- sr-users - kamailio.org

please help to configure tls in kamailio for webrtc client like simpl5
by ThanhTruong 19 Jul '21

19 Jul '21

Dear, I am trying to make a test to see how can webrtc work in kamailio and tls. I was try some link : 1 <https://github.com/havfo/WEBRTC-to-SIP>. https://github.com/havfo/WEBRTC-to-SIP <https://github.com/havfo/WEBRTC-to-SIP> 2. https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/ <https://telecom.altanai.com/2018/09/04/kamailio-webrtc-sip-server/> And Generate CA from the link: https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates <https://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates> Then, I tried to login with simpl5 and i got the message back Disconnected: Failed to connet to the server I do not know how can i fix that issue, look like TLS configure has issue with that. the log is showing something like Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1283]: tls_h_read_f(): protocol level error Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094416:SSL routines:ssl3_read_bytes:sslv3 alert certificate unknown Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1287]: tls_h_read_f(): source IP: 27.65.214.194 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: tls [tls_server.c:1290]: tls_h_read_f(): destination IP: 172.31.44.170 Jul 14 17:56:16 ip-172-31-44-170 sbin/kamailio[13813]: ERROR: <core> [core/tcp_read.c:1493]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f4233c60fc0 r: 0x7f4233c610e8 (-1) My tls.cfg is : [server:default] method = SSLv23 verify_certificate = yes require_certificate = yes private_key = /etc/certs/my.domain/key.pem certificate = /etc/certs/my.domain/cert.pem ca_list = /etc/certs/demoCA/cert.pem [client:default] verify_certificate = yes require_certificate = yes Please help me to register sipml5 or how can i create CA for TLS in kamailio to use with webrtc. Thank all.

4 14

Active Profile for sipreply in SEMS
by Mojtaba 17 Jul '21

17 Jul '21

Hello, Does anybody know how i could change white filter in siprelpy in sems? In Kamailio, I got "200 OK" from UE, then add a new header (SBC: mt-reply) and send it to sems server. The sems server remove it: [#7f410fbfb700/4560] [inplaceHeaderFilter, HeaderFilter.cpp:226] DEBUG: erasing header 'sbc' by whitelist I really don't want to remove it. Actually if I knew how i active profile for sip replying in sems, I would be able to solve this. Thanks with regards. -- --Mojtaba Esfandiari.S

2 2

get defined names in kemi script
by Riccardo Villa 16 Jul '21

16 Jul '21

Hello guys, do you know if it’s possible for a kemi script to know if a name is defined (with #!define) in kamailio.cfg? It would be useful to modify kemi script logic depending on feature/module presence. Thanks in advance for your help. Regards, Riccardo -- -- NOTICE: This electronic mail transmission may contain confidential information and is intended only for the person(s) named. Any use, copying, or disclosure by any other person is strictly prohibited. If you have received this transmission in error, please notify the sender via e-mail.

3 6

rtpengine play_media
by Дмитрий Плаксюк 15 Jul '21

15 Jul '21

Greetings. I'm trying to set up annoncement playback for both parts of the call. I have to play the file after the connection is established. Stop_media (all) must be executed before playing It turns out the construction: .... block_media ("all"); play_media ("all file = / etc / kamailio / sounds / mg1.wav"); Can't find a way to do unblock_media ("all"); after the file has finished playing. There are no sip messages in this process. The call has already been established.

1 0

DMQ node list
by Aidar Kamalov 14 Jul '21

14 Jul '21

Hello, I have a kamailio cluster with two nodes kamailio-1.local: 10.101.2.119 kamailio-2.local: 10.101.2.247 with dmq configs: kamailio-1: modparam("dmq", "server_address", "sip:MY_IP4_ADDR:5060") modparam("dmq", "notification_address", "kamailio-2.local:5060") kamailio-2: modparam("dmq", "server_address", "sip:MY_IP4_ADDR:5060") modparam("dmq", "notification_address", "kamailio-1.local:5060") All works fine, but kamcmd dmq.list_nodes on kamailio-1.local show me: { host: 10.101.2.247 port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: kamailio-1.local port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: kamailio-2.local port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: 10.101.2.119 port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 1 } and kamailio-2.local show me: { host: kamailio-2.local port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 0 } { host: 10.101.2.119 port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: kamailio-1.local port: 5060 resolved_ip: 10.101.2.119 status: active last_notification: 0 local: 0 } { host: 10.101.2.247 port: 5060 resolved_ip: 10.101.2.247 status: active last_notification: 0 local: 1 } So there are only two hosts in the cluster, but there are 4 records in dmq.list_nodes. Also in sip trace I can see that kamailio send DMQ messages to itself too (because it ip marked as local: 0 I think). Is it ok? Is it normal behavior? What should I do to fix it? Should multi_notify and DNS name with 2 A records fix it (and why?)? -- Aydar A. Kamalov

2 1

Re: [SR-Users] Subscriber Management using JSONRPC over HTTP
by Surabhi Garg 14 Jul '21

14 Jul '21

Hi Alex, By subscriber management I mean, auth_db module which actually works on subscriber table to add or alter sip account for authentications. Auth_db Module (kamailio.org)<https://kamailio.org/docs/modules/4.4.x/modules/auth_db.html#auth_db.f.auth…> So, similar like usrloc module which you mentioned having JSONRPC commands to alter Location table. I am looking for JSONRPC Command for subscriber module as well. As far as my understanding, currently there is no JSONRPC command available for subscriber table. So, if this is currently a known not offered feature / limitation shall we raise it as feature request if others also agree to its usefulness. What I'm seeking is more of an acknowledgment that this feature doesn't exist & have it's benefit. As I can see other community members either ends up directly interacting with DB for subscriber table provisioning. Maybe I can contribute on this too. Surabhi Garg Mobile - 91 783 888 7523 | Skype - surabhi.garg9 ________________________________ From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Alex Balashov <abalashov(a)evaristesys.com> Sent: 09 July 2021 01:27 To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Subscriber Management using JSONRPC over HTTP I am not 100% sure what you mean by subscriber management, but if you mean what I think you mean, these might be the droids you’re looking for: https://kamailio.org/docs/modules/5.5.x/modules/usrloc.html#idm771 — Sent from mobile, with due apologies for brevity and errors. On Jul 8, 2021, at 3:55 PM, Surabhi Garg <surabhi.garg(a)outlook.com> wrote: Hi Team, Greetings! I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP. Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well. curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" Surabhi Garg Mobile - 91 783 888 7523 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 2

DMQ_USRLOC module and calls
by Social Boh 13 Jul '21

13 Jul '21

Hello, I'm testing DMQ_USRLOC modulo between two Kamailio servers (A) and (B) The two servers share the same domain and using Amazon Route53 to distributes 50% requests each Server. The problem is when I try to make a call from user A (registered on kamailio A) and the user B (registered on Kamailio B). The call follow this flow: User A --> KamailioA --> UserB go directly to userB because KamailioA know, via DMQ_USRLOC IP and port of UserB. With some Softphone the call work with audio without problem; with other Softphone USERB never answer the INVITE sends from Kamailio. Is it possible use this flow? USER A --> KamailioA --> KamailioB --> User B to resolve this kind of problem or is there other available option? Thank you -- --- I'm SoCIaL, MayBe

5 15

Change routing for Register
by H Yavari 13 Jul '21

13 Jul '21

Hi to all, In a scenario, we need to change the destination of users' registration requests when they send the first 'Register' to Kamailio-A and it replies 401 to the user, next time the user sends 'Register' to Kamailio-B. User------Register-------> Kamailio-A User<-----401--------------Kamailio-A User-------Register--------> Kamailio-B RFC says we can't use the Record-Route for register because it doesn't create a dialog.Is there any option, header that Kamailio-A puts in the 401, and after that users send register to the Kamailio-B? Thank you.BR,Hossein

3 3

Kamailio v5.4.6 Released
by Daniel-Constantin Mierla 13 Jul '21

13 Jul '21

Hello, Kamailio SIP Server v5.4.6 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.4.x. Deployments running previous v5.4.x versions are strongly recommended to be upgraded to v5.4.6. Note that 5.4 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.5, with v5.5.0 being release out of it. For more details about version 5.4.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-4-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 2

Using the dialog module. dialog was not destroyed with BYE message
by woenho＠gmail.com 08 Jul '21

08 Jul '21

According to the documentation a dialog is automatically destroyed when a BYE is received. But, dialog was not destroyed. What can I do? I am testing with kamailio 5.5 version

3 2

Subscriber Management using JSONRPC over HTTP
by Surabhi Garg 08 Jul '21

08 Jul '21

Hi Team, Greetings! I was trying to use HTTP to do the subscriber management (CRUD operations) & for that I am using JSONRPC-S module (JSONRPC over HTTP). But like other modules e.g. dispatcher, location I am unable to find any RPC commands for doing subscriber management (checked in auth_db) module. So is there a way to achieve subscriber management using JSONRPC over HTTP. Few HTTP examples which I am using for dispatcher module are given below. Similar commands I am looking for subscriber management as well. curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.add\", \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" curl http://localhost:8081/RPC -H "Content-Type: application/json" -d " {\"jsonrpc\":\"2.0\",\"method\":\"dispatcher.remove\”, \"params\":[2, \"sip:127.0.0.1:5061\"], \"id\": \"1\"}" Surabhi Garg Mobile - 91 783 888 7523

2 1

How to check If Alias is already Set in Contact Header
by Juha Heinanen 08 Jul '21

08 Jul '21

Transformations (e.g. nameaddr.uri and uri.param.name) on $ct are your friends. -- Juha

2 2

How to check If Alias is already Set in Contact Header
by Terry Tailor 08 Jul '21

08 Jul '21

Hi, I'm having the following scenario UA <---> Prox1 <----> Proxy2 <-----> B2BUA <------> UA Proxy2 and B2BUA are connected on LAN and Proxy1 is adding his own Contact Alias and Proxy 2 is adding his own Contact Alias I'm adding Alias via using the following function set_contact_alias("1") as it will remove other contact alias but is causing the problem in the case of BYE coming from B2BUA. Is there is a way to find if the contact alias is already set? How I can replace the local IP alias with domain name when request is coming from B2BUA and alias is added by Proxy2. Here are logs to show problem in 200 sent by B2BUA and proxy2 add alias 10.44.44.2~6000~1 Now in ACK it is causing the problem SIP/2.0 200 OK Via: SIP/2.0/UDP sip.prox1.example.com:5060 ;rport=5060;received=3.18.1.21;branch=z9hG4bK2d66.a89039bedfe3d1113132f04c94b8a757.0 Via: SIP/2.0/UDP 192.168.1.168:50279 ;received=109.8.2.1;rport=45607;branch=z9hG4bKPj84875d08eafc48bfa6c29de5fe245cea Record-Route: <sip:sip.proxy2.example;lr;did=e84.87e2;rtp=bridge> Record-Route: <sip:sip.prox1.example.com;lr;did=e84.7452;rtp=bridge> From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 INVITE Server: Voip Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:2501@10.44.44.2:6000;alias=10.44.44.2~6000~1> Content-Type: application/sdp Require: timer Content-Length: 281 Ack for 200 ACK sip:2501@10.44.44.2:6000 SIP/2.0 Via: SIP/2.0/UDP sip.proxy2.example.com:5060 ;branch=z9hG4bK2d66.3c10a3b45e5c8caacbc5853d5e6418b2.0 Via: SIP/2.0/UDP sip.proxy1.example.com:5060;rport=5060;received= 3.18.1.21 ;branch=z9hG4bK2d66.788312006d5f7aedc058435cd203fed3.0 Via: SIP/2.0/UDP 192.168.1.168:50279;received= 109.8.2.1 ;rport=45607;branch=z9hG4bKPjf003c3bfe4ba423e9a0dedd647511dd3 Max-Forwards: 68 From: "8001" <sip:8001@sip.example.com>;tag=0b9915abae654e21845fd4d7f577ede9 To: <sip:2501@sip.example.com>;tag=as1f4f6b1c Call-ID: a0095b9868924b0681225a8884a06c22 CSeq: 3687 ACK Content-Length: 0 Thanks in advance

