- sr-users - kamailio.org

ACK Not Relayed to MS Teams
by me.projects＠yahoo.com 23 Nov '20

23 Nov '20

Hi, I am integrating PBX with Teams via Kamailio Call is working but ACK delivery has an issue This is the ACK Kamailio receives from PBX I process the ACK with handle_ruri_alias(); and record_route_preset(AS per GUIDE LINES) But this change is not reflected in RELAY ACK from PBX ACK sip:api-du-b-usea.pstnhub.microsoft.com:443;x-i=0c344c6a-cd75-4095-9891-bf9445c28b82;x-c=5691aa770226562db7d3e16ab090e196/s/1/e857255ef2344e03bd1d12e270ef5e6c;ias=52.114.132.46~5061~3 SIP/2.0 Via: SIP/2.0/UDP PBX_IP:7790;rport;branch=z9hG4bKPjc50440d0-77db-4467-94cb-2f93c4f88538 From: "USER" <sip:13300@PBX_IP>;tag=ab87b9b5-eee3-420c-8472-2eec82a769ac To: <sip:1508@KAMAILIO_IP>;tag=a2ce378b106949cf8bac8f978bb573fb Call-ID: 3dbc6680-fec3-4ba5-8232-6dda8c669bca CSeq: 23811 ACK Route: <sip:KAMAILIO_IP:7790;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sbc.KAMAILIO_FQDN.com;transport=tls;lr;ftag=ab87b9b5-eee3-420c-8472-2eec82a769ac> Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr> Max-Forwards: 70 User-Agent: Asterisk Content-Length: 0 X-Siptrace-Fromip: udp:PBX_IP:7790 X-Siptrace-Toip: udp:KAMAILIO_IP:7790 X-Siptrace-Time: 1606135546 516717 X-Siptrace-Method: ACK X-Siptrace-Dir: in

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textops.subst_uri does not change all branches
by Anthony Alba 23 Nov '20

23 Nov '20

Hello kamailioers, Version 5.4.2. I have a R-URI that I need to change TLS -> TCP in all branches. I do this in request route before t_relay: // AoR has multiple contacts, all with transport=TLS route[REQUEST] { textops.subst_uri("/transport=TLS/transport=TCP/i") //parallel forking //why doesn't this change all branches? t_relay() } If I print out the R-URI in the branch route, I see that only the first R-URI is changed, in the other branches the R-URI transport=TLS is not changed to transport=TCP. Any suggestions on how to get this to work across all Contact addresses? Thanks Anthony Alba

2 1

Issue with ca-list
by George Goglidze 20 Nov '20

20 Nov '20

Hi Folks, I was wondering if somebody could help me with an issue. I’m a newbie here, just installing Kamailio sip server. I’ve enabled TLS, and am trying create a SIP Trunk to external SIP Service which is TLS enabled port 5061. I’ve configured the following in tls.cfg: [server:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/calist.pem In the section above – ca_list = calist.pem contains all the CA’s and Subordinates of the destination server. Private_key and certificate are of my own server (public godaddy signed) [client:default] method = TLSv1.2+ verify_certificate = yes require_certificate = yes private_key = /etc/kamailio/certs/sbc-private.pem certificate = /etc/kamailio/certs/godaddy.pem ca_list = /etc/kamailio/certs/godaddyca.pem In the section above the ca_list is godaddy’s ca and subordinate. In the wireshark I can see that I’m sending out SIP OPTIONS PING (I’m using dispatcher module). Then the server replies with tls SERVER HELLO which includes it’s certificate But for some reason we are rejecting it: Alert (level: fatal, Description: Unknown CA) How should I set this up to make sure the remote server CA’s are verified? Thank you,

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t_uac_send function with From tag
by João Vitor Arruda 20 Nov '20

20 Nov '20

Hello, I'm trying to use the function t_uac_send inside a failure_route as described in https://kamailio.org/docs/modules/5.3.x/modules/tm.html#tm.f.t_uac_send to send a CANCEL out. Although the documentation says "it can include From/To tags" i was not able to get the from tag provided to be used. Another random generated one is used. I've tried the following format without success: t_uac_send("CANCEL", "$ru", "", "", "From: <$fu>;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: $fu;tag=$ft\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") t_uac_send("CANCEL", "$ru", "", "", "From: bob(a)kamailio.org;tag=2w3e\r\nTo: $tu\r\nCall-ID: $ci\r\n", "") In the last one which is copied from the documentation that is how the >From header going out looks like: From: <bob(a)kamailio.org;tag=2w3e>;tag=3393f0703fb0ccaca74109ff37de39f5-36d71ef0 Any tips on how to get the From tag passed to the function to be used? Thanks, Joao Arruda

