- sr-users - kamailio.org

Some questions about drouting module
by Андрей Яри н 23 Aug '21

23 Aug '21

Hello! I test module on Centos 7, kamailio version 5.5 and found some strange things: 1) Definition of table dr_gateways at https://kamailio.org/docs/modules/5.5.x/modules/drouting.html#idm98 - in DB there is field 'attrs' and only notice about it is module param 3.8. attrs_avp but it doesnt describe table field. In OpenSIPS docs i found description * attributes (not used by DR engine, but only pushed to script level when routing to this GW) In kamailio this field is deprecated or its only docs misses param? 2) Previous moment was found when i try to resolve: WARNING: drouting [dr_load.c:529]: dr_load_routing_info(): route <1> does not exist "Definition of dr_rules table" says that field has type of integer, but realy in database its varchar and i think its old docs. But i use KEMI question is - how i can tell module that it must be called in KEMI, not in native? 3) Every time when i call 'kamcmd drouting.reload' i get message CRITICAL: <core> [core/mem/q_malloc.c:502]: qm_free(): BUG: bad pointer 0x7f3e4cea04c0 (out of memory block!) called from core: core/utils/tmrec.c: tmrec_free(393) - ignoring I rollback to 5.4 and still get this error. As i understand its old module and its can be internal error - how critical this error can be? 4) Also planned commands like drouting.dump gws|gw_lists|rules? -- С уважением, Ярин Андрей Анатольевич инженер, группы управления телефонией Группа компаний «Интерсвязь» e-mail: yarin-aa(a)intersvyaz.net

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Kamailio as basic dispatcher for load balancing
by Philipp Trenz 23 Aug '21

23 Aug '21

Hey there, I’m Philipp, an IT Systems Engineer from Potsdam, Germany and want to deepen my knowledge in SIP communication on scale. Currently, my setup is quite simple: A single stateless Asterisk instance, fully managed via ARI, which registers to multiple VoIP providers, only processes incoming calls to play audio files (no outgoing or conference calls). Now, with growing load on the single Asterisk instance, I would like to have one or two Kamailio instances, which will balance the load of incoming calls to multiple Asterisk instances. On which Asterisk instance they end up is irrelevant for me, as the calls can be processed on every instance. I looked into some Kamailio and Astricon conference speeches and checked out some tutorials. But still I’m not quite sure where to start. So here are some basic questions which would really help me out to get going: As I don’t want to have registrations of all my Asterisk instances at the VoIP providers, I should use the UAC module to let Kamailio do the registering to the VoIP providers and register my Asterisk instances to Kamailio, right? Will Kamailio automatically apply new outbound registrations when added to the database or do I have to trigger that manually? Should I use one or two instances of Kamailio, how hard is it to configure them fail-safe? For writing my first config file, should I start blank or from the standard config, what’s best practice? Thanks in advance, looking forward to your replies! Philipp

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How tell if rtp packets are being relayed through rtpengine in kamailio
by Faiz Fazaluddin 23 Aug '21

23 Aug '21

Hi, I want to see if rtp packets are being relayed through kamailio. I am attmepting to connect rtpengine with kamailio, but am not sure if it is working, how can I tell? I have the kamailio server on a different VM than the clients. In wireshark, no rtp packets show on the server VM and rtp packets only seem to be going directly through the client VM. Rtp engine mentions being unable to find hashtables when attempting to implement a delete_node command. Thank you very much for your help, Faiz

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Register Forwarding
by Thomas Grande 21 Aug '21

21 Aug '21

Hi there! I bumped into this post to perform forwarding of REGISTER requests and then saving a local cache on 2xx replies from main Registrar: https://lists.kamailio.org/pipermail/sr-users/2020-October/110779.html I think I understand all the steps described, but some features I need are missing: - How to change the Via headers to perform topology hiding? I understand TOPOS and TOPOH do not work on these types of messages. - Change contact header so that registrar responses traverse the Kamailio box. (Use textops I suppose?) - Besides, is this approach still the best in comparison to OpenSIPs mid-registrar module? Thanks in advance, -- *Thomás Alimena Del Grande* Engenharia - Aligera Tel. 51 3500-0121

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record_route() from websocket UA to tls UA
by Juha Heinanen 20 Aug '21

20 Aug '21

I noticed that when jsSIP UA that has registered over wss calls another SIP UA that has registered over tls, record_route() adds only one Route URI to outgoing INVITE (example below). This causes BYE to fail. This issue may be caused by the fact that both UAs register over the same Kamailio tls listening socket. Still two Route URIs should be added in the same way as is done when one UA registers over tcp and the other over udp. -- Juha --------------------------------------------------------------------- 09:25:43.639833 wss:87.95.73.155:19458 wss:192.27.134.1:5061 INVITE sip:foo@test.tutpro.com SIP/2.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;branch=z9hG4bK2639638 Max-Forwards: 69 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5> ... 09:25:43.649462 tls:192.27.134.1:5061 tls:87.95.73.155:19461 INVITE sip:foo-0x793ee87a90@10.158.141.103:38378;transport=tls SIP/2.0 Record-Route: <sip:192.27.134.1:5061;transport=ws;sn=ext_tls;lr;n2;dtlsf=avp;pm=0;ice> Via: SIP/2.0/TLS 192.27.134.1:5061;branch=z9hG4bK637c.7e15ef5b84ea054cfc8ea3d4e860521f.0 Via: SIP/2.0/WSS jovobf9n4svm.invalid;rport=19458;received=87.95.73.155;branch=z9hG4bK2639638 Max-Forwards: 68 To: <sip:foo@test.tutpro.com> From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb Call-ID: ovhd9fu1fl1a66nec011 CSeq: 1387 INVITE Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>

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install latest sems-server on debian 10 buster
by Mojtaba 18 Aug '21

18 Aug '21

Hello there, Does anybody have experience installing the latest SEMS-Server (1.6.0) on debian 10 buster? I tried to install both 1.6.0. and 1.7-dev and some issues are occurred, but i installed the version 1.3.1 on debian 8 before. Any help would be appreciated. Thanks with best regards -- --Mojtaba Esfandiari.S

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How to insert user=phone header INVITE TO FROM
by César Matheus 18 Aug '21

18 Aug '21

Hi list. mY provider asks me to insert user = phone in my header. because it requires to receive my INVITE with user = phone in TO and FROM However, I add the user = phone in my configuration like this: $ ru = "sip:" + $ rU + "@" + $ sel (cfg_get.pstn1.gw_ip) + ":" + $ sel (cfg_get.pstn1.gw_port) + "; user = phone"; but only the INVITE appears, not in the TO or FROM. INVITE sip: 09872323232(a)X.Y.X.K: 5060; user = phone SIP / 2.0 From: <sip: XXXXYYYYY(a)172.20.12.4: 5060>; tag = 15552444533876 To: <sip: 09872323232(a)172.20.12.2: 5060> You can help me solve it. Thanks. -- César Matheus