1 0

SECSIPID Identity Decode
by Daniel W. Graham 06 Jul '21

06 Jul '21

I am unable to base64url decode the json payload in identity header generated by secsipid. (Using python for test) decoded_payload = url64.decode(‘payload’) UnicodeDecodeError: 'utf-8' codec can't decode byte 0xc2 in position 27: invalid continuation byte Header decodes fine this way but not payload. Is this an issue with the payload encoding? Kamailio 5.5 Daniel W. Graham, CTO CMSInter.net LLC DIRECT (989) 400-4230 INTERNET | TELEPHONE | MANAGED IT

5 28

I found bug in sipdump module.
by woenho 06 Jul '21

06 Jul '21

2 1

Re: [SR-Users] SECSIPID Identity Decode
by Daniel W. Graham 03 Jul '21

03 Jul '21

This was already pointed out, $mb. Thanks, -dan On Jul 2, 2021, at 4:40 PM, Daniel W. Graham <dan(a)cmsinter.net> wrote:

1 0

crash when sending reply
by Juha Heinanen 02 Jul '21

02 Jul '21

Kamailio 5.5 crashed when sending reply (bt full below). Before the crash there had been several mysql related error messages ("Lost connection to MySQL server during query" and "Too many connections") in syslog when async worker tried to insert accounting data. So perhaps this core dump has something to do with that. -- Juha Reading symbols from /usr/bin/sip-proxy...done. [New LWP 21324] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/bin/sip-proxy -f /etc/sip-proxy/sip-proxy.cfg -P /run/sip-proxy/sip-proxy.'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007fd274ce3b6e in run_trans_callbacks_internal (cb_lst=0x7fd2532503a8, type=512, trans=0x7fd253250330, params=0x7fff9b8eb2b0) at t_hooks.c:254 254 t_hooks.c: No such file or directory. (gdb) bt full #0 0x00007fd274ce3b6e in run_trans_callbacks_internal (cb_lst=0x7fd2532503a8, type=512, trans=0x7fd253250330, params=0x7fff9b8eb2b0) at t_hooks.c:254 cbp = 0x10 backup_from = 0x5613ccfb37f0 <def_list+16> backup_to = 0x5613ccfb37f8 <def_list+24> backup_dom_from = 0x5613ccfb3800 <def_list+32> backup_dom_to = 0x5613ccfb3808 <def_list+40> backup_uri_from = 0x5613ccfb37e0 <def_list> backup_uri_to = 0x5613ccfb37e8 <def_list+8> backup_xavps = 0x5613ccf56270 <_xavp_list_head> backup_xavus = 0x5613ccf56278 <_xavu_list_head> backup_xavis = 0x5613ccf56280 <_xavi_list_head> __func__ = "run_trans_callbacks_internal" #1 0x00007fd274ce41f7 in run_trans_callbacks_with_buf (type=512, rbuf=0x7fd253250400, req=0x7fd252d6dd20, repl=0x7fd2758bf0b0, flags=0) at t_hooks.c:303 params = {req = 0x7fd252d6dd20, rpl = 0x7fd2758bf0b0, param = 0x7fd254760f60, code = 486, flags = 0, branch = 0, t_rbuf = 0x7fd253250400, dst = 0x7fd253250450, send_buf = { s = 0x7fd25325b9f0 "@", len = 579}} trans = 0x7fd253250330 #2 0x00007fd274c88c15 in relay_reply (t=0x7fd253250330, p_msg=0x7fd2758bf0b0, branch=0, msg_status=486, cancel_data=0x7fff9b8eb580, do_put_on_wait=1) at t_reply.c:2133 relay = 0 save_clone = 0 buf = 0x7fd275ca0ff0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX.com;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"... res_len = 579 relayed_code = 486 relayed_msg = 0x7fd2758bf0b0 reply_bak = 0x7fd2510f1000 bm = {to_tag_val = {s = 0x7fd25216e4c0 "@\003", len = 1377234752}} totag_retr = 0 reply_status = RPS_COMPLETED uas_rb = 0x7fd253250400 to_tag = 0x7fd25216e820 reason = {s = 0x7fd25216e830 "0", len = 1394935248} onsend_params = {req = 0x9b8eb420, rpl = 0x7fd274c6b10d <futex_get+40>, param = 0x1, code = 1359984720, flags = 32722, branch = 0, t_rbuf = 0x7fff9b8eb460, dst = 0xa0679800, send_buf = { s = 0x19b8eb450 <error: Cannot access memory at address 0x19b8eb450>, len = 1359984720}} ip = {af = 2609820752, len = 32767, u = {addrl = {94643040895493, 0}, addr32 = {3436528133, 22035, 0, 0}, addr16 = {16901, 52437, 22035, 0, 0, 0, 0, 0}, addr = "\005B\325\314\023V\000\000\000\000\000\000\000\000\000"}} __func__ = "relay_reply" #3 0x00007fd274c8de6b in reply_received (p_msg=0x7fd2758bf0b0) at t_reply.c:2680 msg_status = 486 last_uac_status = 180 ack = 0x7fd25216e830 "0" ack_len = 459 branch = 0 reply_status = 32722 onreply_route = 2 cancel_data = {cancel_bitmap = 0, reason = {cause = 0, u = {text = {s = 0x0, len = 1977055608}, e2e_cancel = 0x0, packed_hdrs = {s = 0x0, len = 1977055608}}}} uac = 0x7fd2532505c0 t = 0x7fd253250330 lack_dst = {send_sock = 0x40, to = {s = {sa_family = 49168, sa_data = "\207u\322\177\000\000\220\266\216\233\377\177\000"}, sin = {sin_family = 49168, sin_port = 30087, sin_addr = { s_addr = 32722}, sin_zero = "\220\266\216\233\377\177\000"}, sin6 = {sin6_family = 49168, sin6_port = 30087, sin6_flowinfo = 32722, sin6_addr = {__in6_u = { __u6_addr8 = "\220\266\216\233\377\177\000\000L=\215s\322\177\000", __u6_addr16 = {46736, 39822, 32767, 0, 15692, 29581, 32722, 0}, __u6_addr32 = {2609821328, 32767, 1938636108, 32722}}}, sin6_scope_id = 1971831024}, sas = {ss_family = 49168, __ss_padding = "\207u\322\177\000\000\220\266\216\233\377\177\000\000L=\215s\322\177\000\000\360\300\207u\322\177", '\000' <repeats 13 times>, "\b\000\000\000\000\060Z\264u\322\177\000\000\213\t\342\316\023V\000\000;\b\342\316\023V\000\000\020\313\320\000\000\000\000\000\020\000\000\000\000\000\000\000I\001\000\000\000\000\000\000\000\230g\240\210\347A\357\300\266\216\233\377\177\000\000\000\230g\240\210\347", <incomplete sequence \357>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 0 '\000', proto_pad0 = 0 '\000', proto_pad1 = 0} backup_user_from = 0x5613ccfb37f0 <def_list+16> backup_user_to = 0x5613ccfb37f8 <def_list+24> backup_domain_from = 0x5613ccfb3800 <def_list+32> backup_domain_to = 0x5613ccfb3808 <def_list+40> backup_uri_from = 0x5613ccfb37e0 <def_list> backup_uri_to = 0x5613ccfb37e8 <def_list+8> backup_xavps = 0x5613ccf56270 <_xavp_list_head> backup_xavus = 0x5613ccf56278 <_xavu_list_head> backup_xavis = 0x5613ccf56280 <_xavi_list_head> replies_locked = 1 branch_ret = 0 prev_branch = 1971830800 blst_503_timeout = 32722 hf = 0xb5c750 onsend_params = {req = 0x7fff9b8eb5e0, rpl = 0xef41e788a0679800, param = 0x30, code = 0, flags = 0, branch = 0, t_rbuf = 0x0, dst = 0x7fd27d972cb0 <__syslog>, send_buf = { s = 0x5613ccec69b3 "INFO", len = -1603823616}} ctx = {rec_lev = 0, run_flags = 2, last_retcode = -1, jmp_env = {{__jmpbuf = {0, -5133089567146157739, 0, 140542026919088, 94643042412979, 536870912, -5133089567232140971, -1505551737941960363}, __mask_was_saved = 0, __saved_mask = {__val = {140735803210560, 13683472, 7765887371, 140735803209568, 94643041089583, 0, 17240315422543681536, 0, 0, 0, 0, 94643042412979, 17240315422543681536, 140735803209872, 94643041372964, 2866715017336}}}}} bctx = 0x7fd27587c010 keng = 0x0 ret = 0 evname = {s = 0x7fd274d3696f "on_sl_reply", len = 11} __func__ = "reply_received" #4 0x00005613ccbba0d4 in do_forward_reply (msg=0x7fd2758bf0b0, mode=0) at core/forward.c:764 new_buf = 0x0 dst = {send_sock = 0x0, to = {s = {sa_family = 0, sa_data = '\000' <repeats 13 times>}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 0}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 0, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 0, __ss_padding = '\000' <repeats 117 times>, __ss_align = 0}}, id = 0, send_flags = {f = 0, blst_imask = 0}, proto = 0 '\000', proto_pad0 = 0 '\000', proto_pad1 = 0} new_len = 0 r = 2 ip = {af = 1, len = 6356993, u = {addrl = {140735803210560, 140541891965104}, addr32 = {2609822528, 32767, 1972105392, 32722}, addr16 = {47936, 39822, 32767, 0, 61616, 30091, 32722, 0}, addr = "@\273\216\233\377\177\000\000\260\360\213u\322\177\000"}} s = 0xef41e788a0679800 <error: Cannot access memory at address 0xef41e788a0679800> len = -1685145520 __func__ = "do_forward_reply" #5 0x00005613ccbbc23c in forward_reply (msg=0x7fd2758bf0b0) at core/forward.c:865 No locals. #6 0x00005613ccc5eea9 in receive_msg ( buf=0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len=667, rcv_info=0x7fd253b7cad8) at core/receive.c:587 msg = 0x7fd2758bf0b0 ctx = {rec_lev = 0, run_flags = 0, last_retcode = 1, jmp_env = {{__jmpbuf = {0, -5133089566915471019, 0, 140542026919088, 94643042412979, 536870912, -5133089567001454251, -1505551737941960363}, __mask_was_saved = 0, __saved_mask = {__val = {94643073835016, 65535, 17240315422543681536, 18446744073709551536, 2, 0, 140542026919088, 94643042412979, 536870912, 140735803211024, 94643040635195, 140735803211696, 65535, 65536, 4095, 140735803211580}}}}} bctx = 0x0 ret = 1 tvb = {tv_sec = 0, tv_usec = 0} tve = {tv_sec = 0, tv_usec = 0} diff = 0 inb = {s = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len = 667} netinfo = {data = {s = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX;tag=as6964f29d\r\nCal"..., len = 667}, rcv = 0x7fd253b7cad8, dst = 0x0} keng = 0x0 evp = {data = 0x7fff9b8eba90, obuf = {s = 0x0, len = 0}, rcv = 0x7fd253b7cad8, dst = 0x0, req = 0x0, rpl = 0x0, rplcode = 0, mode = 0} cidlockidx = 0 cidlockset = 0 errsipmsg = 0 exectime = 0 __func__ = "receive_msg" #7 0x00005613ccd1cf15 in receive_tcp_msg (tcpbuf=0x7fd253b7ce60 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"..., len=667, rcv_info=0x7fd253b7cad8, con=0x7fd253b7cac0) at core/tcp_read.c:1424 buf = 0x5613cee206f0 "SIP/2.0 486 Busy Here\r\nTo: <sip:XXXXX@XXXXX;user=phone>;tag=h7g4Esbg_11002529899813\r\nFrom: <sip:XXXXX@XXXXX>;tag=as6964f29d\r\nCal"... bsize = 65535 blen = 65535 __func__ = "receive_tcp_msg" #8 0x00005613ccd1f554 in tcp_read_req (con=0x7fd253b7cac0, bytes_read=0x7fff9b8ebf34, read_flags=0x7fff9b8ebf3c) at core/tcp_read.c:1607 bytes = 667 total_bytes = 667 resp = 1 size = 0 req = 0x7fd253b7cbe8 dst = {send_sock = 0x7fff9b8ebec0, to = {s = {sa_family = 32062, sa_data = "\303\314\023V\000\000p\276\216\233\377\177\000"}, sin = {sin_family = 32062, sin_port = 52419, sin_addr = {s_addr = 22035}, sin_zero = "p\276\216\233\377\177\000"}, sin6 = {sin6_family = 32062, sin6_port = 52419, sin6_flowinfo = 22035, sin6_addr = {__in6_u = {__u6_addr8 = "p\276\216\233\377\177\000\000\030\000\000\000\000\000\000", __u6_addr16 = {48752, 39822, 32767, 0, 24, 0, 0, 0}, __u6_addr32 = {2609823344, 32767, 24, 0}}}, sin6_scope_id = 0}, sas = {ss_family = 32062, __ss_padding = "\303\314\023V\000\000p\276\216\233\377\177\000\000\030", '\000' <repeats 15 times>, "\225q\\\037P\000\000\000\070,\214u\322\177\000\000\000\244\374\314\023V\000\000\000\000\000\000\001 \000\000\070,\214u\322\177\000\000\001\000\000\000\000\000\000\000\300ʷS\322\177\000\000\200\276\216\233\377\177\000\000\070,\214u\322\177\000\000\200\277\216\233\377\177\000\000\b\000\000\000\000\000\000", __ss_align = 20}}, id = 1, send_flags = {f = 1, blst_imask = 0}, proto = 31 '\037', proto_pad0 = 0 '\000', proto_pad1 = 0} c = 49 '1' ret = 0 __func__ = "tcp_read_req" #9 0x00005613ccd23eb4 in handle_io (fm=0x7fd2758c2c38, events=1, idx=-1) at core/tcp_read.c:1857 ret = 0 n = 1404553920 read_flags = RD_CONN_SHORT_READ con = 0x7fd253b7cac0 s = 1404554328 resp = 1 t = 526152085 ee = 0x0 __func__ = "handle_io" #10 0x00005613ccd0dd1e in io_wait_loop_epoll (h=0x5613ccfca240 <io_w>, t=2, repeat=0) at core/io_wait.h:1070 n = 1 r = 0 fm = 0x7fd2758c2c38 revents = 1 __func__ = "io_wait_loop_epoll" #11 0x00005613ccd25981 in tcp_receive_loop (unix_sock=39) at core/tcp_read.c:1978 __func__ = "tcp_receive_loop" #12 0x00005613ccb8561b in tcp_init_children (woneinit=0x7fff9b8ec338) at core/tcp_main.c:5139 r = 2 i = 7 reader_fd_1 = 39 pid = 0 si_desc = "tcp receiver (generic)\000\000\205\220\274\314\023V\000\000\020Î\233\377\177\000\000\301\200\313\314\000\000\000\000AS\000\000\000\000\000\000۸\352\314\023V\000\000\246\000\216\233\377\177\000\000\034\352\340\314\023V\000\000\340\302\216\233\001\000\000\000\360\311\017Q\322\177\000\000\340\302\216\233\377\177\000\000\340\227:S\001\000\000\000\000Î\233\377\177\000\000\274\353\340\314\023V\000" si = 0x0 __func__ = "tcp_init_children" #13 0x00005613cca62bba in main_loop () at main.c:1857 i = 8 pid = 21313 si = 0x0 si_desc = "udp receiver child=7 sock=XXX.XXX.XXX.XXX:5060\000\000\000\003\000\000\000 \000\000\000\000\230g\240\210\347A\357\000\000\000\000\000\000\000\000\210P\347\314\023V\000\000\000\000\000\000\000\000\000\000\260,\227}\322\177\000\000\263i\354\314\023V\000\000\000\000\000 \000\000\000\000\260Ď\233\377\177\000\000\t\205\317\314\023V\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #14 0x00005613cca6d4aa in main (argc=17, argv=0x7fff9b8ec9f8) at main.c:3053 cfg_stream = 0x5613cecb6290 c = -1 r = 0 tmp = 0x7fff9b8ecf3f "" tmp_len = 2109797928 port = 32722 proto = 2109962736 ahost = 0x0 aport = 0 options = 0x5613cce77ce0 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3769828628 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0x0 st = {st_dev = 21, st_ino = 17087, st_nlink = 2, st_mode = 16832, st_uid = 109, st_gid = 115, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1620845793, tv_nsec = 887985330}, st_mtim = {tv_sec = 1625090397, tv_nsec = 157181302}, st_ctim = {tv_sec = 1625090401, tv_nsec = 197141517}, __glibc_reserved = {0, 0, 0}} tbuf = "\000\000\000\000\000\000\000\000\b\000\000\000\000\000\000\000\060\207\303}\322\177\000\000\000\000\000\000\000\000\000\000\240q\237\233\377\177\000\000\364\212\301}\322\177\000\000\b\000\000\000\000\000\000\000\230r\237\233\377\177\000\000\060\207\303}\322\177\000\000\230ǎ\233\377\177\000\000\224ǎ\233\377\177", '\000' <repeats 18 times>, "\230r\237\233\377\177\000\000\250q\237\233\377\177\000\000'\376\237}\322\177\000\000&\260be\000\000\000\000\000Ɏ\233\377\177\000\000\300\212\225\001\000\000\000\000`Ȏ\233\377\177\000\000PȎ\233\377\177\000\000\230ǎ\233\377\177\000\000X\207\303}\322\177\000\000\001", '\000' <repeats 31 times>... option_index = 0 long_options = {{name = 0x5613cce79f3f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x5613cce754e0 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x5613cce79f44 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x5613cce79f4a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x5613cce79f50 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x5613cce79f59 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x5613cce79f63 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x5613cce79f6d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x5613cce79f78 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x5613cce79f81 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x5613cce79f8c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x5613cce79f92 "cfg-print", has_arg = 0, flag = 0x0, val = 1033}, {name = 0x5613cce79f9c "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main"