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Kamailio suddenly stopes responding to sip messages
by Uttam Giri 19 Nov '20

19 Nov '20

Hi, We are using Kamailio with Jitsi client where we observed sometimes Kamailio stops responding to sip messages. As a result, my Jitsi client goes offline and never back online. Also in the Kamailio syslog there are no error logs(debug enabled) related to this issue. As a temporary workaround, we had to restart the Kamailio server all the time. Does anyone observed a similar issue and can suggest any resolution. Kamailio version:- version: kamailio 5.3.5 (x86_64/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES, TLS_PTHREAD_MUTEX_SHARED ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 7.5.0 UDP process when Kamailio stops responding:-. root@ip-10-68-16-14:/home/ugiri# kamctl ps { "jsonrpc": "2.0", "result": [ { "IDX": 0, "PID": 15436, "DSC": "main process - attendant" }, { "IDX": 1, "PID": 15442, "DSC": "udp receiver child=0 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 2, "PID": 15443, "DSC": "udp receiver child=1 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 3, "PID": 15444, "DSC": "udp receiver child=2 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 4, "PID": 15445, "DSC": "udp receiver child=3 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 5, "PID": 15446, "DSC": "udp receiver child=4 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 6, "PID": 15447, "DSC": "udp receiver child=5 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 7, "PID": 15448, "DSC": "udp receiver child=6 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 8, "PID": 15449, "DSC": "udp receiver child=7 sock=10.68.16.14:5060 (100.26.5.214:5060 )" }, { "IDX": 9, "PID": 15450, "DSC": "slow timer" }, { "IDX": 10, "PID": 15451, "DSC": "timer" }, { "IDX": 11, "PID": 15452, "DSC": "secondary timer" }, { "IDX": 12, "PID": 15453, "DSC": "JSONRPCS FIFO" }, { "IDX": 13, "PID": 15454, "DSC": "JSONRPCS DATAGRAM" }, { "IDX": 14, "PID": 15455, "DSC": "ctl handler" }, { "IDX": 15, "PID": 15456, "DSC": "Http Async Worker" }, { "IDX": 16, "PID": 15457, "DSC": "Http Async Worker" }, { "IDX": 17, "PID": 15458, "DSC": "TIMER NH" }, { "IDX": 18, "PID": 15459, "DSC": "tcp main process" } ], "id": 17709 } Thanks, Uttam

2 3

Re-Invite from callee With RTPEngine between RTP--SRTP
by me.projects＠yahoo.com 19 Nov '20

19 Nov '20

Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP So I m using rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP") 1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP I detect source IPs and to manage this all. When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1. How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee. Please advise.

1 0

Presence of plain text username and password in kamailio.cfg
by Ahmed Marsou 19 Nov '20

19 Nov '20

Hi; I want to remove all plain text usernames an passwords from kamailio.cfg file. Like modparam("auth_db", "db_url", "dbdriver://username:password @dbhost/dbname") or this modparam("sqlops","sqlcon","ca=>dbdriver://username:password @dbhost/dbname") Can you help me with some ideas of how can I handle that? Thank you.

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Add Reason Header to kamailio generated CANCEL
by Alfonso Pinto 18 Nov '20

18 Nov '20

Hi everyone, I've got a specific case: when the inv_fr times out, I need to add a Reason header to the CANCEL generated by kamailio. I've tried to see if I could do it in the onsend_route, but that one is not triggered for the generated CANCEL. I also checked event_route[tm:local-request], but that one isn't triggered either for the generated CANCEL. Is there any way to do it? Or maybe to have any pointer about where to look in the code so I may try to trigger event_route[tm:local-request] for these generated CANCELs? Regards, Alfonso

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record-route and INFO
by David Villasmil 18 Nov '20