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crash in timer
by Juha Heinanen 17 Aug '21

17 Aug '21

Any idea what could cause the crash shown in the bt below? This is with Kamailio 5.5. -- Juha [New LWP 12061] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". bt 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 0x0000563d3b390c5d in timer_list_expire (t=1584008014, h=0x7f7a1e76bb40, slow_l=0x7f7a1e76d680, slow_mark=63909) at core/timer.c:846 846 core/timer.c: No such file or directory. (gdb) bt full #0 0x0000563d3b390c5d in timer_list_expire (t=1584008014, h=0x7f7a1e76bb40, slow_l=0x7f7a1e76d680, slow_mark=63909) at core/timer.c:846 tl = 0x7f7a226ecbd8 ret = 0 #1 0x0000563d3b391219 in timer_handler () at core/timer.c:922 saved_ticks = 1584008014 run_slow_timer = 0 i = 421 __func__ = "timer_handler" #2 0x0000563d3b391742 in timer_main () at core/timer.c:961 No locals. #3 0x0000563d3b2c94de in main_loop () at main.c:1839 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=46.182.160.20:5060\000\000\000\000\003\000\000\000 \000\000\000\000\071CU\356xL\375\000\000\000\000\000\000\000\000\210\300m;=V\000\000\000\000\000\000\000\000\000\000\260\334\371Jz\177\000\000\263\331r;=V\000\000\000\000\000 \000\000\000\000\340\b3;\374\177\000\000\t\365U;=V\000" nrprocs = 8 woneinit = 1 __func__ = "main_loop" #4 0x0000563d3b2d44aa in main (argc=17, argv=0x7ffc3b330e28) at main.c:3053 cfg_stream = 0x563d3b9ea290 c = -1 r = 0 tmp = 0x7ffc3b331f3f "" tmp_len = 1260628273 port = 32634 proto = 0 ahost = 0x0 aport = 0 --Type <RET> for more, q to quit, c to continue without paging--c options = 0x563d3b6dece0 ":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 = 1970794883 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0xffff00001f80 <error: Cannot access memory at address 0xffff00001f80> st = {st_dev = 21, st_ino = 21649, 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 = 1620846563, tv_nsec = 71986646}, st_mtim = {tv_sec = 1628122861, tv_nsec = 116913769}, st_ctim = {tv_sec = 1628122861, tv_nsec = 616909103}, __glibc_reserved = {0, 0, 0}} tbuf = "\000\000\000\000\000\000\000\000\b\000\000\000\000\000\000\000\060\067&Kz\177\000\000\000\000\000\000\000\000\000\000\240\241>;\374\177\000\000\364:$Kz\177\000\000\b\000\000\000\000\000\000\000\230\242>;\374\177\000\000\060\067&Kz\177\000\000\310\v3;\374\177\000\000\304\v3;\374\177", '\000' <repeats 18 times>, "\230\242>;\374\177\000\000\250\241>;\374\177\000\000'\256\002Kz\177\000\000&\260be\000\000\000\000\060\r3;\374\177\000\000\300\212\225\001\000\000\000\000\220\f3;\374\177\000\000\200\f3;\374\177\000\000\310\v3;\374\177\000\000X7&Kz\177\000\000\001", '\000' <repeats 31 times>... option_index = 0 long_options = {{name = 0x563d3b6e0f3f "help", has_arg = 0, flag = 0x0, val = 104}, {name = 0x563d3b6dc4e1 "version", has_arg = 0, flag = 0x0, val = 118}, {name = 0x563d3b6e0f44 "alias", has_arg = 1, flag = 0x0, val = 1024}, {name = 0x563d3b6e0f4a "subst", has_arg = 1, flag = 0x0, val = 1025}, {name = 0x563d3b6e0f50 "substdef", has_arg = 1, flag = 0x0, val = 1026}, {name = 0x563d3b6e0f59 "substdefs", has_arg = 1, flag = 0x0, val = 1027}, {name = 0x563d3b6e0f63 "server-id", has_arg = 1, flag = 0x0, val = 1028}, {name = 0x563d3b6e0f6d "loadmodule", has_arg = 1, flag = 0x0, val = 1029}, {name = 0x563d3b6e0f78 "modparam", has_arg = 1, flag = 0x0, val = 1030}, {name = 0x563d3b6e0f81 "log-engine", has_arg = 1, flag = 0x0, val = 1031}, {name = 0x563d3b6e0f8c "debug", has_arg = 1, flag = 0x0, val = 1032}, {name = 0x563d3b6e0f92 "cfg-print", has_arg = 0, flag = 0x0, val = 1033}, {name = 0x563d3b6e0f9c "atexit", has_arg = 1, flag = 0x0, val = 1034}, {name = 0x0, has_arg = 0, flag = 0x0, val = 0}} __func__ = "main"

1 0

Info: First group of presentations at Kamailio World Online 2021
by Daniel-Constantin Mierla 17 Aug '21

17 Aug '21

Hello, the first group of presentations selected for the next Kamailio World Online, September 1-2, 2021, has been published, aiming to provide a good knowledge base of using Kamailio for different deployment types: segregation of networks with an SBC-like role, controlling robots, RTP streams management with RTPEngine, configuration variables or transformations, SIP attacks handling and asynchronous SIP routing. You can find more details at: * https://kamailioworld.com/k09-online/ The schedule will be completed during the next days, with another group of interesting sessions and open discussion panels. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

No docker hub automated builds
by Sergey Safarov 14 Aug '21

14 Aug '21

I found that not more automated builds available on docker hub https://www.docker.com/blog/changes-to-docker-hub-autobuilds/ Looks like we need to make some changes for the alpine docker image. Maybe use github actions for this task? https://docs.docker.com/ci-cd/github-actions/ Sergey

1 0

RTPEngine: Async connection/communication between Kamailio and RTPEngine
by Mitesh Thakkar 12 Aug '21

12 Aug '21

Hi, Couple of week before I have posted feature request - https://github.com/kamailio/kamailio/issues/2807 As suggested, I am trying to figure out how I can achieve it with async or sworker modules. Can someone help me to understand how to use those modules to achieve async connectivity? Regards, Miteshkumar Thakkar

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auth challenge freeswitch with Zopier
by Joli Martinez 10 Aug '21

10 Aug '21

Hello, I am using dsiprouter as a public facing server for pass through registrations to a fusion pbx box. This setup works great and has for a long time now. The issue I am having is that we are switching soft phone clients to copier. Zopier uses a push server. When an incoming call comes into the freeswitch server, it fires off an invite to the Zopier registered client. For some reason ONLY in this scenario Kamailio is doing an auth challenge to Freeswitch preventing to call to be sent to the soft phone. How can I stop this from happening? Other soft phones and hard phones work great with inbound calls except for Zopier. thanks,