3 9

UAC_AUTH with RTPEngine NAT
by Paul Smith 02 Jul '21

02 Jul '21

Hi, [Note I posted this yesterday from the wrong email address.] I am struggling to see why my SDP is not being set correctly on the INVITE to my supplier with Proxy-Authentication when I use uac_auth(). The initial INVITE to my supplier has correct SDP set by rtpengine_manage. Then supplier replies with a 407. My failure route correctly handles the Auth, and also calls NATMANAGE again... but this time the SDP is unchanged and the private IP and original media information from the original device is relayed to my supplier. kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and using example from uac module for uac_auth. My failure route calls NATMANAGE. Is there anything special about uac_auth? Do I need some extra magic to apply the message body changes after I have run rtpengine_manage(). I can see that the NATMANAGEr test for nat_uac_test("8") is true, and rtpengine_manage() s being called. But the outgoing SDP is not changed. Thanks for any hints! Paul ==================== Extract of kamailio.cfg failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } route(NATMANAGE); xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb"); if(t_check_status("401|407")) { # $avp(auser) = "test"; # $avp(apass) = "test"; # $avp(apass) = "36d0a02793542b4961e8348347236dbf"; if (uac_auth()) { t_relay(); } exit; } }

3 3

UAC_AUTH with RTPEngine NAT
by Paul Smith 01 Jul '21

01 Jul '21

Hi, I am struggling to see why my SDP is not being set correctly on the INVITE to my supplier with Proxy-Authentication when I use uac_auth(). The initial INVITE to my supplier has correct SDP set by rtpengine_manage. Then supplier replies with a 407. My failure route correctly handles the Auth, and also calls NATMANAGE again... but this time the SDP is unchanged and the private IP and original media information from the original device is relayed to my supplier. kamailio.cfg isbased on the default config. Running kamailio 5.4.6 and using example from uac module for uac_auth. My failure route calls NATMANAGE. Is there anything special about uac_auth? Do I need some extra magic to apply the message body changes after I have run rtpengine_manage(). ==================== Extract of kamailio.cfg failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } route(NATMANAGE); xlog("L_INFO","In failure route, just finshed NATMANAGE and now body is $mb"); if(t_check_status("401|407")) { # $avp(auser) = "test"; # $avp(apass) = "test"; # $avp(apass) = "36d0a02793542b4961e8348347236dbf"; if (uac_auth()) { t_relay(); } exit; } }

1 0

Kamailio BSD license
by stevecarellccie＠yahoo.com 01 Jul '21

01 Jul '21

Hello, I am trying to use Kamailio in our application which would be deployed in certain environment in addition to a cloud infrastructure. As I understand it, parts of the Kamailio code are covered under the BSD license - are you able to share what those are? I am trying to determine if Kamailio can be used in a customer environment or should it be strictly restricted to cloud environments due to the GPL v2 licenses Thanks,Steve

2 1

install stirshaken module by apt at Debian 10
by Yuriy Nasida 30 Jun '21

30 Jun '21

Hi, I know that it's possible to compile manually but what about apt method to install stirshaken module? I am checking the list of modules using: apt search kamailio But... Nothing looks like stirshaken. Thus, only several modules available and for others we should compile manually ? Please advise Thanks

2 1

Websocket In-dialoge SIP routing failed post network loss due to aliases
by Shahid Hussain 30 Jun '21

30 Jun '21

Hi, Websocket module documentation has a code reference to use aliases for SIP routing. However, aliases will not work in the following setup and situation. 1. Kamailio is configured with active and standby node 2. Ping is implemented from webclient and kamailio responds with pong. 3. Two clients ClinetA and ClientB registered themselves to Kamailio. 4. After SIP negotiation (INVITE-200OK) each client learnt about below aliases. Alias of ClientA: alias=172.27.6.98~58336 Alias of ClientB: alias=172.27.6.98~58342 So normally if ClientB wants to send SIP message to ClientA, SIP URI from ClientB looks like below ACK sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 4. Call is in a connected state. Following is the issue. i. Switchover (or network lost or reboot) at Kamailio happened ii. Due to ping pong both the clients detected network loss individually and re-registered themselves. iii. Aliases of both the clients got changed. New Alias of ClientA: alias=172.27.6.98~ 58346 New Alias of ClientB: alias=172.27.6.98~ 58348 iv. But ClientB doesn’t know that ClientA also re-registered and ClientA’s alias got changed and vice-versa v. Because of this ClientB still sends BYE with Initial alias BYE sip:v9d0gtl6@q0lrdlj63pik.invalid;alias=172.27.6.98~58336~5;transport=ws SIP/2.0 Would like to know what is the recommended solution for this problem using alias or is it a limitation of using alias? Thanks & Regards, Shahid

4 11

Re: [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 29 Jun '21

29 Jun '21

Thanks Muhammad, That seems to have done the trick. I am now able to bridge WebRTC to UDP. Regards, Peter Manley M S shaheryarkh at gmail.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Guidance%20building%20a%20WebRTC%20to%20Asterisk%20Intermediate%0A%20Proxy&In-Reply-To=%3CCAFZQphyZE_tGB9%3D5tFQBtXAv5twMh%3D9aCNFAvveGf0-L%3Dy_qwQ%40mail.gmail.com%3E> Sat Jun 26 03:57:27 CEST 2021 * Previous message (by thread): [SR-Users] Guidance building a WebRTC to Asterisk Intermediate Proxy<https://lists.kamailio.org/pipermail/sr-users/2021-June/112799.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112800> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112800> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112800> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112800> ________________________________ The problem seems to that the kamailio is trying to route it over UDP port using WSS transport, which will never work. You are creating a transport bridge (UDP <-> WSS i.e. stateless transport <-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going to be tricky. I strongly discourage it since things get messy for sequential requests. But you can try the following, Here are some PATH module parameters and functions, you will need to setup. 1. You will need to enable "use_received" parameter to ensure connection tracking. 2. It seems you may have enabled "received_format", so you make sure you do NAT detection and use "handle_ruri_alias" for initial requests. 3. You must also enabled "enable_r2", but with caution, since both ends (i.e. UAC and UAS) can be using same kamailio (proxy) socket for in/out. 4. Make sure to use "add_path_received" instead of "add_path". 5. If using OUTBOUND module then make sure it is loaded BEFORE the PATH module. Thank you. -- Muhammad Shahzad Shafi Burraq Technologies Tel: +49 176 99 83 10 85 On Sat, Jun 26, 2021 at 12:47 AM Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm working on building an RTP Proxy which will allow both regular SIP to > SIP transactions and WebRTC to SIP transactions to an Asterisk Server. > > > > SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > AND > > > > WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > > Asterisk Server > > > > The Kamailio/RTPEngine Proxy is stateless (does not use the tm or > registerer modules), and uses the Path module to replace the Route header > with a Path header. > > > > It works grep for SIP/UDP to SIP/UDP transactions. > > > > HOWEVER... It only partially works for WSS to SIP/UDP transactions. The > REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. > > > > When the Asterisk server sends a SIP Options message, I get the following > error: > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < > sip:l4umfumh at 192.0.2.97<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>;transport=ws;alias=209.141.117.212~59751~6 -> > <null> (IP:172.16.0.48:sp) > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for > WebSocket could not be found > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not > create Via header > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> > [core/forward.c:550]: forward_request(): building failed > > Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> > [core/receive.c:425]: receive_msg(): error while trying script > > > > It appears that the WebSocket corresponding to the WSS leg cannot be > located when the request is initiated from the Asterisk Server. > > > > Any thoughts on how I can get this to work in the other direction > (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and > Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite > do what I want it to do (i.e. it appears to act as a registrar server and > doesn't forward REGISTER requests on to another registrar). > > > > > > Thanks and regards, > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