18 Nov '20

Hello guys, I have a setup like this: FS(private)-->(private)Kamailio1(Public)-->Kamailio2(public)-->provider. I have mhomed=0, enable_double_rr=1, I'm listening on 2 different ports, 1 private, 1 public (with advertise) and on Kamailio1 the DISPATCH ROUTE I force the socket $fs = "LISTEN_PRIVATE_PROTO:LISTEN_PRIVATE_IF:5080"; To the private ip when the destination IP is a private IP. This all works fine. But When calling OUT (i.e. public) kamailio adds itself only for the private IP: *INVITE sip:12345678@mydmain.com <sip%3A12345678(a)mydmain.com> SIP/2.0Record-Route: <sip:172.31.25.124:5080;lr;did=9da.3a61;nat=yes>Via: SIP/2.0/UDP 172.31.25.124:5080;branch=z9hG4bKf05e.979fd257ac7f288804ef146d12c1e0fb.0Via: SIP/2.0/UDP 172.31.6.1:5080;received=172.31.6.1;rport=5080;branch=z9hG4bKQ3H8mgHaUjaBm* So when INFO messages come from the provider to kamailio2, kamailio 2 tries to send the INFO straight to "172.21.25.124". Should i manually add a record-route when calling OUT with record_route_advertised_address("1.2.3.4:5090"); Or is there a better way of getting this working? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

1 0

Info: URL expander browser extension for Kamailio resources
by Daniel-Constantin Mierla 18 Nov '20

18 Nov '20

Hello, short note to the community forum in case someone finds it useful -- I wrote a tiny browser extension to facilitate navigation to various web resources related to Kamailio project, helping to jump quicker to web pages by typing some keywords on the address bar in the modern browsers. Installation and usage instructions are in the readme of the project, available at: * https://github.com/miconda/KSR-URL-Expander A few other details are in a news article: * https://www.kamailio.org/w/2020/11/browser-url-expander-for-kamailio-resour… Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

2 1

UAC Module
by Javier Valencia 18 Nov '20

18 Nov '20

Hi there! Configuring the UAC module to use "remote registers" in a Kamailio 4.4.3. Everything it's working except for the flags database field to which I always put a value of 0. But when Kamailio starts to launch the REGISTRATION that field does not persist in the DB. In the kamailio.cfg file I have not made any changes, I only use the remote registers DB feature. Is it normal behavior? Do I have any parameters not set or incorrectly set? Regards, JV

2 2

utility function to check if ip address is private?
by Juha Heinanen 18 Nov '20

18 Nov '20

I looked in the wiki for a utility function that would check if given ip address is private, but didn't find one. Does it exist? -- Juha

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how to add URI parameter
by Sergiu Pojoga 18 Nov '20

18 Nov '20

Hi there, Does Kamailio have a function to add URI parameters? I looked around but couldn't find one. I refuse to believe Kamailio doesn't have one. There's a *add_uri_param(param)*, but that's strictly for R-URI. I need it for something else, Contact. Thanks, --Sergiu

2 1

help: can anybody give me a configuration template of SBC?
by 陈理军 18 Nov '20

18 Nov '20

Hi I want to configure Kamailio SIP server to act as a SBC. I had read the article of Kamailio working as SBC to connect MS Team project:   https://skalatan.de/en/blog/kamailio-sbc-teams But I can not find the kamailio.cfg file for this scenario. Can anybody give me a configuration template of SBC? BRs Albert

3 2

Multiple Registrations via WebSocket & Cisco phones
by Melek Oktay 17 Nov '20

17 Nov '20

Hi, I am using FreeSwitches behind the Kamailio proxy server and I am trying to allow multiple registration to my extensions. So, following configuration is sample of my Kamailio modparam("registrar", "xavp_cfg", "reg") ...... $xavp(reg=>max_contacts) = 10; save("location"); .... I saw my phones could register with the same account credentials via several phones such as Cisco, Zoiper, Yealing etc. When the call is forward to this extension, all of them are ringing. Very Nice. But, when I am trying to REGISTER WebRTC supports soft-phones to my system and with the same account credentials, my extensions are not ringing like in the previous scenario. WebRTC uses Websocket (WS) technology and clients register to Kamailio via usrloc module. When the call is forward to this extension, Kamailio try to replicate WebRTC'S INVITE packet to other phones (Cisco, Yealing, zoiper etc) and none of them understand incoming INVITE request because of WebRTC supported protocols (ICE ,a=candidate) , in a brief, phones could not recognize/understand incoming WebRTC request. This is a really tough issue for me, how can I send appropriate INVITEs for each of them.