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Integration with multiple MS Teams instances
by Володимир Іванець 10 Aug '21

10 Aug '21

Hello all! I was able to connect Kamailio with MS Teams and now trying to add one more Teams instance. It looks like I have some misconfiguration or there is a bug. My test server has 2 domain records pointing at it (kamailio.domain1.com and kamailio.domain2.com). My tls.cfg configuration file looks like this. As you can see the Default section is configured with a kamailio.domain1.com sertificate: *[server:default]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[client:default]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[server:172.16.30.206:5062 <http://172.16.30.206:5062>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *server_name = "kamailio.domain1.com <http://kamailio.domain1.com>"* *server_id = "**"kamailio.domain1.com <http://kamailio.domain1.com>"* *[client:172.16.30.206:5062 <http://172.16.30.206:5062>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain1.com/server/key.pem <http://kamailio.domain1.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain1.com/server/cert.pem <http://kamailio.domain1.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain1.com/CA/cert.pem <http://kamailio.domain1.com/CA/cert.pem>* *[server:172.16.30.206:5063 <http://172.16.30.206:5063>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain2.com/server/key.pem <http://kamailio.domain2.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain2.com/server/cert.pem <http://kamailio.domain2.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain2.com/CA/cert.pem <http://kamailio.domain2.com/CA/cert.pem>* *server_name = "kamailio.domain2.com <http://kamailio.domain2.com>"* *server_id = "**"kamailio.domain2.com <http://kamailio.domain2.com>"* *[client:172.16.30.206:5063 <http://172.16.30.206:5063>]* *method = TLSv1.0+* *require_certificate = no* *verify_certificate = no* *private_key = /var/kamailio/certificates/kamailio.domain2.com/server/key.pem <http://kamailio.domain2.com/server/key.pem>* *certificate = /var/kamailio/certificates/kamailio.domain2.com/server/cert.pem <http://kamailio.domain2.com/server/cert.pem>* *ca_list = /var/kamailio/certificates/kamailio.domain2.com/CA/cert.pem <http://kamailio.domain2.com/CA/cert.pem>* The dispatcher configuration table looks like this: +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ | 1 | 1 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 3 | socket=tls:172.16.30.206:5062;ping_from=sip:kamailio.domain1.com | MS Teams 1 | | 2 | 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 3 | socket=tls:172.16.30.206:5063;ping_from=sip:kamailio.domain2.com | MS Teams 2 | +----+-------+----------------------------------------------+-------+----------+--------------------------------------------------------------------+-------------+ When Kamailio is started only connection with the first trunk is established: *# kamcmd tls.list* *{* * id: 1* * timeout: 0* * src_ip: 52.114.75.24* * src_port: 5061* * dst_ip: 172.16.30.206* * dst_port: 0* * cipher: ECDHE-RSA-AES256-GCM-SHA384 TLSv1.2 Kx=ECDH Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* *{* * id: 2* * timeout: 0* * src_ip: 52.114.75.24* * src_port: 7810* * dst_ip: 172.16.30.206* * dst_port: 5062* * cipher: AES256-GCM-SHA384 TLSv1.2 Kx=RSA Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* *{* * id: 3* * timeout: 596* * src_ip: 52.114.75.24* * src_port: 7811* * dst_ip: 172.16.30.206* * dst_port: 5062* * cipher: AES256-GCM-SHA384 TLSv1.2 Kx=RSA Au=RSA Enc=AESGCM(256) Mac=AEAD* * ct_wq_size: 0* * enc_rd_buf: 0* * flags: 2* * state: established* *}* Here is what I can see in Kamailio log file when it sends an OPTIONS request to the second trunk. Kamailio uses Default tls configuration and MS Teams don't accept it: *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: ALERT: <script>: == TRACE. tm:local-request. fs is tls:172.16.30.206:5063 <http://172.16.30.206:5063>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tm [uac.c:352]: t_run_local_req(): apply new updates without Via to sip msg* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/msg_translator.c:1796]: check_boundaries(): no multi-part body* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request:* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:612]: parse_msg(): method: <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): uri: <sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK169b.6411b4c3000000000000000000000000.0>; state=16* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [47]; uri=[sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (47)[<sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>^M* *], to tag (0)[]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): SIP Request:* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:612]: parse_msg(): method: <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:614]: parse_msg(): uri: <sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:616]: parse_msg(): version: <SIP/2.0>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK169b.6411b4c3000000000000000000000000.0>; state=16* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=10* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [47]; uri=[sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (47)[<sip:sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>;transport=tls>^M* *], to tag (0)[]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS>* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tm [uac.c:189]: uac_refresh_hdr_shortcuts(): cseq: [CSeq: 10]* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 52.114.75.24* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5061, type 3, socket -1* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: <core> [core/tcp_main.c:1498]: tcpconn_add(): hashes: 2831:67:0, 1* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:162]: tls_get_connect_server_name(): xavp with outbound server name not found* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:142]: tls_get_connect_server_id(): xavp with outbound server id not found* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_server.c:228]: tls_complete_init(): Using initial TLS domain TLSc<default> (dom 0x7f35509da688 ctx 0x7f3550b7a568 sn [])* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_domain.c:1177]: tls_lookup_private_key(): Private key lookup for SSL_CTX-0x7f3550b7a568: (nil)* *Jul 29 16:46:14 kamailio /usr/sbin/kamailio[11809]: DEBUG: tls [tls_domain.c:747]: sr_ssl_ctx_info_callback(): SSL handshake started* *...* If I change the Default configuration to use kamailio.domain2.com certificate, the second trunk will connect but the first one will fail. I tried to set "$xavp(tls=>server_name)" and "$xavp(tls[0]=>server_id)" variables to the event_route[tm:local-request] section but log still stated that server Name and ID were not found. Can someone please point me in the right direction, how can I make Kamailio use the correct certificates when establishing multiple TLS connections? Thanks a lot! Regards, Volodymyr Ivanets

3 12

Ringing Timeout
by Edward Romanenco 09 Aug '21

09 Aug '21

Hey all, I'm trying to find the best way for settings a timeout for the 'ringing' stage of a call - meaning - I would like to wait for 20 seconds between receiving status 180 / 183 / early media from the remote end and 200, if I fail to receive the expected response in this timeframe - call should be dropped. Most of the documented timeouts I saw count between the first invite to any of the aforementioned stages - but I'm actually looking for the opposite. Edward

2 1

help
by Said Hassani 08 Aug '21

08 Aug '21

Dear Team, I would like to know how to link Kamailio with openIMSCore?I would like to use Kamailio as an Application server.Please guide me or if there is someone who has worked on this subject, I would like to benefit from his help.