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Kamailio + RTPEngine - Destinations Behind NAT
by Edward Romanenco 29 Jun '21

29 Jun '21

Hi everyone, I'm using Kamailio (Ver 5.5.1, Public IP X.X.X.11, Private IP 172.18.0.20) + RTPEngine (Ver 9.4.1.1, Public IP X.X.X.7, Private IP 172.18.0.50) alongside Freeswitch (Private IP 172.18.0.40) in my backend and delivering calls to a mobile application (call flow is Freeswitch -> Kamailio + RTPEngine -> Applicaiton). While calls made to NATted destination generally works fine - from time to time, we've experiencing one-sided audio due to the rtpengine sending the media stream to the remote end's private IP instead of his public one. For instance, in the attached logs - media was sent towards 192.168.1.93 instead of 79.179.1.2, which is the expected destination. Problem appears almost randomly; I can make 20 fine calls in a row for the same destination until experiencing it, and I didn't find any difference as it comes to the SIP invite exchange between the two. Has anyone ever experienced the same behavior and can share some general instructions on how to troubleshoot it? Relevant Kamailio config section is attached. Invite Kamailio->Device INVITE sip:972587102881@79.179.1.2:56308;transport=TLS;ob SIP/2.0 Record-Route: <sip:X.X.X.11:443;transport=tls;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;r2=on;lr=on;ftag=cvcHy5UcS9pjj> Via: SIP/2.0/TLS X.X.X.11:443;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;received=172.18.0.40;rport=5060;branch=z9hG4bKH234Q58HpHNKK Max-Forwards: 66 From: "972522149596" <sip:FreeSWITCH@172.18.0.20:5099>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20:5099> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 CSeq: 37932220 INVITE Contact: <sip:gw+kam-in@172.18.0.40:5060;transport=udp;gw=kam-in;alias=172.18.0.40~5060~1> User-Agent: FreeSWITCH-mod_sofia/1.10.6-release+git~20210325T131609Z~1ff9d0a60e~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 567 X-SESSION-ID: 721533765 X-INCOMING: PSTN X-from-freeswitch: +972522149596 X-FS-Support: update_display,send_info Remote-Party-ID: "972522149596" <sip:972522149596@172.18.0.20:5099>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 s=FreeSWITCH c=IN IP4 X.X.X.7 t=0 0 m=audio 30018 RTP/AVP 102 101 a=rtpmap:102 opus/48000/2 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=rtpmap:101 telephone-event/48000 a=fmtp:101 0-16 a=sendrecv a=rtcp:30019 a=ptime:20 a=ice-ufrag:bKPZt0dW a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host Response Device->Kamailio SIP/2.0 200 OK Via: SIP/2.0/TLS X.X.X.11:443;received=X.X.X.11;branch=z9hG4bK24bf.52b1dd68e26010ba103bd8d6fdb53cf4.0 Via: SIP/2.0/UDP 172.18.0.40;rport=5060;received=172.18.0.40;branch=z9hG4bKH234Q58HpHNKK Record-Route: <sip:X.X.X.11:443;transport=tls;lr;r2=on;ftag=cvcHy5UcS9pjj> Record-Route: <sip:172.18.0.20:5099;lr;r2=on;ftag=cvcHy5UcS9pjj> Call-ID: be55bd1e-537d-123a-65a9-0242ac120028 From: "972522149596" <sip:FreeSWITCH@172.18.0.20>;tag=cvcHy5UcS9pjj To: <sip:972587102881@172.18.0.20>;tag=5f75877e-58c2-44f3-a5b3-43f8786c42a1 CSeq: 37932220 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: <sip:972587102881@79.179.1.2:56308;transport=TLS;ob> Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 348 v=0 o=- 3833960833 3833960834 IN IP4 192.168.1.93 s=pjmedia b=AS:117 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 96 120 c=IN IP4 192.168.1.93 b=TIAS:96000 a=rtcp:4003 IN IP4 192.168.1.93 a=sendrecv a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:120 telephone-event/48000 a=fmtp:120 0-16 a=ssrc:816694098 cname:72cdb5f53fc2e8fb Kamailio NAT Detect/Manage Routes # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT xinfo("NATDETECT: Adding the rport parameter to the first Via header of the received message\n"); force_rport(); xinfo("NATDETECT: Trying to guess if client's request originated behind a nat. Testing 19.\n"); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xinfo("NATDETECT: REGISTER request. Creating a URI consisting of the source IP, port, and protocol and storing the URI in an Attribute-Value-Pair.\n"); fix_nated_register(); } else { if(is_first_hop()) { xinfo("NATDETECT: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } xinfo("NATDETECT: Setting flag FLT_NATS.\n"); setflag(FLT_NATS); } else { xinfo("NATDETECT: Didn't detect natted request by any of the given heuristics\n"); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xinfo("NATMANAGE: This is a request with to-tag that has 'nat=yes', setting branch flag FLB_NATB.\n"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) { xinfo("NATMANAGE: No NAT flag detected for this request\n"); return; } else { xinfo("NATMANAGE: NAT flag detected for this request\n"); } if(!is_present_hf("x-purpose")) { xinfo("NATMANAGE: x-purpose header does not appear in this request, SDP is searched for occurrence of RFC1918 or RFC6598 addresses.\n"); if(nat_uac_test("8")) { xinfo("NATMANAGE: Found occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin direction=pub direction=priv"); } } else { xinfo("NATMANAGE: No occurrence of RFC1918 or RFC6598 addresses. Checking if there is a match of source address or uri with an address in the group of the dispatcher-list.\n"); if(ds_is_from_list()) { xinfo("NATMANAGE: There is a match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=priv direction=pub"); } else { xinfo("NATMANAGE: No match with an address in the group of the dispatcher-list.\n"); rtpengine_manage("ICE=force replace-session-connection replace-origin trust-address direction=pub direction=priv"); } } } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xinfo("NATMANAGE: This is a request without to-tag and top-executed route block is branch_route, adding record route param ;nat=yes.\n"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xinfo("NATMANAGE: Adding an ;alias=ip~port~transport parameter to the contact URI containing the received ip, port, and transport protocol.\n"); set_contact_alias(); } } } #!endif return; } RTPEngine Logs 2021-06-29T13:07:05.440658779Z [1624972025.439547] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'offer' from 172.18.0.20:53880 2021-06-29T13:07:05.440690853Z [1624972025.439641] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'offer' from 172.18.0.20:53880: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:05.440695330Z o=FreeSWITCH 1624937621 1624937622 IN IP4 172.18.0.40 2021-06-29T13:07:05.440698353Z s=FreeSWITCH 2021-06-29T13:07:05.440700926Z c=IN IP4 172.18.0.40 2021-06-29T13:07:05.440703539Z t=0 0 2021-06-29T13:07:05.440706179Z m=audio 34404 RTP/AVP 102 101 2021-06-29T13:07:05.440708722Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.440711294Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.440714053Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.440716746Z a=fmtp:101 0-16 2021-06-29T13:07:05.440719258Z a=ptime:20 2021-06-29T13:07:05.440721823Z ", "ICE": "force", "direction": [ "priv", "pub" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "172.18.0.40" ], "from-tag": "cvcHy5UcS9pjj", "command": "offer" } 2021-06-29T13:07:05.441069790Z [1624972025.439694] NOTICE: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] Creating new call 2021-06-29T13:07:05.441088698Z [1624972025.439809] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:05.441091829Z [1624972025.439815] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 101 2021-06-29T13:07:05.441094966Z [1624972025.439821] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:05.441097777Z [1624972025.439828] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:05.441100540Z [1624972025.439833] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:05.441103390Z [1624972025.439839] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:05.441106153Z [1624972025.439844] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:05.441108913Z [1624972025.439849] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:05.441111679Z [1624972025.440116] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34404 2021-06-29T13:07:05.441124990Z [1624972025.440124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 172.18.0.40:34405 2021-06-29T13:07:05.441663481Z [1624972025.441074] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'offer' from 172.18.0.20:53880 (elapsed time 0.001414 sec) 2021-06-29T13:07:05.441694377Z [1624972025.441103] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'offer' to 172.18.0.20:53880: { "sdp": "v=0 2021-06-29T13:07:05.441701068Z o=FreeSWITCH 1624937621 1624937622 IN IP4 X.X.X.7 2021-06-29T13:07:05.441706421Z s=FreeSWITCH 2021-06-29T13:07:05.441711414Z c=IN IP4 X.X.X.7 2021-06-29T13:07:05.441716274Z t=0 0 2021-06-29T13:07:05.441721177Z m=audio 30018 RTP/AVP 102 101 2021-06-29T13:07:05.441726204Z a=rtpmap:102 opus/48000/2 2021-06-29T13:07:05.441731246Z a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 2021-06-29T13:07:05.441736873Z a=rtpmap:101 telephone-event/48000 2021-06-29T13:07:05.441742017Z a=fmtp:101 0-16 2021-06-29T13:07:05.441827012Z a=sendrecv 2021-06-29T13:07:05.441844744Z a=rtcp:30019 2021-06-29T13:07:05.441848881Z a=ptime:20 2021-06-29T13:07:05.441852444Z a=ice-ufrag:bKPZt0dW 2021-06-29T13:07:05.441856394Z a=ice-pwd:H4qyL7mPUOvoiAjosiLasXAdzj 2021-06-29T13:07:05.441860642Z a=candidate:41sXGEsTafxHi8ms 1 UDP 2130706431 X.X.X.7 30018 typ host 2021-06-29T13:07:05.441864955Z a=candidate:41sXGEsTafxHi8ms 2 UDP 2130706430 X.X.X.7 30019 typ host 2021-06-29T13:07:05.441868967Z ", "result": "ok" } 2021-06-29T13:07:07.001082810Z [1624972027.000176] DEBUG: [core] timer run time = 0.000029 sec 2021-06-29T13:07:07.817899628Z [1624972027.816949] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> in_ssrc changed for: :0 new: 30adc352 2021-06-29T13:07:07.817932021Z [1624972027.816987] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] >>> out_ssrc changed for 172.18.0.40:34405 new: 30adc352 2021-06-29T13:07:07.817936388Z [1624972027.816998] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.817939853Z [1624972027.817025] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.817942689Z [1624972027.817030] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.817945696Z [1624972027.817036] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.817948500Z [1624972027.817124] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.818762433Z [1624972027.817995] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 201 2021-06-29T13:07:07.818790347Z [1624972027.818031] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] RR from 30adc352 about 0: FL 0 TL 0 HSR 0 J 0 LSR 0 DLSR 0 2021-06-29T13:07:07.818794422Z [1624972027.818035] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] No payload type known for RTCP RR, discarding 2021-06-29T13:07:07.818797834Z [1624972027.818042] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [rtcp] Calling handler for RTCP packet type 202 2021-06-29T13:07:07.818801095Z [1624972027.818122] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30019]: [core] Forward to sink endpoint: 172.18.0.40:34405 2021-06-29T13:07:07.847992398Z [1624972027.846284] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Received command 'answer' from 172.18.0.20:47817 2021-06-29T13:07:07.848048472Z [1624972027.846325] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Dump for 'answer' from 172.18.0.20:47817: { "supports": [ "load limit" ], "sdp": "v=0 2021-06-29T13:07:07.848058346Z o=- 3833960833 3833960834 IN IP4 192.168.1.93 2021-06-29T13:07:07.848063378Z s=pjmedia 2021-06-29T13:07:07.848067785Z b=AS:117 2021-06-29T13:07:07.848071736Z t=0 0 2021-06-29T13:07:07.848075782Z a=X-nat:0 2021-06-29T13:07:07.848079588Z m=audio 4002 RTP/AVP 96 120 2021-06-29T13:07:07.848084210Z c=IN IP4 192.168.1.93 2021-06-29T13:07:07.848089110Z b=TIAS:96000 2021-06-29T13:07:07.848092756Z a=rtcp:4003 IN IP4 192.168.1.93 2021-06-29T13:07:07.848096578Z a=sendrecv 2021-06-29T13:07:07.848100173Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848104153Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848108660Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848112776Z a=fmtp:120 0-16 2021-06-29T13:07:07.848116856Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848120776Z ", "ICE": "force", "direction": [ "pub", "priv" ], "replace": [ "session-connection", "origin" ], "call-id": "be55bd1e-537d-123a-65a9-0242ac120028", "received-from": [ "IP4", "79.179.1.2" ], "from-tag": "cvcHy5UcS9pjj", "to-tag": "5f75877e-58c2-44f3-a5b3-43f8786c42a1", "command": "answer" } 2021-06-29T13:07:07.848128079Z [1624972027.846411] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Default sink codec is opus/48000/2 2021-06-29T13:07:07.848132255Z [1624972027.846418] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Output DTMF payload type is 120 2021-06-29T13:07:07.848136164Z [1624972027.846424] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for opus/48000/2 2021-06-29T13:07:07.848139819Z [1624972027.846429] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec opus/48000/2 2021-06-29T13:07:07.848143930Z [1624972027.846434] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for opus/48000/2 2021-06-29T13:07:07.848169208Z [1624972027.846441] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Creating codec handler for telephone-event/48000 2021-06-29T13:07:07.848174300Z [1624972027.846445] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Sink supports codec telephone-event/48000 2021-06-29T13:07:07.848178752Z [1624972027.846449] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [codec] Using passthrough handler for telephone-event/48000 2021-06-29T13:07:07.848183460Z [1624972027.846586] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4002 2021-06-29T13:07:07.848188805Z [1624972027.846593] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [core] set FILLED flag for stream 192.168.1.93:4003 2021-06-29T13:07:07.848193111Z [1624972027.846597] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [ice] Shutting down ICE agent (nothing to do) 2021-06-29T13:07:07.848197042Z [1624972027.847228] INFO: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Replying to 'answer' from 172.18.0.20:47817 (elapsed time 0.000890 sec) 2021-06-29T13:07:07.848201623Z [1624972027.847243] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028]: [control] Response dump for 'answer' to 172.18.0.20:47817: { "sdp": "v=0 2021-06-29T13:07:07.848205754Z o=- 3833960833 3833960834 IN IP4 172.18.0.50 2021-06-29T13:07:07.848209696Z s=pjmedia 2021-06-29T13:07:07.848213446Z b=AS:117 2021-06-29T13:07:07.848217361Z t=0 0 2021-06-29T13:07:07.848220816Z a=X-nat:0 2021-06-29T13:07:07.848224363Z m=audio 30018 RTP/AVP 96 120 2021-06-29T13:07:07.848228432Z c=IN IP4 172.18.0.50 2021-06-29T13:07:07.848232603Z b=TIAS:96000 2021-06-29T13:07:07.848235992Z a=ssrc:816694098 cname:72cdb5f53fc2e8fb 2021-06-29T13:07:07.848240381Z a=rtpmap:96 opus/48000/2 2021-06-29T13:07:07.848243923Z a=fmtp:96 useinbandfec=1 2021-06-29T13:07:07.848247725Z a=rtpmap:120 telephone-event/48000 2021-06-29T13:07:07.848386453Z a=fmtp:120 0-16 2021-06-29T13:07:07.848395759Z a=sendrecv 2021-06-29T13:07:07.848399822Z a=rtcp:30019 2021-06-29T13:07:07.848403748Z a=ptime:20 2021-06-29T13:07:07.848407203Z a=ice-ufrag:4Q9H38CE 2021-06-29T13:07:07.848410827Z a=ice-pwd:OR8EfUn5dCLbhZqyiwsTUOyUdW 2021-06-29T13:07:07.848414680Z a=candidate:f75EKiqnFiVfQdoZ 1 UDP 2130706431 172.18.0.50 30018 typ host 2021-06-29T13:07:07.848419247Z a=candidate:f75EKiqnFiVfQdoZ 2 UDP 2130706430 172.18.0.50 30019 typ host 2021-06-29T13:07:07.848432370Z ", "result": "ok" } 2021-06-29T13:07:07.872251466Z [1624972027.871342] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Created candidate pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 between 172.18.0.50 and 172.18.0.40:34404, type prflx 2021-06-29T13:07:07.872276197Z [1624972027.871381] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] Triggering check for f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 2021-06-29T13:07:07.872286600Z [1624972027.871396] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE pair f75EKiqnFiVfQdoZ:33d9f4832b3cc9a0:1 has been nominated by peer 2021-06-29T13:07:07.872289762Z [1624972027.871401] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [ice] ICE not completed yet and no usable candidates 2021-06-29T13:07:07.872292668Z [1624972027.871407] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Successful STUN binding request from 172.18.0.40:34404 2021-06-29T13:07:07.919668940Z [1624972027.918862] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> in_ssrc changed for: 172.18.0.40:34404 new: 70f328c1 2021-06-29T13:07:07.919698433Z [1624972027.918897] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] >>> out_ssrc changed for 192.168.1.93:4002 new: 70f328c1 2021-06-29T13:07:07.919703321Z [1624972027.918911] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64423 TS 2880) 2021-06-29T13:07:07.939554198Z [1624972027.938733] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64424 TS 3840) 2021-06-29T13:07:07.959769679Z [1624972027.959017] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64425 TS 4800) 2021-06-29T13:07:07.979861917Z [1624972027.979067] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64426 TS 5760) 2021-06-29T13:07:07.999566603Z [1624972027.998777] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64427 TS 6720) 2021-06-29T13:07:08.019668998Z [1624972028.018802] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64428 TS 7680) 2021-06-29T13:07:08.040856248Z [1624972028.039663] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64429 TS 8640) 2021-06-29T13:07:08.061133612Z [1624972028.059847] DEBUG: [be55bd1e-537d-123a-65a9-0242ac120028 port 30018]: [core] Forward to sink endpoint: 192.168.1.93:4002 (RTP seq 64430 TS 9600) Edward

