.x seems to be not easily fixable, as these regressions are caused from internal design decisions. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

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