3 4

XCAP server with simservs support
by H Yavari 05 Aug '21

05 Aug '21

Hi to all, As I found there isn't any support for this AUID : simservs.ngn.etsi.org in Kamailio.Anyone can confirm that we can't use Kamailio as mmtel for IMS with the current xcap_server module?!!Thank you. Regards,Hossein

2 2

no debug trace of incoming request
by Juha Heinanen 05 Aug '21

05 Aug '21

Is it somehow possible that ngrep shows incoming INVITE arriving over TCP to Kamailio's listening address and port, but there is no debug trace of the request (e.g. receive_msg(): --- received sip message ...) in syslog? -- Juha

3 3

Issues with TOPOS when not relaying an in-dialog request
by Marrold 05 Aug '21

05 Aug '21

Hi, I am currently using TOPOS to make Kamailio behave more like a B2BUA from the clients perspective. It's working well, however I have discovered a scenario where it fails. To aid with some interoperability I am sending (sl_send_reply) 200 OK to an in-dialog SUBSCRIBE request during a call. Once this has happened, the subsequent BYE from the B leg goes to the incorrect (Private) IP. If I disable this SUBSCRIBE, or relay it to the B leg, the BYE goes to the expected IP address. Does anyone know where I am going wrong? I have tried not calling record_route() for these messages but the result is the same. I've attached the ladder diagram of the call below. Note the subscribe is not relayed in this scenario. Thanks! [image: image.png]

4 8

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 04 Aug '21

04 Aug '21

> Serdar, > > Have you tried to clean variables before calling new ds_select_domain(), > that are using by dispatcher module failover? > > Like > https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.… <https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…> > and so on? > > But as I got, you're saying, that calling ds_select_domain() with > different setid's in a case of fail, not really fails, but using "old" > available destinations from previous attempt? > > Regards, > Igor Igor, also thanks for your interest. > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel I remove xavp value using "xavp_rm("_dsdst_")", i supposed that my problem was solved but in my other tests, i recognised that xavp_rm removed the first index of "xavp(_dsdst_)" not all previous destinations(indexes). after researching, i found a post at https://lists.kamailio.org/pipermail/sr-users/2020-May/109192.html and i removed all ellements of list as below, if(defined $xavp(_dsdst_)) { while($xavp(_dsdst_[0]) != $null) { xlog("L_INFO", "--- Loaded Dispatchers --- Grp : $xavp(_dsdst_[0]=>grp)\n"); xlog("L_INFO", "--- Loaded Dispatchers --- Uri : $xavp(_dsdst_[0]=>uri)\n"); $xavp(_dsdst_[0]) = $null; } } it worked but i am not sure it is a best solution. Is there another simple way to delete all indexes of xavp? Best regards, Serdar

1 0

Converting Cisco SIP NOTIFY DTMF to SIP INFO (Binary operations)
by Marrold 04 Aug '21

04 Aug '21

Hi, I am using an old Cisco 2811 with an FXS card and trying to get it to convert between POTS and SIP. Mostly for "fun". For the most part it works, however there are some problems with DTMF in the call. Because the end device uses pulse dialling it seems that only the Cisco Proprietary SIP NOTIFY method of sending DTMF is supported. It sends 4 bytes in the message body that indicate the DTMF digit sent. I'd like to convert it to the more standard SIP INFO method. The docs about the encoding are here - https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/sip/configuration/1… I looked at the transformations but couldn't find anything relevant. Is it possible to extract the binary from the body and decode it? Thanks Matthew

1 0

which is better for B2BUA
by Mojtaba 04 Aug '21

04 Aug '21

Hello there, According to , https://lists.kamailio.org/pipermail/sr-users/2016-March/092058.html, which talked about B2BUA (just signalling) in Kamailio. As i have experienced working with SEMS, freeswitch and Kamailio while using B2BUA feature, Each of them have pros and cons: 1- The sems is a light sip engine server with several applications (like as sbc) for using b2bua. All incoming and outgoing calls could go to sems server for doing b2bua like this: Incoming<=======>Kamailio<========>Sems<========>Kamailio<=======>outgoing 2- In sems, you could disable rtp realying. It forces sems to work just as b2bua without anchoring RTP 3- Easy to use different active profiles in routing. Just a couple of things there are in SEMS. For example, the sems adds itself (local IP) in Via header, and it couldn't be common in b2bua. like this: . . . INVITE sip:200@cloud.domain.com;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 10.10.10.200;branch=z9hG4bK2f53.07a3fd9edaa8c8d609ab2ac6b01a087f.0 *================================>Kamailio public IP* *Via: SIP/2.0/UDP 127.0.0.1:5080;received=127.0.0.1;branch=z9hG4bKsPRMNast;rport=5080 ===============================>private loopback sems ip* From: <sip:100@cloud.domain.com ;transport=UDP>;tag=6FFDB493-60EABB3600016ECD-379F9700 To: <sip:200@cloud.domain.com;transport=UDP> CSeq: 10 INVITE Call-ID: Y2M1ODQxNTZmMjdkZWZjN2U5MmMyYjBmN2Y2OGY1ODQ._leg2 Route: <sip:mo@127.0.0.1:5060;lr> Max-Forwards: 69 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Z 3.3.25608 r25552 Content-Type: application/sdp Contact: <sip:127.0.0.1:5080;transport=udp> Content-Length: 246 . . . For this reason, Is there a way to avoid this issue? I know it is possible to do this by using other modules like textops and retransformation ? And why in Kamailio, there is no b2bua module to perform all b2bua functionality, yet? Thanks with Best Regards. -- --Mojtaba Esfandiari.S

6 9

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 03 Aug '21

03 Aug '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 2

Re: [SR-Users] How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 02 Aug '21