1 1

Guidance building a WebRTC to Asterisk Intermediate Proxy
by Peter Manley 26 Jun '21

26 Jun '21

Hello, I'm working on building an RTP Proxy which will allow both regular SIP to SIP transactions and WebRTC to SIP transactions to an Asterisk Server. SIP UAC < ---UDP/SIP --- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server AND WebRTC UAC < ---WSS--- > Kamailio/RTPEngine Proxy < ---UDP/SIP----- > Asterisk Server The Kamailio/RTPEngine Proxy is stateless (does not use the tm or registerer modules), and uses the Path module to replace the Route header with a Path header. It works grep for SIP/UDP to SIP/UDP transactions. HOWEVER... It only partially works for WSS to SIP/UDP transactions. The REGISTER/401(Unauthorized), REGISTER/200(OK) transaction works well. When the Asterisk server sends a SIP Options message, I get the following error: Jun 25 15:47:03 VRTPENGINE kamailio[4953]: INFO: <script>: => OPTIONS < sip:l4umfumh@192.0.2.97;transport=ws;alias=209.141.117.212~59751~6 -> <null> (IP:172.16.0.48:sp) Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/msg_translator.c:2815]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/msg_translator.c:2003]: build_req_buf_from_sip_req(): could not create Via header Jun 25 15:47:03 VRTPENGINE kamailio[4953]: ERROR: <core> [core/forward.c:550]: forward_request(): building failed Jun 25 15:47:03 VRTPENGINE kamailio[4953]: WARNING: <core> [core/receive.c:425]: receive_msg(): error while trying script It appears that the WebSocket corresponding to the WSS leg cannot be located when the request is initiated from the Asterisk Server. Any thoughts on how I can get this to work in the other direction (Asterisk to Kamailio)? I've been looking at the WebRTC to SIP Client and Server (https://github.com/havfo/WEBRTC-to-SIP), but that doesn't quite do what I want it to do (i.e. it appears to act as a registrar server and doesn't forward REGISTER requests on to another registrar). Thanks and regards, Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

2 1

using both ldap and local authenticate
by Thành Trần Duy 25 Jun '21

25 Jun '21

Hi all, I am using kamailio as PBX with some softphones and hard phones. I want user on a pc to login with LDAP and users using a hard phone login normally (username/password). Can I use 2 types of authentication at the same time? Many thanks, Tran Duy Thanh

2 2

Possible memory leak on 5.5.x (new)?
by George Diamantopoulos 25 Jun '21

25 Jun '21

Hello all, I'm still investigating the (most likely non-kamailio-related) memory leak of my previous message to the list, there have been no developments so far. I'll update if anything changes. This concerns a new finding which seems to affect kamailio 5.5.x. I have two kamailio instances receiving the same traffic via round-robin. I upgraded only one of them to 5.5.1 and left the other to 5.4.6 as I feared of any issues arising. I was lucky to do so, because with identical configuration, 5.5.x seems to run out of SHM very quickly. Here are links to graphs produced by our monitoring system: Old kamailio (no memory leak): https://pasteboard.co/K8fVBiD.png New kamailio (possible leak): https://pasteboard.co/K8fVS9N.png The configuration uses mtree, htable, vars and vns extensively. Has anyone come across anything similar? Let me know if I can provide any further information to help disect this. Thanks! BR, George

2 1

rr module: record_route_advertised_address()/record_route_preset() do ignore custom user AVP
by Valentin Christoph 25 Jun '21

25 Jun '21

Hi, Maybe a discussion for the DEV mailing list???? Want to overwrite the FQDN by a NATed IP address in the topmost Record-Route of an outgoing INVITE request. Cannot set the advertise parameter (this must tell the FQDN for all other situations), only in some special situations I want to overwrite the FQDN in Record-Route. My solution would be to implement in the Kamailio source code (rr module) that not only record_route() uses the get_custom_user() function, but the other both functions uses it, too. Any better proposals? All the best Christoph

2 3

Kamailio $Rp variable showing wrong port
by Shahid Hussain 25 Jun '21

25 Jun '21

Hi, I have configured the dispatcher default socket as below IP , the expectation here is that kamailio should send all traffic from this IP (port will be ephemeral) over TCP. modparam("dispatcher", "ds_default_socket", "tcp:172.27.45.219:5060") This system also listens on port 8080 which is meant for websocket communication (dispatcher communication is over non websocket). Kamailio is sending an OPTION message to dispatcher IP over TCP on 5060 and getting 200 OK response but Kamailio considering the response is on websocket port 8080 ,$Rp value is 8080. onreply_route { if (($Rp == 8080 ) && !(proto == WS)) { xlog("L_WARN", "SIP response received on $Rp protocol:proto RIP:$Ri SIP:$si method:$rm cid:$ci replystatus:$rs len:$ml\n"); xlog("L_WARN", "SIP message received: $mb\n"); drop; } ..... .... } Following is the output of above logs: Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP response received on 8080 protocol:tcp RIP:172.27.45.219 SIP:172.27.44.125 method:OPTIONS cid:1ab6573a4640823b-6209@172.27.45.198 replystatus:200 len:416 Jun 24 15:10:08 localhost /usr/sbin/kamailio[6223]: WARNING: {2 10 OPTIONS 1ab6573a4640823b-6209(a)172.27.45.198} <script>: SIP message received: SIP/2.0 200 OK#015#012Via: SIP/2.0/TCP 172.27.45.198:5060;branch=z9hG4bK4526.92b79f34000000000000000000000000.0;received=172.27.45.219#015#012From: <sip:dispatcher@localhost>;tag=1ae6c95d92f9ea9ce927f715de5c48c8-eb141029#015#012To: <sip:172.27.44.125:5060;transport=tcp>;tag=a9d542e0#015#012Call-ID: 1ab6573a4640823b-6209(a)172.27.45.198#015#012CSeq: 10 OPTIONS#015#012Allow: INVITE, ACK, BYE, OPTIONS, CANCEL, INFO, UPDATE, NOTIFY#015#012Content-Length: 0 Also I have attached a pcap. Dispatcher.list configuration: 1 sip:172.27.44.125:5060;transport=tcp 0 0 duid=sample-cas;maxload=1000 Thanks, Shahid