2 2

The Algorithm “13” - latency optimized dispatching
by Julien Chavanton 14 Nov '20

14 Nov '20

About The Algorithm “13” - latency optimized dispatching, Is now reviewed once and tested, it will most likely be ready to merge soon. I want to share my thoughts on it one more time as it is not too late to get more feedback before we merge. I think it is the best algorithm in most use cases, here is why : It is providing round-robin and fail-over with automatic de-prioritization of slow/unresponsive gateways. You probably asked yourself the following questions in the past : "How do I set the thresholds to put a gateway out of service ?" *ds_probing_threshold*, *ds_inactive_threshold* and timers ... - If your thresholds are too strict, you may end up running out of gateway. - If your thresholds are too tolerant, you may end up adding excessive delays to call establishment and using degraded gateways. The automatic de-prioritization can help to address this concern more efficiently by providing more flexibility. - it can react faster than lets say 2 consecutive timeouts. - it will not disable gateways but simply de-prioritize / reorder them if needed. The only main drawback I can imagine is when you always need to evenly distribute calls using round-robin. It may be needed sometimes but in this case it means you are willing accept to send calls to a degraded gateway or trough degraded network paths. Even if you may select to preset a mixture of round-robin sets, thanks to *ds_select_routes* however it will stay static, needs to be configured precisely, and will not react to degradation automatically. I hope this will help use to protect QoS and lower latency of calls routed by Kamailio. Feel free to let me know what you think Julien

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rport parameter in VIA header in uac remote registration
by Jetse Verschuren 12 Nov '20

12 Nov '20

Hi, I'm currently trying to connect Kamailio, running in docker, to my FritzBox. The FritzBox is the modem provided by my ISP, with a builtin voip server which I can use to dial out (basically a PSTN gateway). In my setup I have a custom router/firewall/NAT between the FritzBox and the server. When I connect LinPhone to the FritzBox, from behind the NAT, it works fine. But when I use the Kamailio uac remote registration, I don't get a response. After some investigation with wireshark, I found that the the SIP response from the FritzBox comes on the wrong port, and gets rejected by the NAT. The request comes from port 1025, but the response is sent to port 5060. I compared the requests from LinPhone and Kamailio, and combined with some research, found it probably was the rport option in the VIA header. >From what I read, it basically signals the uas to use the source port to reply to, instead of the port indicated in the request, which would be necessary to effectively go through a NAT. After reading a lot of documentation (and even parts of the source), I wasn't able to find an option to enable this option. Is there some option I completely missed, or is it not possible? Or am I completely wrong in my "diagnosis"? Thanks in advance, Jetse

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Having trouble Setup Kamailio as a load balancer for Asterisk servers using module dispatcher
by Usama Shaikh 11 Nov '20