02 Aug '21

Hello Daniel, I removed xavps using with xavp_rm and problem solved. Thanks for your helping Serdar > Hello, > > delete the xavps based on the names you set via modparams xavp_dst and > xavp_ctx. > > Cheers, > Daniel > > On 30.07.21 11:14, Serdar GÜÇLÜER wrote: > >/Hello all, />//>/I have a trouble that is related to the way of usage of dispatcher > module. />//>/I am working with Kamailio 5.3.2 and using dispatcher module as load />/balancer to route calls to the media gateways. />//>/My dispatching routes are below as simply, />//>/route[DISPATCH] { />/ if(ds_select_domain("2", "4")) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable - No MGW"); />/ exit; />/ } />/} />/route[MYRELAY] { />/ t_on_failure("MYFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if ($T_reply_code == 408 || $T_reply_code == 503) { />/ if(ds_next_domain()) { />/ route(MYRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/ } else { />/ if(ds_select_domain("6", "4")) { />/ route(MYVMRELAY); />/ } else { />/ send_reply(486,"Busy"); />/ exit; />/ } />/ } />/} />/route[MYVMRELAY] { />/ t_on_failure("MYVMFAILURE"); />/ if(!t_relay()) { />/ sl_reply_error(); />/ } />/ exit; />/} />/failure_route[MYVMFAILURE] { />/ route(NATMANAGE); />/ revert_uri(); />/ if (t_is_canceled()) { />/ exit; />/ } />/ if(ds_next_domain()) { />/ route(MYVMRELAY); />/ } else { />/ send_reply("503","Service Unavailable"); />/ exit; />/ } />/} />//>//>/I have multiple media gateways(setid=2) and voicemail />/servers(setid=6). As can be seen from the configuration, />/after first routing, for transaction reply code except 408 or 503, I />/am routing call to the voicemail server using dispatcher. />/In that phase, dispatcher module is remembering previously loaded />/destinations that come from first ds_select_domain />/and module try to route these destinations in the case that all />/voicemail servers are unavailable. />/So basically, I want to unload all destinations comes from previous />/ds_select_domain. />//>/From the documentation, i tried ds_load_update and ds_load_unset />/methods but not worked for me. />//>/Is there any way to overcome that problem or any right usages of these />/methods. />//>/Thank you, />//>/Serdar />//>//>/__________________________________________________________ />/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 > <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> / > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Access via script to location data in the memory
by Social Boh 31 Jul '21

31 Jul '21

Hello List, is there a way to access location data from kamailio script? I mean data I can see with the command: kamctl ul show Regards -- --- I'm SoCIaL, MayBe

3 11

Guidelines on E164 conversions
by Melissa Frasik 30 Jul '21

30 Jul '21

Hi all, I'm running into some issues with E164 conversions and was wondering if there was a decent standardized method for conversions. Conversions using $rU seem to be working great, but $fU is behaving badly. This is what I have so far: xlog("$ci - caller id was $avp(src=>callerid)"); if($avp(src=>callerid) =~ "^\+"){ xlog("$ci - caller id already E164"); }else if($(rU{s.substr,0,$(avp(src=>countrycode){s.len})}) == $avp(src=>countrycode)){ $avp(src=>callerid) = "+" + $avp(src=>callerid); $fU = $avp(src=>callerid); $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - Adding plus to make caller id e164"); }else{ $fU = "+" + $avp(src=>countrycode) + $fU; $avp(uacreplacefromdisplay) = $fU; $avp(uacreplacefromuri) = $fu; xlog("$ci - converting domestic caller id number to $fU ($fu) - $avp(uacreplacefromdisplay) $avp(uacreplacefromuri)"); } Previously, I'd found a suggestion mentioning using uac_replace_from in the route[RELAY] (which is the purpose of the uacreplacefrom avps), but $fU does not appear to be changing its value. Is there something I'm doing incorrectly?

2 4

How to unload previously loaded destinations in the case of applying multiple ds_select_domain
by Serdar GÜÇLÜER 30 Jul '21

30 Jul '21

Hello all, I have a trouble that is related to the way of usage of dispatcher module. I am working with Kamailio 5.3.2 and using dispatcher module as load balancer to route calls to the media gateways. My dispatching routes are below as simply, route[DISPATCH] { if(ds_select_domain("2", "4")) { route(MYRELAY); } else { send_reply("503","Service Unavailable - No MGW"); exit; } } route[MYRELAY] { t_on_failure("MYFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if ($T_reply_code == 408 || $T_reply_code == 503) { if(ds_next_domain()) { route(MYRELAY); } else { send_reply("503","Service Unavailable"); exit; } } else { if(ds_select_domain("6", "4")) { route(MYVMRELAY); } else { send_reply(486,"Busy"); exit; } } } route[MYVMRELAY] { t_on_failure("MYVMFAILURE"); if(!t_relay()) { sl_reply_error(); } exit; } failure_route[MYVMFAILURE] { route(NATMANAGE); revert_uri(); if (t_is_canceled()) { exit; } if(ds_next_domain()) { route(MYVMRELAY); } else { send_reply("503","Service Unavailable"); exit; } } I have multiple media gateways(setid=2) and voicemail servers(setid=6). As can be seen from the configuration, after first routing, for transaction reply code except 408 or 503, I am routing call to the voicemail server using dispatcher. In that phase, dispatcher module is remembering previously loaded destinations that come from first ds_select_domain and module try to route these destinations in the case that all voicemail servers are unavailable. So basically, I want to unload all destinations comes from previous ds_select_domain. From the documentation, i tried ds_load_update and ds_load_unset methods but not worked for me. Is there any way to overcome that problem or any right usages of these methods. Thank you, Serdar

2 1

Setting up uacreg
by Duncan Turnbull 30 Jul '21

30 Jul '21

Hi there I am a new user of Kamailio and we are trying to use it to be as a front end for our asterisk pbx. We are running on Ubuntu 18.04 and Kamailio 5.3.8 with Siremis Rather than proxying the request through to asterisk we are trying to use uacreg to send a login to asterisk. Asterisk will think all the users are appear from the proxy but thats okay. Initially this is just for external users but eventually all phones etc will register via Kamailio and we will have the trunks there (and split them across another kamailio but thats another job) If I add a user to the uacreg then when I register to Kamailio it sends a register request but to the realm in the uacreg table and the matching port Kamailio is running on. Is this because somewhere we have set Kamailio to directly proxy on and we need to turn that off first? This is our uacreg table mysql> select * from uacreg; +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | id | l_uuid | l_username | l_domain | r_username | r_domain | realm | auth_username | auth_password | auth_ha1 | auth_proxy | expires | flags | reg_delay | socket | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ | 1 | testuser | testuser | ourdomain.com | 88 | 10.8.8.20 | 10.8.8.20 | 88 | password | '' | sip:10.8.8.20:5060 | 360 | 0 | 3 | | +----+--------+------------+------------+------------+-----------+-----------+---------------+---------------+----------+--------------------+---------+-------+-----------+--------+ 1 row in set (0.00 sec) All pointer, guides and recommendations will be welcome Thanks very much Cheers Duncan

3 8

Guidance tracking down "qm_free(): BUG: freeing already freed pointer"
by Anthony Joseph Messina 29 Jul '21