3 10

S-CSCF and I-CSCF
by Jayanth Acharya 24 Jun '21

24 Jun '21

Hi, Have found example configuration for S-CSCF, I-CSCF and P-CSCF in Kamailio repository, and a S-CSCF module for the registered user data, but no recent updates. Wondering if anyone can share insights into how widely it has been used, or interops done in say telco networks with it ? Does it handle third-party registrations, is it actually able to do a Cx interaction with HSS ? Finally, how have these components that make up the IMS-core from SIP signaling standpoint, been tested ? Are there any signaling only test harnesses ? cheers, Jay

2 1

Errors after upgrade to 5.5
by Daniel W. Graham 23 Jun '21

23 Jun '21

Receiving constant errors for all functions after upgrade to 5.5, any change that could have caused this? ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1251 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1309 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1321 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1342 us] ERROR: <core> [db_query.c:68]: db_do_submit_query(): alert - query execution too long [1493 us] for [select `password`,`id` from `subscriber` where `username`='x'] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1667 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:968] took too long [1694 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:988] took too long [1817 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (16)] cfg [/usr/local/etc/kamailio/kamailio.cfg:1038] took too long [1835 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [corefunc (5)] cfg [/usr/local/etc/kamailio/kamailio.cfg:452] took too long [1869 us] ERROR: <core> [core/action.c:1598]: run_actions(): alert - action [www_authenticate (26)] cfg [/usr/local/etc/kamailio/kamailio.cfg:951] took too long [1405 us] ……. -dan

2 4

AVPs in Switch/Case
by Joel Serrano 22 Jun '21

22 Jun '21

Hi guys, Quick question, are AVPs valid in "switch" as "case"'s? Example: $avp(var1) = "101"; $avp(var2) = "201"; switch($rU) { case "$avp(var1)": ... break; case "$avp(var2)": ... break; default: send_reply("403", "Forbidden"); exit; } For some reason I can't get the case to match, and I'm thinking it's not using the actual value of $avp(var1) and instead it's using literally "$avp(var1)" as a string. NOTE: I also tried without putting the AVPs in quotes, but Kamailio complains and doesn't start. Is this doable or do I need to use if..else if..else if..else for this? Thanks, Joel.

3 3

Web Sockets connection error: bad request, state=7, error=4
by Peter Manley 22 Jun '21

22 Jun '21

Hello, I'm using Kamailio 5.3.2 and am trying to establish a WebRTC connection for using SIP through Kamailio on port 8089. I am getting the following error in the Syslog: Jun 22 09:20:38 VRTPENGINE kamailio[43050]: ERROR: <core> [core/tcp_read.c:1535]: tcp_read_req(): bad request, state=7, error=4 buf:#012GET /ws HTTP/1.1#015#012Host: rtpengine.kombea.io:8089#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/91.0.4472.106 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://vicidev.kombea.io#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9#015#012Sec-WebSocket-Key: 1Pda50NxHxMenZaMW7cUkA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012#012parsed:#012GET /ws HTTP/1.1#015#012Host: rtpengine.kombea.io:8089#015#012Connection: Upgrade#015#012Pragma: no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/91.0.4472.106 Safari/537.36#015#012Upgrade: websocket#015#012Origin: https://vicidev.kombea.io#015#012Sec-WebSocket-Version: 13#015#012Accept-Encoding: gzip, deflate, br#015#012Accept-Language: en-US,en;q=0.9#015#012Sec-WebSocket-Key: 1Pda50NxHxMenZaMW7cUkA==#015#012Sec-WebSocket-Extensions: permessage-deflate; client_max_window_bits#015#012Sec-WebSocket-Protocol: sip#015#012#015#012 Any idea on what would cause this? Thanks Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

2 2

execute callback every X minutes mid call
by Avremy Cohen 22 Jun '21

22 Jun '21

Hi I am trying to execute code every X min in active calls and I cant seem to find any way to do this. I basically have a function that checks if the user has balance and permission to carry on the call and the callback should trigger it every X mins. The call_control module doesnt work for me since A. It doesn't trigger a function it just ends the callback B. it uses an external application to calculate it any advice is appreciated Avi

3 3

Re: [SR-Users] Unable to establish TCP connection with Kamailio
by Peter Manley 17 Jun '21

17 Jun '21

Hello Ovidiu, Interestingly, that fixed the problem. I can now continue to make progress. Why does not forking and/or logging to stderr cause this to happen? Thank you! Regards, Peter Manley Ovidiu Sas osas at voipembedded.com<mailto:sr-users%40lists.kamailio.org?Subject=Re%3A%20%5BSR-Users%5D%20Unable%20to%20establish%20TCP%20connection%20with%20Kamailio&In-Reply-To=%3CCAND0LkvxyxE%3D5CPxZTe-Em43X6E5nep6z9n7a8f1zo3YbKksMw%40mail.gmail.com%3E> Thu Jun 17 23:05:14 CEST 2021 * Previous message (by thread): [SR-Users] Unable to establish TCP connection with Kamailio<https://lists.kamailio.org/pipermail/sr-users/2021-June/112759.html> * Messages sorted by: [ date ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/date.html#112760> [ thread ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/thread.html#112760> [ subject ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/subject.html#112760> [ author ]<https://lists.kamailio.org/pipermail/sr-users/2021-June/author.html#112760> ________________________________ It seems that you have fork set to no, Let kamailio fork and collect logs via syslog (not via stderror). -ovidiu On Thu, Jun 17, 2021 at 16:26 Peter Manley <peter.manley at kombea.com<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> wrote: > Hello, > > > > I'm running Kamailio Version 5.3.2 on Ubuntu 20.04. Here my build > information: > > > > version: kamailio 5.3.2 (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, 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_BLACKLIST, > HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED > > ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, > BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > id: unknown > > compiled with gcc 9.2.1 > > > > I have the following configuration set in my kamalio.cfg file: > > > > ##!define WITH_MYSQL > > ##!define WITH_AUTH > > ##!define WITH_USRLOCDB > > ##!define WITH_TLS > > ##!define WITH_HOMER > > ##!define WITH_WEBSOCKETS > > ##!define WITH_ANTIFLOOD > > ##!define WITH_IPV6 > > ##!define WITH_BRIDGE_ON_FAIL > > ##!define WITH_LOCALHOST_WS > > ##!define WITH_LOCALHOST_SIP > > > > #!substdef "!MY_SIP_PORT!5060!g" > > #!substdef "!MY_SIPS_PORT!5061!g" > > #!substdef "!MY_WS_PORT!8088!g" > > #!substdef "!MY_WSS_PORT!8089!g" > > > > #!substdef "!MY_LOG_FACILITY!LOG_DAEMON!g" > > > > #!substdef "!MY_IP4_ADDR!172.16.0.57!g" > > #!substdef "!IP4_LOCALHOST!127.0.0.1!g" > > #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" > > #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" > > #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" > > #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" > > > > #!ifdef WITH_IPV6 > > #!substdef "!MY_IP6_ADDR![fe80::d6e4:2f5f:e5b7:7dff]!g" > > #!substdef "!IP6_LOCALHOST![::1]!g" > > #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" > > #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" > > #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" > > #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" > > #!endif > > > > #!substdef "!MY_DOMAIN!rtpengine.kombea.io!g" > > debug=10 // debug level (cmd line: -dddddddddd) > > children=4 > > #fork=yes > > force_rport=yes // Add recieved port to VIA > > #log_stderror=no // (cmd line: -E) > > > > // Uncomment these lines to enter debugging mode > > fork=no > > log_stderror=yes > > > > check_via=no // (cmd. line: -v) > > dns=no // (cmd. line: -r) > > rev_dns=no // (cmd. line: -R) > > > > log_facility=MY_LOG_FACILITY > > > > // Define the listening ports and addresses > > port=MY_SIP_PORT > > tls_port_no=MY_SIPS_PORT > > > > > > // Define TCP parameters > > tcp_children=4 > > tcp_no_connect=no > > tcp_rd_buf_size=16384 > > tcp_max_connections=2048 > > tcp_connection_lifetime=3605 > > disable_tcp=no > > #!ifdef WITH_TLS > > enable_tls=yes > > #!endif > > > > > > listen=MY_IP4_ADDR > > #!ifdef WITH_LOCALHOST_SIP > > listen=IP4_LOCALHOST > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_IP6_ADDR > > #!ifdef WITH_LOCALHOST_SIP > > listen=IP6_LOCALHOST > > #!endif > > #!endif > > > > #!ifdef WITH_WEBSOCKETS > > listen=MY_WS4_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WS4_ADDR > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_WS6_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WS6_ADDR > > #!endif > > #!endif > > #!ifdef WITH_TLS > > listen=MY_WSS4_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WSS4_ADDR > > #!endif > > #!ifdef WITH_IPV6 > > listen=MY_WSS6_ADDR > > #!ifdef WITH_LOCALHOST_WS > > listen=LOCALHOST_WSS6_ADDR > > #!endif > > #!endif > > #!endif > > #!endif > > > > force_rport=yes // Add received port to VIA > > > > loadmodule "pv.so" // Pseudo-Variable support > > loadmodule "xlog.so" // Formatted logging support > > loadmodule "sl.so" // Stateless request handling > > loadmodule "siputils.so" // SIP utility helpers > > loadmodule "sanity.so" // Sanity check > > loadmodule "maxfwd.so" // Support for Max-Forward header > > loadmodule "textops.so" // Text operations on SIP messages > > loadmodule "textopsx.so" // Extended Text operations on SIP messages > > loadmodule "sdpops.so" // SDP operations > > loadmodule "nathelper.so" // NAT traversal helpers > > loadmodule "rr.so" // Record Route handling > > loadmodule "outbound.so" // Outbound support > > loadmodule "path.so" // Path handling > > > > #!ifdef WITH_TLS > > loadmodule "tls.so" // TLS Support > > #!endif > > > > #!ifdef WITH_WEBSOCKETS > > loadmodule "xhttp.so" // XHTTP support > > loadmodule "websocket.so" // WebSocket support > > #!endif > > loadmodule "rtpengine.so" // RTPEngine control > > > > #!ifdef WITH_DEBUG > > loadmodule "debugger.so" > > #!endif > > > > When I start Kamailio, I see the following in the output to stderr: > > > > Listening on > > udp: 172.16.0.57:5060 > > tcp: 172.16.0.57:5060 > > Aliases: > > tcp: VRTPENGINE:5060 > > udp: VRTPENGINE:5060 > > > > WARNING: no fork mode > > 0(6299) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt > as the io watch method (auto detected) > > > > Here's the problem: > > > > When I connect via UDP, everything works fine. I can connect and see all > SIP traffic. > > > > However, if I try to connect via TCP, I see the initial SYN packet in > attempting to establish a connection. > > > > Kamailio responds with an [RST,ACK] message and the TCP connection is > never established. No further TCP traffic is sent. > > > > Any ideas on what I'm doing wrong? > > <https://www.google.com/maps/search/3400%0D%0A+North+Ashton+Blvd,+Suite+470+…> > > > > *Peter H. Manley, Ph.D.* > > Chief Software Architect > > KomBea Corporation > > 3400 North Ashton Blvd, Suite 470 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > Lehi, Utah 84043 > <https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> > > www.kombea.com<http://www.kombea.com> > <https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> > > > > Office:+1.801.610.5808 > > Mobile +1.801.243.7773 > > > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions > * sr-users at lists.kamailio.org<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> > Important: keep the mailing list in the recipients, do not reply only to > the sender! > Edit mailing list options or unsubscribe: > * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- VoIP Embedded, Inc. http://www.voipembedded.com<http://www.voipembedded.com/>

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Unable to establish TCP connection with Kamailio
by Peter Manley 17 Jun '21