11 Nov '20

Hello Everyone. I am having trouble to setting up kamailio for load balancing for asterisk servers. I am using module dispatcher for load balancing AND two different servers for kamailio and asterisk. I am doing it first time and have not exact idea about its all configs… but i have researched and done so far, now need help from you guys. Currently, trying to setup with the one kamailio and one asterisk then will go ahead if I get any success… using MariaDB MySQL for the dispatcher data. Simply I want to do is : 1. Register my asterisk devices with kamailioIP (and I am able to do this using below configs) Devices are getting register with the asterisk from with kamailioIP 2. I want to receive calls on the registered devices with KamailioIP (and i am not able to do this) WORK FLOW: KAMAILIO IP = XX.XX.XX.95 on port 5060 and Asterisk IP = XX.XX.XX.164 on port 5060 SIP Device(ZOIPER) -------->>>>KAMAILIO SERVER(device register/load balancer)--------->>> ASTERISK SERVER Please have a look into my configuration files and let me know if anything is missing or wrong in these. Or anything to add or remove from the confs… my kamailio.cfg #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_ASTERISK #define WITH_NAT #!define WITH_DEBUG #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)lists.sip-router.org> # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!ifdef WITH_ASTERISK #!define DBASTURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL7 log_name="kamailio" children=4 /* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ #auto_aliases=no /* add local domain aliases */ #alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:0.0.0.0:5060 #listen=udp: KamailioIP:5060 /* port to listen to * - can be specified more than once if needed to listen on many ports */ port=5060 #!ifdef WITH_TLS enable_tls=yes #!endif # life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605 ####### Custom Parameters ######### # These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id # #!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif #!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif #!ifdef WITH_ASTERISK asterisk.bindip = " ASTERISKIP.164" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = " KamailioIP .95" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif ####### Modules Section ######## # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!else mpath="/usr/local/kamailio-devel/lib64/kamailio/modules/" #!endif #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "sctp.so" loadmodule "dialog.so" loadmodule "siptrace.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif #!ifdef WITH_ASTERISK loadmodule "uac.so" loadmodule "dispatcher.so" loadmodule "jsonrpcs.so" loadmodule "htable.so" # ---------------------- mod dispatcher params ------------------------------ modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_ping_from", "sip:dispathcer@ KamailioIP ") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "ds_ping_method", "INFO") #modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 1) #modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=403;code=404;code=484;code=488;code=481;class=3") modparam("dispatcher", "ds_ping_reply_codes", "class=2;class=3;class=4") #modparam("dispatcher", "priority_col", "dstpriority") #modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") # do failover modparam("dispatcher", "flags", 2) #modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") #modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") #modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") #!endif # ----------------- setting module-specific parameters --------------- modparam("siptrace", "db_url", DBURL) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") #!ifdef WITH_JSONRPC modparam("jsonrpcs", "transport", 7) #!endif # ---------------------- mod uac params ------------------------------ modparam("uac", "reg_contact_addr", "ASTERISKIP:5060") modparam("uac", "reg_db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("uac", "reg_db_table", "uacreg") modparam("uac", "reg_timer_interval", 60) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) modparam("uac","auth_realm_avp","$avp(s:auth_realm_avp)") modparam("uac","auth_username_avp","$avp(s:auth_username_avp)") modparam("uac","auth_password_avp","$avp(s:auth_password_avp)") # ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "ruri_matching", 0) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "db_insert_update", 1) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "secret") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- # ip ban htable with autoexpire after 5 minutes #modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") modparam("htable", "htable", "stats=>size=6;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- #modparam("debugger", "cfgtrace", 1) #modparam("debugger", "breakpoint", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { route(DISPATCH); # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) if(is_method("OPTIONS")) { sl_send_reply("200","Keepalive"); exit; } if(is_method("NOTIFY")) { sl_send_reply("200","OK"); exit; } # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); } exit; } } # Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging ## setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } } # USER location service route[LOCATION] { #!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { # if new call from out there - send to Asterisk # - non-INVITE request are routed directly by Kamailio # - traffic from Asterisk is routed also directy by Kamailio append_hf("FromIp: $src_ip\r\n", "Call-ID"); route(TOASTERISK); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # Authentication route route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_ASTERISK # do not auth traffic from Asterisk - trusted! if(route(FROMASTERISK)) return; #!endif #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } if (is_method("REGISTER") || from_uri==myself){ if (!allow_trusted()) { xlog('wokred here for ipauth check'); sl_send_reply("403", "Forbidden"); } } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sip_users", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; } # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; rtpproxy_manage(); if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if(t_check_status("401")) { uac_auth(); } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } t_relay(); #!endif } # Test if coming from Asterisk route[FROMASTERISK] { # if($si==$sel(cfg_get.asterisk.bindip) # && $sp==$sel(cfg_get.asterisk.bindport)) # return 1; # return -1; # dispatch: is this asterisk ip ? if (ds_is_from_list("1")) return 1; return -1; } # Send to Asterisk route[TOASTERISK] { # $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" # + $sel(cfg_get.asterisk.bindport); # append_hf("X-Asterisk-Context: dexter-phones\r\n"); # route(RELAY); # exit; # select asterisk port append_hf("X-Asterisk-Context: dexter-phones\r\n"); route(DISPATCH); } route[DISPATCH] { # round robin dispatching on gateways group '1' #xlog("<<<<<<<<<<<<<<<<trying for dispatch>>>>>>>>>>>>>>>>>>>"); #if (is_method("INVITE")) { # dst_select( "GROUP", "HASH METHOD") #ds_select_dst("1","4"); #sl_send_reply("100","Trying"); # forward(); # exit(); #} if (!ds_select_domain("1", "4")) { send_reply("404", "No destination"); exit; } t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Forward REGISTER to Asterisk route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } “kamctl dispatcher dump” looks like this: { "jsonrpc": "2.0", "result": { "NRSETS": 1, "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": "sip:ASTERISKIP:5060", "FLAGS": "AP", "PRIORITY": 1, "ATTRS": { "BODY": "rweight=50;weight=50;cc=1", "DUID": "", "MAXLOAD": 0, "WEIGHT": 50, "RWEIGHT": 50, "SOCKET": "", "SOCKNAME": "", "OBPROXY": "" } } }] } }] }, "id": 13403 } And also have set trunk in ASTERISK sip.conf register => kamailioSIP:XXXX@KamailioIP:5060 [kamailio] type=friend host= KamailioIP port=5060 transport=udp allow=all allow=gsm allow=alaw,ulaw insecure=invite,port Permit = 0.0.0.0/0.0.0.0 sipdebug=yes username=kamailioSIP canreinvite=no secret=XXXX qualify=yes dtmfmode=auto when i Register my asterisk devices with kamailioIP (and I am able to do this) Devices are getting register with the asterisk from with kamailioIP asterisk -r : (here 8101 and 8100 is registered with Kamailio IP ) [Nov 9 10:36:02] -- Registered SIP '8101' at XX.XX.XX.95:5060 [Nov 9 10:36:02] -- Unregistered SIP '8101' [Nov 9 10:36:03] -- Registered SIP '8101' at XX.XX.XX..95:5060 [Nov 9 10:36:03] == Manager 'listencron' logged on from 127.0.0.1 [Nov 9 10:36:04] == Manager 'sendcron' logged off from 127.0.0.1 [Nov 9 10:36:07] -- Unregistered SIP '8100' [Nov 9 10:36:09] -- Registered SIP '8100' at XX.XX.XX..95:5060 [Nov 9 10:36:09] > Saved useragent "Z 3.15.40006 rv2.8.20" for peer 8100 [Nov 9 10:36:09] NOTICE[8453]: chan_sip.c:23869 handle_response_peerpoke: Peer '8100' is now Reachable. (51ms / 2000ms) when i dial 8100 to 8101 i get in my asteris -r NOTICE[8453][C-0000000b]: chan_sip.c:26002 handle_request_invite: Call from '8101' (95.217.223.95:5060) to extension '8100' rejected because extension not found in context 'none-dial'. and not able to call extension to extension, PLEASE HELP to do this, I have search a lot and tried so much of stuff in configs but no success still kindly guide me if anything I am missing to do in the process or anything I am doing wrong. I will be eagerly waiting for any help or pointer… Thanks in advance