29 Jul '21

I'm seeking guidance on how to track down "qm_free(): BUG: freeing already freed pointer" which occurs only on branched calls. These errors don't crash so I don't get any core dumps. The different log entries below are the result of me selectively disabling sections of the script that apply in branch route, all to no avail. I'm running Kamailio on the current tip of the 5.5 branch (1f9f6fff6e). I'm reviewing https://www.kamailio.org/wiki/tutorials/troubleshooting/memory in the meantime as a place to start. version: kamailio 5.5.1-5.git1f9f6fff6e.fc34 (x86_64/linux) 7abebb 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. CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f47e90), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f04b8f06c70), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f08e8f9cf80), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa60d8), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae2a018), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afae431e8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f2afafa3900), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5fd4044ef0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ffbb2d2e2f0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6d1a50b1d8), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c165c70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c17adb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc19c125790), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4d969c9d48), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc29bdca3d0), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2eccc80), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fc2c2ec52c0), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f910c04e240), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa351286b18), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa3512a7ae8), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d9e70), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa0777d7a80), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92cf20), called from core: core/data_lump.c: free_lump(470), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c8afa00), called from core: core/data_lump.c: free_lump(470), first free textops: textops.c: add_hf_helper(3474) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f4f2c92aa48), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fe55825eb30), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/sdp/sdp.c: init_p_payload_attr(183) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f6526db8378), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fcd9b465980), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7ff7ca92ff60), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2dc938), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f978a2b7eb8), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fa1556c1610), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd81dfbba78), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750d9988), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fb2750e1b28), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/parse_addr_spec.c: free_to_params(895) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f32f7b45d30), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f35849e1940), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7fd3a2c04280), called from core: core/parser/sdp/sdp.c: free_sdp(825), first free core: core/parser/msg_parser.c: reset_ua(994) - ignoring CRITICAL: <core> [core/mem/q_malloc.c:519]: qm_free(): BUG: freeing already freed pointer (0x7f5de857a040), called from core: core/parser/msg_parser.c: reset_instance(916), first free core: core/parser/hf.c: free_hdr_field_lst(217) - ignoring -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

2 6

BYE not accepted by MS TEAMS
by João Vitor Arruda 29 Jul '21

29 Jul '21

Hello all, I recently built an SBC to integrate Kamailio with MS Teams following Henning Westerholt blob post and all the other awesome tips I was able to find here in the list. Everything is working great. The only missing piece I would like some thoughts on is about why MS Teams is not accepting the BYE sent from Kamailio. It eventually replies with "504 Server Time-out" The below trace shows the entire call. |||||||||||||||||||| ==================== tag: rcv pid: 3381 process: 63 time: 1627070235.086307 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ INVITE sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone;transport=tls SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91 RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 1103 MIN-SE: 300 SUPPORTED: timer USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY PRIVACY: id SESSION-EXPIRES: 3600 v=0 o=- 599625 0 IN IP4 127.0.0.1 s=session c=IN IP4 52.113.58.181 b=CT:10000000 t=0 0 m=audio 50122 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118 c=IN IP4 52.113.58.181 a=rtcp:50123 a=ice-ufrag:fTy/ a=ice-pwd:tQKGN0Szi7Q/vTdf6SiXUSpj a=rtcp-mux a=candidate:1 1 UDP 2130706431 52.113.58.181 50122 typ srflx raddr 10.0.33.230 rport 50122 a=candidate:1 2 UDP 2130705918 52.113.58.181 50123 typ srflx raddr 10.0.33.230 rport 50123 a=candidate:2 1 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=candidate:2 2 tcp-act 2121006078 52.113.58.181 49152 typ srflx raddr 10.0.33.230 rport 49152 a=label:main-audio a=mid:1 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:PC00Mgk39GjTYJ+mUSNQg++TU6zJKar4VYDj4/ET|2^31 a=sendrecv a=rtpmap:104 SILK/16000 a=rtpmap:9 G722/8000 a=rtpmap:103 SILK/8000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=rtpmap:118 CN/16000 a=ptime:20 |||||||||||||||||||| ==================== tag: snd pid: 3381 process: 63 time: 1627070235.089106 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 100 trying -- your call is important to us FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone> CSEQ: 1 INVITE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bK60895c91;rport=4672 Server: kamailio (5.3.7 (x86_64/linux)) Content-Length: 0 |||||||||||||||||||| ==================== tag: snd pid: 3291 process: 18 time: 1627070235.111825 date: Fri Jul 23 19:57:15 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 180 Ringing FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Accept:application/sdp Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer l:0 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 |||||||||||||||||||| ==================== tag: snd pid: 3298 process: 21 time: 1627070245.043621 date: Fri Jul 23 19:57:25 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 183 Session Progress FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: snd pid: 3293 process: 19 time: 1627070255.938770 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.76.76 dstport: 4672 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 200 OK FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 t:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e CSEQ: 1 INVITE m:<sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp> User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces u:talk,hold,conference,refer c:application/sdp Content-Disposition:session l:323 Via: SIP/2.0/TLS 52.114.76.76:5061;rport=4672;branch=z9hG4bK60895c91 Record-Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on>,<sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XBranch: z9hG4bKe7d4.5ae277881edb9c04a7ac65af016e582e.0 v=0 o=FreeSWITCH 1627045233 1627045234 IN IP4 SBC_IP_ADDR s=FreeSWITCH c=IN IP4 SBC_IP_ADDR t=0 0 m=audio 49692 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:eMGqj0jubamtvLsI/E/YSj8UqNKgUB5DTHGWc7PY a=nortpproxy:yes |||||||||||||||||||| ==================== tag: rcv pid: 3370 process: 57 time: 1627070255.957735 date: Fri Jul 23 19:57:35 2021 proto: tls ipv4 srcip: 52.114.76.76 srcport: 4672 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ ACK sip:+5515XXXXXXXXX@192.168.1.254:5060;transport=udp SIP/2.0 FROM: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 TO: <sip:+5515XXXXXXXXX@SBC-FQDN:5061>;user=phone;tag=e55N8paBUgZac CSEQ: 1 ACK CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e MAX-FORWARDS: 70 VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKe3747d68 CONTACT: <sip:api-du-b-usea2.pstnhub.microsoft.com:443 ;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c> CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUNO.0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> Route: <sip:192.168.1.253:5060 ;r2=on;ftag=bafa0f5a573749a7b494917c0309f544;lr=on> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: snd pid: 3302 process: 23 time: 1627070275.228363 date: Fri Jul 23 19:57:55 2021 proto: tls ipv4 srcip: SBC_IP_ADDR srcport: 5061 dstip: 52.114.75.24 dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ BYE sip:52.114.76.76:4672;x-i=a82f1d71-4e21-42a8-8389-2ae26e54610f;x-c=c9b2aa4f658f5788aa7090bef0f5e35e/d/8/b14037a741dd46a79c74dce15912782c SIP/2.0 Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on> Record-Route: <sip:192.168.1.253:5060;r2=on;ftag=e55N8paBUgZac;lr=on> Via: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 v:SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m f:<sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac t:"Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 i:c9b2aa4f658f5788aa7090bef0f5e35e CSeq:38981345 BYE User-Agent:FreeSWITCH-mod_sofia/1.10.3-release~64bit Allow:INVITE,ACK,BYE,CANCEL,OPTIONS,MESSAGE,INFO,UPDATE,REGISTER,REFER,NOTIFY k:path,replaces Reason:Q.850;cause=16;text="NORMAL_CLEARING" Max-Forwards: 64 l:0 Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr> P-SR-XUID: atpsh-60fb1b5d-d35-1 |||||||||||||||||||| ==================== tag: rcv pid: 3385 process: 65 time: 1627070300.423499 date: Fri Jul 23 19:58:20 2021 proto: tls ipv4 srcip: 52.114.75.24 srcport: 5061 dstip: SBC_IP_ADDR dstport: 5061 ~~~~~~~~~~~~~~~~~~~~ SIP/2.0 504 Server Time-out FROM: <sip:+5515XXXXXXXXX@SBC-FQDN:5061;user=phone>;tag=e55N8paBUgZac TO: "Last, First Name"<sip:+8979994;ext=01180@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=bafa0f5a573749a7b494917c0309f544 CSEQ: 38981345 BYE CALL-ID: c9b2aa4f658f5788aa7090bef0f5e35e VIA: SIP/2.0/TLS SBC_IP_ADDR:5061;branch=z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0 CONTENT-LENGTH: 0 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SERVER: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.EUWE.3 Via: SIP/2.0/UDP 192.168.1.254;received=192.168.1.254;rport=5060;branch=z9hG4bKgvXrQrcNjNj3m Record-Route: <sip:SBC-FQDN:5061;transport=tls;r2=on;ftag=e55N8paBUgZac;lr=on>,<sip:192.168.1.253:5060 ;r2=on;ftag=e55N8paBUgZac;lr=on> P-SR-XBranch: z9hG4bK92fc.fcb147cf8e5e6e6e960d0f2401ad6383.0