17 Jun '21

Hello, I'm running Kamailio Version 5.3.2 on Ubuntu 20.04. Here my build information: version: kamailio 5.3.2 (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, 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_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 9.2.1 I have the following configuration set in my kamalio.cfg file: ##!define WITH_MYSQL ##!define WITH_AUTH ##!define WITH_USRLOCDB ##!define WITH_TLS ##!define WITH_HOMER ##!define WITH_WEBSOCKETS ##!define WITH_ANTIFLOOD ##!define WITH_IPV6 ##!define WITH_BRIDGE_ON_FAIL ##!define WITH_LOCALHOST_WS ##!define WITH_LOCALHOST_SIP #!substdef "!MY_SIP_PORT!5060!g" #!substdef "!MY_SIPS_PORT!5061!g" #!substdef "!MY_WS_PORT!8088!g" #!substdef "!MY_WSS_PORT!8089!g" #!substdef "!MY_LOG_FACILITY!LOG_DAEMON!g" #!substdef "!MY_IP4_ADDR!172.16.0.57!g" #!substdef "!IP4_LOCALHOST!127.0.0.1!g" #!substdef "!MY_WS4_ADDR!tcp:MY_IP4_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS4_ADDR!tls:MY_IP4_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS4_ADDR!tcp:IP4_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS4_ADDR!tls:IP4_LOCALHOST:MY_WSS_PORT!g" #!ifdef WITH_IPV6 #!substdef "!MY_IP6_ADDR![fe80::d6e4:2f5f:e5b7:7dff]!g" #!substdef "!IP6_LOCALHOST![::1]!g" #!substdef "!MY_WS6_ADDR!tcp:MY_IP6_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS6_ADDR!tls:MY_IP6_ADDR:MY_WSS_PORT!g" #!substdef "!LOCALHOST_WS6_ADDR!tcp:IP6_LOCALHOST:MY_WS_PORT!g" #!substdef "!LOCALHOST_WSS6_ADDR!tls:IP6_LOCALHOST:MY_WSS_PORT!g" #!endif #!substdef "!MY_DOMAIN!rtpengine.kombea.io!g" debug=10 // debug level (cmd line: -dddddddddd) children=4 #fork=yes force_rport=yes // Add recieved port to VIA #log_stderror=no // (cmd line: -E) // Uncomment these lines to enter debugging mode fork=no log_stderror=yes check_via=no // (cmd. line: -v) dns=no // (cmd. line: -r) rev_dns=no // (cmd. line: -R) log_facility=MY_LOG_FACILITY // Define the listening ports and addresses port=MY_SIP_PORT tls_port_no=MY_SIPS_PORT // Define TCP parameters tcp_children=4 tcp_no_connect=no tcp_rd_buf_size=16384 tcp_max_connections=2048 tcp_connection_lifetime=3605 disable_tcp=no #!ifdef WITH_TLS enable_tls=yes #!endif listen=MY_IP4_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP4_LOCALHOST #!endif #!ifdef WITH_IPV6 listen=MY_IP6_ADDR #!ifdef WITH_LOCALHOST_SIP listen=IP6_LOCALHOST #!endif #!endif #!ifdef WITH_WEBSOCKETS listen=MY_WS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WS6_ADDR #!endif #!endif #!ifdef WITH_TLS listen=MY_WSS4_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS4_ADDR #!endif #!ifdef WITH_IPV6 listen=MY_WSS6_ADDR #!ifdef WITH_LOCALHOST_WS listen=LOCALHOST_WSS6_ADDR #!endif #!endif #!endif #!endif force_rport=yes // Add received port to VIA loadmodule "pv.so" // Pseudo-Variable support loadmodule "xlog.so" // Formatted logging support loadmodule "sl.so" // Stateless request handling loadmodule "siputils.so" // SIP utility helpers loadmodule "sanity.so" // Sanity check loadmodule "maxfwd.so" // Support for Max-Forward header loadmodule "textops.so" // Text operations on SIP messages loadmodule "textopsx.so" // Extended Text operations on SIP messages loadmodule "sdpops.so" // SDP operations loadmodule "nathelper.so" // NAT traversal helpers loadmodule "rr.so" // Record Route handling loadmodule "outbound.so" // Outbound support loadmodule "path.so" // Path handling #!ifdef WITH_TLS loadmodule "tls.so" // TLS Support #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" // XHTTP support loadmodule "websocket.so" // WebSocket support #!endif loadmodule "rtpengine.so" // RTPEngine control #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif When I start Kamailio, I see the following in the output to stderr: Listening on udp: 172.16.0.57:5060 tcp: 172.16.0.57:5060 Aliases: tcp: VRTPENGINE:5060 udp: VRTPENGINE:5060 WARNING: no fork mode 0(6299) INFO: <core> [core/tcp_main.c:5042]: init_tcp(): using epoll_lt as the io watch method (auto detected) Here's the problem: When I connect via UDP, everything works fine. I can connect and see all SIP traffic. However, if I try to connect via TCP, I see the initial SYN packet in attempting to establish a connection. Kamailio responds with an [RST,ACK] message and the TCP connection is never established. No further TCP traffic is sent. Any ideas on what I'm doing wrong? Peter H. Manley, Ph.D. Chief Software Architect KomBea Corporation 3400 North Ashton Blvd, Suite 470<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> Lehi, Utah 84043<https://urldefense.proofpoint.com/v2/url?u=https-3A__maps.google.com_-3Fq-3…> www.kombea.com<https://urldefense.proofpoint.com/v2/url?u=http-3A__www.kombea.com_&d=DwMF-…> Office:+1.801.610.5808 Mobile +1.801.243.7773

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ATIS-1000085 STIR/SHAKEN DIV PASSporT
by Steven Wheeler 17 Jun '21

17 Jun '21

My Google searches aren't turning up any results, probably because this standard isn't finalized yet, but is anyone aware of a module which implements DIV PASSporTs for diverted calls? More details on the standard here: https://transnexus.com/blog/2020/shaken-div-std-letter-ballot/ Steven Wheeler

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Re: [SR-Users] Roadmap to the next major release - v5.5.0
by Henning Westerholt 16 Jun '21

16 Jun '21

Hello, If I remember correctly then in Debian Buster, the necessary library is not included and therefore the package cannot be build. You could use bullseye, the upcoming version, of compile by yourself. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: David Villasmil <david.villasmil.work(a)gmail.com> Sent: Wednesday, June 16, 2021 8:54 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Roadmap to the next major release - v5.5.0 I don't see the module in deb http://deb.kamailio.org/kamailio55 buster main Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337 On Wed, Jun 16, 2021 at 9:12 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, if you are referring to the module, its available in this release: https://kamailio.org/docs/modules/5.5.x/modules/secsipid.html Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: sr-users <sr-users-bounces(a)lists.kamailio.org<mailto:sr-users-bounces@lists.kamailio.org>> On Behalf Of David Villasmil Sent: Wednesday, June 16, 2021 10:03 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; miconda(a)gmail.com<mailto:miconda@gmail.com> Subject: Re: [SR-Users] Roadmap to the next major release - v5.5.0 Hello Daniel, Is it possible at all to get secsipid in? Thanks David On Thu, 25 Feb 2021 at 15:21, Daniel-Constantin Mierla <miconda(a)gmail.com<mailto:miconda@gmail.com>> wrote: Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.5.0. The 5.4.0 was release by end of July 2020, therefore we are approaching the end of our usual development cycle, also during the last online devel meeting we considered the 2nd part of spring 2021 as a target time frame for 5.5. To narrow it down, I would propose to freeze the development at the end of March 2021, test during April and maybe beginning of May, then release v5.5.0. So far we have a lot of development to existing components, by the freeze date I expect 3-5 new modules to be in the repo as well. If anyone wants a different time line towards 5.5.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Regards, David Villasmil email: david.villasmil.work(a)gmail.com<mailto:david.villasmil.work@gmail.com> phone: +34669448337

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Roadmap to the next major release - v5.5.0
by Daniel-Constantin Mierla 16 Jun '21

16 Jun '21

Hello, it is the time to plan a bit the road to the next major release, to be versioned 5.5.0. The 5.4.0 was release by end of July 2020, therefore we are approaching the end of our usual development cycle, also during the last online devel meeting we considered the 2nd part of spring 2021 as a target time frame for 5.5. To narrow it down, I would propose to freeze the development at the end of March 2021, test during April and maybe beginning of May, then release v5.5.0. So far we have a lot of development to existing components, by the freeze date I expect 3-5 new modules to be in the repo as well. If anyone wants a different time line towards 5.5.0, let's discuss and choose the one that suits most of the developers. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Using Kamailio with Asterisk/pjsip outbound_publish - early state is not notified
by Barry Flanagan 15 Jun '21

15 Jun '21

Hi Using Kamailio 5.4 with Asterisk 18 and using the pjsip outbound-publish to allow Kamailio to handle BLF for phones. BLF sort of works, in that when a line is on a call, the BLF reflects that. However the early/ringing state does not generate a NOTIFY to the phone, although Asterisk is sending the Publish and is receiving a 200OK reply from Kamailio. I would presume that since basic BLF is working that this early state PUBLISH should generate a NOTIFY but it doesn't Module params are as follows: # ----- presence params ----- modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:192.168.245.250:5060") modparam("presence", "local_log_level", 3) modparam("presence", "publ_cache", 0) modparam("presence", "timeout_rm_subs", 0) modparam("presence", "sip_uri_match", 1) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) modparam("presence_dialoginfo", "force_single_dialog", 0) modparam("presence_dialoginfo", "force_dummy_dialog", 1) Any help appreciated! -Barry

1 0

Kamailio v5.5.1 Released
by Daniel-Constantin Mierla 15 Jun '21

15 Jun '21

Hello, Kamailio SIP Server v5.5.1 stable release is out. This is a maintenance release of the latest stable branch, 5.5, that includes fixes since the release of v5.5.0. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.5.x. Deployments running previous v5.5.x versions are strongly recommended to be upgraded to v5.5.1. For more details about version 5.5.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2021/06/kamailio-v5-5-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.5.1
by Daniel-Constantin Mierla 15 Jun '21

15 Jun '21

Hello, I am considering to release Kamailio v5.5.1 sometime next week, likely on Tuesday or Wednesday (June 15 or 16, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

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Is Kamailio GPL or GPLv2
by Yan Gao 15 Jun '21

15 Jun '21

Hello, I have a question about the license of Kamailio. It is said to be GPL in https://www.kamailio.org/w/: "Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second...." But GPLv2 in https://www.kamailio.org/w/info/ : "Kamailio is released under GNU Public License v2 (GPLv2). Beginning with 3.0.0, the application includes parts of code under BSD license that can be used such as individual components." Which one is the correct description? Thanks & Regards, Yan

2 2

vsf/vst question
by David Escartín 15 Jun '21

15 Jun '21

Hello all! as we knew vsf/vst were added to the RR header when using uac_replace for From/To headers. BUt we observed that some versions ago (we are using now 5.4) that stop happening with this setup modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) modparam("rr", "enable_full_lr", 0) modparam("rr", "enable_double_rr", 1) modparam("rr", "append_fromtag", 1) and only ftag, did are added as param to RR However we have seen that when using multihomed kamailio instance and doing uac_replace, vsf/vst are included also with ftag and did params. Is that behaviour known or expected? thanks a lot and regards david -- <http://www.sonoc.io/> David Escartín NOC engineer *www.sonoc.io* <http://www.sonoc.io/> [image: LinkedIn] <https://www.linkedin.com/company/sonoc> [image: Twitter] <https://twitter.com/sonoc_> [image: Facebook] <https://www.facebook.com/SONOCio/> This e-mail is for the exclusive use of its recipients and may contain business secrets or other confidential or privileged information. Any unauthorised use, copying, publication or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please inform us immediately by replying to this e-mail and delete it, including any attachments or copies on your system. In accordance with the GDPR (EU) 2016/679 and the LOPDGDD 3/2018, we inform you that this e-mail address and/or any other personal data you have provided us with will be treated by SONOC with absolute confidentiality and with the only purpose of providing you with the requested services, due to your condition as a client, supplier or because you have requested information from us at any time. These data will only be kept for as long as required to comply with legal obligations. You can exercise your rights at any time by sending an e-mail to: *dataprotection(a)sonoc.io* <dataprotection(a)sonoc.io>*.* <dataprotection(a)sonoc.io>

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using path in Subscriber table
by Thành Trần Duy 14 Jun '21

14 Jun '21

Hi all, First of all, I would like to present the system model we are applying: - dSIPRouter : Sip relay for all users connected from the internet (installed with RTPEngine to handle media data). - kamailio 5.3.5: core system, registration server. - Asterisk 17: SBC. Sorry if our model seems a bit backwards but we have so many users (>12k configured users, > 3k concurrent registered users, > 500 concurrent users at cps 10), I don't know of any solution that can work with such a large load. With call outbound (from user to PSTN via SBC), everything is fine. But with internal or inbound calls, it wasn't successful. I showed that: - After dSIPRouter, the register message has Path parameter in SIP Header, - Kamailio store sip:user@domain in column contact at location table. - Using modparam registrar with use_path=1, path_mode=0, path_use_received=1, I stored this parameter in DB. - When calling internal or inbound, for the search destination of a phone, I am using sip_trace() function. But the result is always column contact. - With this uri, kamailio wasn't known and sent the invite to proxy. - Final error message: 404. I need one of these solutions: - get path column when result of sip_trace() like sip:user@domain (with phone in internal network, it like sip:user@ipaddress). - write path value to column contact when the register message has it. - sip_trace can get a path instead of contact if it has. Or you can have other solutions, pls help me. We can talk and test via my Webex PMR. Many thanks, Thanh Duy Tran

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set empty custom global variable
by Sergey Safarov 14 Jun '21

14 Jun '21

In Kamailio config I use empty global string variable local.trace_troubleshooting_net = "" descr "trace all SIP messages from given network" When I check variable value I get blank string kamcmd cfg.get local trace_troubleshooting_net I can set new value [root@host]# kamcmd cfg.sets local trace_troubleshooting_net "2600::/48" [root@host]# kamcmd cfg.get local trace_troubleshooting_net 2600::/48 Now I want to set the empty custom global variable using a command kamcmd cfg.sets group trace_troubleshooting_net "" But get error error: 400 - error at parameter 2: expected string type but record doesn't match type How to properly set empty values?