3 3

Re: [SR-Users] Checking user registration
by ahmed moghazy 11 Nov '20

11 Nov '20

When i used lockup location it is done but if the user registered one time on server the location will be saved on kamailio so i want a method to check registered users live every second for example and i want to delete locations of un registered users ? Thank u. > On 10/11/2020, at 12:41 PM, ahmed moghazy <ahmed.m.moghazi(a)gmail.com> wrote: >

2 2

Checking callee registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want to check if callee is registered or not before routing call because this made error and make caller unregistered untill i restarted the linphone

3 3

Checking user registration
by ahmed moghazy 10 Nov '20

10 Nov '20

I want kamailio to reject call and send message user is not registered when a caller is trying to call unregistered user (callee) , Thank you.

2 1

Expire timeout for notify message answer
by Victor Litvinov 10 Nov '20

10 Nov '20

Hello dear community. We use a presence module with the enabled option "timeout_rm_subs". And we face issues that subscriptions remove too early. We expect that it will be removed after the fr_timer, but it is removed after 1 second. We I can see in log (i'm add additional logs): when we send NOTIFY: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309038 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:536]: retr_buf_handler(): new interval 4000 ms / 64 ticks (max 4000 ms) when timer expire: tm [timer.c:491]: retr_buf_handler(): timer retr_buf_handler @1436309054 (timer_ln 0x7f4708f45bd0 -> rbuf 0x7f4708f45bb0 -> [0x7f4708f458a0 <-> 11592:145228 527]) tm [timer.c:492]: retr_buf_handler(): fr_expire 1436309054 T1 4000 T2 4000 fr_timeout 80 tm [timer.c:367]: final_response_handler(): transaction [0x7f4708f458a0 <-> 11592:145228527] scheduled for deletion tm [timer.c:424]: final_response_handler(): stop retr. and send CANCEL (0x7f4708f458a0) and then subscription remove. As we can see the fr_expire variable was set to ticks+16 ( 1 second). And another strange thing that T1 and T2 timers are the same. So the question is if this behaviour is correct, how can we configure the presence module fr_timer as an expiration value for the answer to notify? And can we use it without retransmissions?