2 2

Accessing Transformations in KEMI
by Ben Kaufman 28 Jul '21

28 Jul '21

Sorry to bother on this, but is it possible to access Kamailio transformations in KEMI, and if so, how? In Kamailio's native scripting language I could do: $var(pai_user) = $(hdr(P-Asserted-Identity){tobody.user}); In KEMI, I can easily get the full header like this: KSR.hdr.get("P-Asserted-Identity") But I don't see how to access the transformations. Am I missing it? Ben Kaufman

3 6

Password for pi_framework.xml - Xhttp_pi module
by Abdirahman Osman 27 Jul '21

27 Jul '21

Hello everyone, In the xhttp_pi file of pi_framework.xml,is there a way I can pass the password to the xhttp_pi module as a pre-processing define? <db_url id="postgres">postgres://kamailiorw:kamailiorw@localhost /kamailiorw</db_url> Thank you! Abdirahman

2 1

Info: Kamailio World Online - Sep 1-2, 2021 - Celebrating 20 Years Of Development
by Daniel-Constantin Mierla 26 Jul '21

26 Jul '21

Hello, given the current situation across the world, no physical Kamailio World Conference is planned in 2021. However, this year the project celebrates 20 years of development -- according to commit history, the first one was done on the 3rd of September, 2021: > ❯ git log --reverse > commit 512dcd9838f5aeb0bfccd02cfc156067b7f02b71 > Author: Andrei Pelinescu-Onciul <andrei(a)iptel.org> > Date: Mon Sep 3 21:27:11 2001 +0000 > Initial revision To mark this notable milestone, we are going to organize an online event during Sep 1-2, 2021, similar to the one from 2020 -- the website for the event is available at: * https://www.kamailioworld.com/k09-online/ Mark the date in your calendar, more details about the content of the event will be published in the near future! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

New user checking on SIP proxying use case
by Nick Hurlburt 22 Jul '21

22 Jul '21

Hello all, My name is Nick Hurlburt, and work with a nonprofit tech organization that provides a software platform for child helplines around the world. I am a new user to Kamailio, and I'm interested in figuring out if my use case is appropriate, getting some tips on how to go about it, and possibly enlisting some expert help. Our service is built on top of Twilio, but we are working in some countries where Twilio doesn't have phone numbers. So in those areas, we are working with local telecoms on ways to integrate using a SIP connection. In one particular example, the telecom company can send SIP INVITEs but can only address them to a static IP. However, Twilio needs to receive the invite addressed to a fully-qualified domain name (eg, example-setup.sip.twilio.com). The idea arose to use Kamailio as a proxy to accept the static IP-addressed request and then rewrite the request to pass it along to a domain name. Then once the connection is established between the two endpoints, the media can flow directly between them. I have a software engineering background but still coming up to speed when it comes to VoIP technology. I'd be interested in: - does this sound like a good use case for Kamailio? - is there any specific documentation you could point me to for this use case? (I have been looking at the Kamailio documentation, but a quick pointer could save a lot of time in digging through it) - if anyone from the community would be open to helping us develop a prototype/proof-of-concept instance of this, especially if you're open to contributing to a nonprofit on a volunteer or discounted rate, I'd love to talk more Thank you, Nick

4 4

Kamailio/rtpengine not populating mos avp stats?
by Anthony Joseph Messina 22 Jul '21

22 Jul '21

I'm using Kamailio compiled from the current 5.5 tip (1f9f6fff6e), and rtpengine compiled from the current master (6e160da4). While rtpengine is working quite well, within the past month or so, I've noticed that Kamailio isn't populating the mos* related AVPs. I see continued upstream rtpengine work stream handling, etc. -- should I wait or try to debug? Thanks -A -- Anthony - https://messinet.com CCED 8BFB 88D9 18F6 9F68 93DC 45C4 A8CE A64E 88E3

3 6

How to get Stir/Shaken certificates
by ryan embgrets 22 Jul '21

22 Jul '21

Greetings. Does someone know where one can get stir/shaken certificates? Are they same as letsencrypt or do I need to obtain them from some other authority. If someone could provide me with a link to that authority it will be very helpful. Thanks again for your help and time. Ryan

4 5

Question about MS Teams integration
by Володимир Іванець 22 Jul '21

22 Jul '21

Hello all! Sorry, this is not a direct Kamailio question. I would like to ask for some help from those who already did this. I'm trying to integrate my testKamailio installation with MS Teams. I added the "mydomain.com" to the Domains section (I kept the " mydomain.onmicrosoft.com"). When I tried to create SBC with a " kamailio-dev2.mydomain.com" FQDN I got this error message: We* can't use the "kamailio-dev2.mydomain.com <http://kamailio-dev2.mydomain.com>" domain as it hasn't be set up in the organization. Try again. If you continue to have problems, contact Microsoft customer support.* There were no issues creating a dummy SBC with a " kamailio-dev2.mydomain.onmicrosoft.com" FQDN. Does anyone know how to set up a domain in the organization from MS perspective? Thank you very much! Regards, Volodymyr Ivanets.