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Kamailio 5.5.0 | CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 26
by Laurent Schweizer 11 Jun '21

11 Jun '21

Dear all, I setup a Kamailio 5.5.0 (from GIT) on a ubuntu Ubuntu 18.04.5 LTS and I have some error with the libssl/ libmysql, seems not happening every time. Regarding the config file, it's a config running correctly on a Kamailio 5.3.x on a centos 7 I have attached the returned error and the result of the core file. Any idea ? BR Laurent version: kamailio 5.5.0 (x86_64/linux) 66370c flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLOCKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 66370c compiled on 07:23:15 Jun 11 2021 with gcc 7.5.0 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: loading modules under config path: /usr/local/lib64/kamailio/modules/ Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: Listening on Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: 80.132.9.112:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: 80.132.9.112:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: 80.132.9.112:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: 80.132.9.112:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tls: 80.132.9.112:5061 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: Aliases: Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tls: zrh02-srv03:5061 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: zrh02-srv03:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: tcp: zrh02-srv03:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: zrh02-srv03:6000 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: udp: zrh02-srv03:5060 Jun 11 21:35:45 zrh02-srv03 kamailio[7129]: *: sipbkp.demo.ch:* Jun 11 21:35:45 zrh02-srv03 /usr/local/sbin/kamailio[7135]: WARNING: tls [tls_init.c:793]: tls_h_mod_init_f(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 13107200 and 6553600 bytes Jun 11 21:35:45 zrh02-srv03 /usr/local/sbin/kamailio[7135]: NOTICE: tls [tls_domain.c:1136]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Jun 11 21:35:46 zrh02-srv03 kernel: [304361.950608] traps: kamailio[7147] general protection ip:7f5d209a344b sp:7ffe6ae5adb8 error:0 in libcrypto.so.1.1[7f5d207b5000+29b000] Jun 11 21:35:46 zrh02-srv03 systemd[1]: Started Kamailio SIP Server. Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7161]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 26 Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7135]: ALERT: <core> [main.c:788]: handle_sigs(): child process 7147 exited by a signal 11 Jun 11 21:35:54 zrh02-srv03 /usr/local/sbin/kamailio[7135]: ALERT: <core> [main.c:791]: handle_sigs(): core was generated Jun 11 21:35:54 zrh02-srv03 systemd[1]: kamailio.service: Main process exited, code=exited, status=1/FAILURE Jun 11 21:35:54 zrh02-srv03 systemd[1]: kamailio.service: Failed with result 'exit-code'. This GDB was configured as "x86_64-linux-gnu". Type "show configuration" for configuration details. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/. Find the GDB manual and other documentation resources online at: http://www.gnu.org/software/gdb/documentation/. For help, type "help". Type "apropos word" to search for commands related to "word"... Reading symbols from /usr/local/sbin/kamailio...done. [New LWP 7147] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -m 5000 -M 128 -P /run/kamailio/kamailio.pid -f /confi'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f5d209a344b in X509_LOOKUP_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f5d209a344b in X509_LOOKUP_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f5d209a3a5b in X509_STORE_CTX_get_by_subject () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f5d209a418d in X509_STORE_CTX_get1_issuer () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #3 0x00007f5d209a9207 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #4 0x00007f5d209a9f96 in X509_verify_cert () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #5 0x00007f5d20caf337 in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #6 0x00007f5d20cd037b in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #7 0x00007f5d20cd2995 in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #8 0x00007f5d20ccc48c in ?? () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #9 0x00007f5d20cb84e4 in SSL_do_handshake () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #10 0x00007f5d215cbdb1 in ?? () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #11 0x00007f5d2159f4de in mysql_real_connect () from /usr/lib/x86_64-linux-gnu/libmysqlclient.so.20 #12 0x00007f5d21b410b1 in db_mysql_new_connection (id=0x7f5d21ee2b30) at km_my_con.c:157 #13 0x00007f5d211456dc in db_do_init2 (url=0x7f5d1e9dd4e0 <ul_db_url>, new_connection=0x7f5d21b3f6a4 <db_mysql_new_connection>, pooling=DB_POOLING_PERMITTED) at db.c:317 #14 0x00007f5d211443f9 in db_do_init (url=0x7f5d1e9dd4e0 <ul_db_url>, new_connection=0x7f5d21b3f6a4 <db_mysql_new_connection>) at db.c:270 #15 0x00007f5d21b3459e in db_mysql_init (_url=0x7f5d1e9dd4e0 <ul_db_url>) at km_dbase.c:202 #16 0x00007f5d1e7a1987 in child_init (_rank=-1) at usrloc_mod.c:471 #17 0x000055ddd6c45602 in init_mod_child (m=0x7f5d21e03818, rank=-1) at core/sr_module.c:864 #18 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e03bb8, rank=-1) at core/sr_module.c:860 #19 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e04498, rank=-1) at core/sr_module.c:860 #20 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e056a8, rank=-1) at core/sr_module.c:860 #21 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06390, rank=-1) at core/sr_module.c:860 #22 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06b38, rank=-1) at core/sr_module.c:860 #23 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e06fb8, rank=-1) at core/sr_module.c:860 #24 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e075f0, rank=-1) at core/sr_module.c:860 #25 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e090c0, rank=-1) at core/sr_module.c:860 #26 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e09500, rank=-1) at core/sr_module.c:860 #27 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0a400, rank=-1) at core/sr_module.c:860 #28 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0aa70, rank=-1) at core/sr_module.c:860 #29 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0b0b0, rank=-1) at core/sr_module.c:860 #30 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0b5d8, rank=-1) at core/sr_module.c:860 #31 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0bdd0, rank=-1) at core/sr_module.c:860 #32 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0c968, rank=-1) at core/sr_module.c:860 #33 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e0f7f8, rank=-1) at core/sr_module.c:860 #34 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e105d8, rank=-1) at core/sr_module.c:860 #35 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e109b0, rank=-1) at core/sr_module.c:860 #36 0x000055ddd6c4521d in init_mod_child (m=0x7f5d21e10ea0, rank=-1) at core/sr_module.c:860 #37 0x000055ddd6c45ef9 in init_child (rank=-1) at core/sr_module.c:909 #38 0x000055ddd6b9eebe in fork_process (child_id=-1, desc=0x55ddd6ee7847 "slow timer", make_sock=1) at core/pt.c:337 #39 0x000055ddd6abaa62 in main_loop () at main.c:1814 #40 0x000055ddd6ac6095 in main (argc=9, argv=0x7ffe6ae5d298) at main.c:3053

1 0

Kamailio SBC+ Freeswitch as media server
by John Tuxies 09 Jun '21

09 Jun '21

in a Debian system i am running Kamailio 5.5 installed from repo and with the basic setup i have 2 users 101 & 102 to be able to call each other. So far so good. i added Freeswitch in the same system and tried to follow this guide http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc even though it is old and there are many changes. Freeswitch is installed from deb repo as well. after following the guide i tried to adapt to newer versions of kamailio and freeswitch, but i am getting errors. The problem is that the users now are able to register to Kamailio, but unable to call each other. I can see users being registered in Kamailio: *root@kamfs: ~ $ kamctl online101102root@kamfs: ~ $* while when 101 calls 102 i am getting: *2021-06-08 10:35:19.015670 [NOTICE] switch_channel.c:1118 New Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [f3b8f5a0-652a-4640-b2d3-9b7cc21d7f91]2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_NEW (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [INFO] sofia.c:10362 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> receiving invite from 192.168.1.15:5060 <http://192.168.1.15:5060> version: 1.10.6 -release-18-1ff9d0a60e 64bit call-id: ZDFiZWFkZmVmYjgxZDEwOTgyYTYxZjkxNzkwMzlmYmQ.2021-06-08 10:35:19.015670 [DEBUG] sofia.c:10456 verifying acl "domains" for ip/port 192.168.1.15:0 <http://192.168.1.15:0>.2021-06-08 10:35:19.015670 [WARNING] sofia.c:10569 IP 192.168.1.15 Rejected by acl "domains"2021-06-08 10:35:19.015670 [NOTICE] sofia.c:2445 Hangup sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [CS_NEW] [CALL_REJECTED]2021-06-08 10:35:19.015670 [DEBUG] sofia.c:1549 Channel is already hungup.2021-06-08 10:35:19.015670 [DEBUG] sofia.c:1549 Channel is already hungup.2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:604 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State NEW2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_HANGUP (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:848 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Callstate Change DOWN -> HANGUP2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State HANGUP2021-06-08 10:35:19.015670 [DEBUG] mod_sofia.c:453 Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> hanging up, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:60 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard HANGUP, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:850 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State HANGUP going to sleep2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State Change CS_HANGUP -> CS_REPORTING2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:585 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_REPORTING (Cur 1 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State REPORTING2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:174 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard REPORTING, cause: CALL_REJECTED2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:936 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State REPORTING going to sleep2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:611 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State Change CS_REPORTING -> CS_DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_session.c:1736 Session 5 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Locked, Waiting on external entities2021-06-08 10:35:19.015670 [NOTICE] switch_core_session.c:1754 Session 5 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Ended2021-06-08 10:35:19.015670 [NOTICE] switch_core_session.c:1758 Close Channel sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> [CS_DESTROY]2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) Running State Change CS_DESTROY (Cur 0 Tot 5)2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State DESTROY2021-06-08 10:35:19.015670 [DEBUG] mod_sofia.c:364 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> SOFIA DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:181 sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15> Standard DESTROY2021-06-08 10:35:19.015670 [DEBUG] switch_core_state_machine.c:749 (sofia/internal/101(a)192.168.1.15 <101(a)192.168.1.15>) State DESTROY going to sleepfreeswitch@kamfs>* i am attaching the kamailio.cfg file and the whole /etc/freeswitch folder, if someone could help me to solve this issue, please. Kind regards, John

3 2

Any way to force socket used to reach databases ?
by David VILLAUME 08 Jun '21

08 Jun '21

Hello, I’m trying to initiate a redis connection in ndb_redis for topos_redis on a keepalived slave host, but host tries to use the VIP instead of the second socket I defined. Is there a way to influence the ‘database source socket’ so the server can connect with a secondary address even if it does not own the keepalived VIP (tried mhomed options without improvment) ? Regards, David

2 1

List of Header Names
by Evan Verma 07 Jun '21

07 Jun '21

hello! new to working with sip/kamailio. I am currently trying to get a list of all of the header names on the invite the SBC receives. I am having trouble finding a way to do this. Does anyone know of a good solution?

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