2 1

How to modify INVITE Request Line string
by Jose Figueroa 09 Nov '20

09 Nov '20

Hello everybody, Today I received a request from a carrier which needs us to send the calls using a prefix in the INVITE header. I am reading the documentation for kamailio 4.3.x but I havent found anything I can use in order to modify the Request Line (INVITE) string. I need to change from this: INVITE sip:123456789@IP:5060 to this: INVITE sip:PREFIX123456789@IP:5060 If you can guide me to the right direction would be more than great. Thanks -- Jose Figueroa Senior Software Engineer

2 2

(no subject)
by ahmed moghazy 09 Nov '20

09 Nov '20

Hello, When i make a call from specific user to another it didnt make call and it gave missed call to another one after a minute so the call doesnot complete i dont know why , Thank you.

2 1

Kamailio ims stable version
by Pavithra Mohanraja 08 Nov '20

08 Nov '20

Hi, How to do git clone of kamailio source code with minor version. I could only do git checkout of 5.3 version and it is automatically taking latest stable version. I don't need in this way. I need to do git clone of exact version of kamailio source code . Example - if I need 5.3.6 version.. exact version source code has to be git checked out from kamailio website. How to do that .. kindly let me know.. Thanks, Pavithra

2 3

Re: [SR-Users] Proxy configuration that exit dialog afer INVITE -> OK
by Yufei Tao 07 Nov '20

07 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. I've put a few tips and some resources here about starting with Kamailio, which hopefully may help you a bit: http://tao-communications-ltd.simplesite.com/447576389 Cheers, Yufei >* Message: 7 *>* Date: Thu, 5 Nov 2020 12:26:16 +0200 *>* From: "Adrian Tabacioiu" <Adi.Tabacioiu at c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* To: sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>* Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE *>* -> OK *>* Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel at mail.c-s.ro <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>> *>* Content-Type: text/plain;charset=iso-8859-1 *>>>* Hello all, *>>* Am I new in the usage of kamailio, sorry if I ask somehow something simple *>* or terribly wrong . *>>* I need to test a configuration of Kamailio which exit the dialog between *>* UA after the first OK (that means it will not receive the following ACK, *>* BYE). *>>* I understand that in theory this is possible in two configuration types: *>* - no Record Route header *>* - or stateless proxy *>>* First question: *>* Am I correct with the choices above? *>* Can I find example configuration for this two modes ? *>>* Question 2: *>* two or more proxy should be inter-connected, removing the Record Route *>* header would interfere with proxy routing ? (should not the "Via" *>* information be sufficient?) *>>* Thank you in advance for help. *>>* Best regards, *>* Adrian Tabacioiu *>

2 1

Kamailio dump configuration and exit?
by Brandon Armstead 07 Nov '20

07 Nov '20

Is there a similar hook to make kamailio dump its current configuration it is reading including ... included files, i.e. similar to nginx -T root@main:/home/brandon# nginx -h nginx version: nginx/1.14.2 Usage: nginx [-?hvVtTq] [-s signal] [-c filename] [-p prefix] [-g directives] Options: -?,-h : this help -v : show version and exit -V : show version and configure options then exit -t : test configuration and exit -T : test configuration, dump it and exit -q : suppress non-error messages during configuration testing -s signal : send signal to a master process: stop, quit, reopen, reload -p prefix : set prefix path (default: /usr/share/nginx/) -c filename : set configuration file (default: /etc/nginx/nginx.conf) -g directives : set global directives out of configuration file Thanks for your time in advance!

1 0

send_reply() related feature request
by Juha Heinanen 06 Nov '20

06 Nov '20

Many times (especially when there is a hacking attempt) I want to close TCP or TLS session after sending reply. So, for example, I write: set_reply_close(); send_reply("403", "Forbidden"); It would be more convenient if these two calls could be combined into one, for example: send_reply_and_close("403", "Forbidden"); -- Juha