4 9

serial forking kamailio - twilio
by Filippo Graziola 21 Jul '21

21 Jul '21

Hi all, I have a question for my kamailio configuration which works like this: - INVITE arrives to kamailio - HTTP async request to API endpoint - Resume, use of rtjson module for parallel forking and loop to kamailio itself - One of the invite is "transformed" in serial forking with rtjson module and it is sent with a delay to TWILIO SIP Domain (landline/mobile phone numbers) Twilio is giving me some troubles because it answers with a 200 OK but the call it is still in establishing/ringing on the "landline" side and as expected when reaching the timeout (no answer from phone number on TWILIO side) kamailio is receiving a BYE. Now the question, how to handle next number to call through Twilio? I was thinking of using rtjson_next_route() when handling the in-dialog BYE but as it is not a failure route so it does not work, or I'm doing it wrong. Thanks in advance for any help/hint. Filippo

2 2

Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record
by Henning Westerholt 20 Jul '21

20 Jul '21

Hello, I have not tested it, but the dialog database will probably only help in a restart scenario with regarding to dialogs, at least with a standard cfg. Regarding the example below, this would mean starting the primary Kamailio again, then the cdrs should be generated during the hangup from the primary. You can test also to restart the secondary Kamailio, it should then also read the dialogs from the replicated table and generate cdrs at hangup. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Social Boh <social(a)bohboh.info> Sent: Tuesday, July 20, 2021 2:02 PM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello, with dialog db_mode 1 and Primary/Primary MariaDB Replication If I switch Off primary Kamailio during a call and move the call to Replica and then hangup, no entry on acc_cdrs table. acc table is ok. Regards --- I'm SoCIaL, MayBe El 20/07/2021 a las 4:09 a. m., Henning Westerholt escribió: Hello, Usage of DB to store dialogs. See https://kamailio.org/docs/modules/devel/modules/dialog.html#dialog.p.db_mode – in particular mode 1 and 3. 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 Social Boh Sent: Tuesday, July 20, 2021 12:55 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hello what means "DB loading" ? Regards --- I'm SoCIaL, MayBe El 19/07/2021 a las 9:25 a. m., Henning Westerholt escribió: Hello, so, you are saying that in a DMQ replicated case the acc CDR generation is not working in failover cases. I just looked briefly to the code; others might have more details. But I think this is because the acc module do not register to a callback function for dialog loading on startup for the DMQ replication. This is done for DB loading, but seems to be not implemented yet for DMQ synchronisation. You could open a feature request on our github tracker for this. 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 Social Boh Sent: Monday, July 19, 2021 1:58 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org><mailto:sr-users@lists.kamailio.org> Subject: Re: [SR-Users] Kamailio KeepAlived HA acc_cdrs table record Hi, I'm using 5.4.6 I switch off Kamailio on the first server and keepalived move the virtual IP on the second server so First Kamailio is off. This during the call. Before switch off Kamailio and during the call I can see data dialog on both server but in the LOG /var/log/messages in my case: server 1 INVITE server 2 BYE acc_cdrs table empty on both servers. acc table first server INVITE; acc table second server BYE Regards --- I'm SoCIaL, MayBe El 18/07/2021 a las 2:33 p. m., M Arqum CH escribió: Hi SoCIal, which Kamailio version you are using. did you able to get the dialog information using the dialog command. https://www.kamailio.org/docs/modules/devel/modules/dialog.html#dlg.r.dlg_l… when you received bye on secondary server, is that server recognized the bye. On Sun, Jul 18, 2021 at 3:38 AM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, if during a call I stop Primary Kamailio the Secondary Kamailio take de Virtual IP but don't save call data: In the Primary database, acc table, I have INVITE data but not BYE data; no data in acc_cdrs for the call. In the Secondary Kamailio, acc table, I have BYE data but not INVITE data. in the file LOG no data about call cdr. Maybe a solution is write call data whene the call end? Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 11:34 a. m., M Arqum CH escribió: Make sure that your table columns matched the field's you are trying to insert, please check the syslog for acc module log, acc module print the cdr in log facility too, if acc module facing any issue while writing cdr in db, it will throw error in logs. Hope this will help you. On Sat, Jul 17, 2021, 7:33 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Hello, i tested with DMQ module and happens the same. DMQ module loaded before dialog module and on dialog module parameters: modparam("dialog", "enable_dmq", 1) During a call I can see on both Kamailio servers the dialog data but when switch off the active kamailio server and the second kamailio server take the control and hangup the call, no data acc_cdrs table. Regards --- I'm SoCIaL, MayBe El 17/07/2021 a las 1:24 a. m., M Arqum CH escribió: Hi Social, Replication of dialog using dmq will solve your problem, you need to load dmq module before loading dialog module and enable replication of dialogs in dialog module, please check dialog module conf option for more details. Cheers On Fri, Jul 16, 2021, 7:32 PM Social Boh <social(a)bohboh.info<mailto:social@bohboh.info>> wrote: Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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 __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users(a)lists.kamailio.org<mailto:sr-users@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

1 0

Kamailio KeepAlived HA acc_cdrs table record
by Social Boh 20 Jul '21

20 Jul '21

Im' testinf Kamailio HA with Keepalived. I have a problem with acc_cdrs table in this situation: A call B and when the two users are in the call I switch off the active Kamailio. Keepalived pass the Virtual IP to the secondary Kamailio and audio still work because I'm using a remote RTPENGINE server for both Kamailio. The problem is when the call end. On the acc table i can see INVITE and BYE entries but on the acc_cdrs table nothing. Second problem the dialog still present on the table dialog in a Primary/Primary replication. Is there a way to solve it? Thank you Regards -- --- I'm SoCIaL, MayBe

3 11

Kamailio python_exec getting randomly TypeError
by Voip support 19 Jul '21

19 Jul '21

Dear Kamailio users, I am trying to use a python script using the module app_python3 to do some billing calculations. The only argument i want to pass to the python script is duration which i calculate in event_route[dialog:start] { $dlg_var('call_start') = $TS; event_route[dialog:end] { $var('v_duration') = $TS - $dlg_var('call_start'); So my $var('v_duration') has the duration of the call in seconds. Then i execute from my kamailio.cfg: python_exec("calculateCost","$var('v_duration')"); The script works randomly - sometimes all is good and sometimes it throw error : app_python3 [python_support.c:156]: python_handle_exception(): apy_exec: calculateCost(4): Unhandled exception in the Python code: TypeError: an integer is required (got type NoneType) The above exception was the direct cause of the following exception: Traceback (most recent call last): File "/etc/kamailio/kamailio-python.py", line 282, in calculateCost duration = int(duration) It looks like the kamailio log show calculateCost(4) so argument 4 is passed to python function but once it executes it appear like there is no string/integer but NoneType argument. I cannot figure this out why this randomly happens - maybe it's an app_python3 module issue? Looking to hear back from you! BR, Tom

2 1

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

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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

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