2 7

Re: [SR-Users] sr-users Digest, Vol 186, Issue 5
by Yufei Tao 05 Nov '20

05 Nov '20

Hi, Here's an example kamailio.cfg that uses record_route so in-dialog requests will go through the Kamailio proxy: https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg There are many other example cfg files: https://github.com/kamailio/kamailio/tree/master/misc/examples Any server that wants to see the in-dialog messages needs to add itself to the record route list when receiving the initial INVITE for example. If a server is not in the list of record routes, then in-dialog requests will not be sent to the server. Headers that are most important in terms of routing are R-URI, Route, Record-Route, Contact, Via. Via header is for routing responses, while Record-Route is for in-dialog requests. Cheers, Yufei > Message: 7 > Date: Thu, 5 Nov 2020 12:26:16 +0200 > From: "Adrian Tabacioiu" <Adi.Tabacioiu(a)c-s.ro> > To: sr-users(a)lists.kamailio.org > Subject: [SR-Users] Proxy configuration that exit dialog afer INVITE > -> OK > Message-ID: <7449102a9f3dbe38b6dba753d1cffcd8.squirrel(a)mail.c-s.ro> > Content-Type: text/plain;charset=iso-8859-1 > > > Hello all, > > Am I new in the usage of kamailio, sorry if I ask somehow something simple > or terribly wrong . > > I need to test a configuration of Kamailio which exit the dialog between > UA after the first OK (that means it will not receive the following ACK, > BYE). > > I understand that in theory this is possible in two configuration types: > - no Record Route header > - or stateless proxy > > First question: > Am I correct with the choices above? > Can I find example configuration for this two modes ? > > Question 2: > two or more proxy should be inter-connected, removing the Record Route > header would interfere with proxy routing ? (should not the "Via" > information be sufficient?) > > Thank you in advance for help. > > Best regards, > Adrian Tabacioiu > >

1 0

How to Add Customer Table During Installation
by me.projects＠yahoo.com 05 Nov '20

05 Nov '20

Hi, I want add some custom Table Creation during installation of Kamailio. I want to add my Custom Table during the execution of kamdbctl create command. Currently I am adding my script in share/kamailio/mysql/auth_db-create.sql Thanks,

2 3

tsilo - failure to add branches (-1)
by Jurijs Ivolga 05 Nov '20

05 Nov '20

Hi, I have several Kamailio proxies - loadbalancer which is used as TLS offload and Authorization server and behind registrar servers. In this case I can't make tsilo work on the registrar server. I'm always getting: "tm [t_append_branches.c:172]: t_append_branches(): failure to add branches (-1)" If I use just one Kamailio which is used as Authorization and Registrar server then tsilo works as expected. Looks like kamailio where I run ts_append tries to connect to UAC directly and not through Loadbalancer. Any ideas? How can I troubleshoot this? Thank you! Jurijs

3 9

Proxy configuration that exit dialog afer INVITE -> OK
by Adrian Tabacioiu 05 Nov '20

05 Nov '20

Hello all, Am I new in the usage of kamailio, sorry if I ask somehow something simple or terribly wrong . I need to test a configuration of Kamailio which exit the dialog between UA after the first OK (that means it will not receive the following ACK, BYE). I understand that in theory this is possible in two configuration types: - no Record Route header - or stateless proxy First question: Am I correct with the choices above? Can I find example configuration for this two modes ? Question 2: two or more proxy should be inter-connected, removing the Record Route header would interfere with proxy routing ? (should not the "Via" information be sufficient?) Thank you in advance for help. Best regards, Adrian Tabacioiu

1 0

Kamailio in VMware with Fault Tolerance on
by Patrick Wakano 03 Nov '20

03 Nov '20

Hello list, Hope you are all good! Recently, the issue https://github.com/kamailio/kamailio/issues/2535 has been investigated and the utilization of the feature vsphere Fault Tolerance is linked as a source of network latency and probably CPU allocation latency. This increases the chances of the mentioned issue to happen. So I would just like to ask if anyone out there is using Kamilio in a VMware environment with Fault Tolerance on? How is the experience? Kind regards, Patrick Wakano

3 6

Re: [SR-Users] htable + event_routes + mod-init + htable:expired
by Brandon Armstead 03 Nov '20

03 Nov '20

Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} Nov 2 21:13:28 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351608 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjA4LCJleHAiOjE2MDQzNTUyMDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351608,"exp":1604355208,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"sUQImDqDfE-xdA1SQ-7I5TnFMZaPh8WB2c-S-ut1hog"}}} type: str } } } Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjI4LCJleHAiOjE2MDQzNTUyMjgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351628,"exp":1604355228,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"stW5JzPR-Gz3oVvQEb1wxF7I0GcZ1KjkkQhinHGEivQ"}}} Nov 2 21:13:48 main kamailio[1334]: INFO: <script>: AUTH_TOKEN_TIME, 1604351628 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 0 size: 1 slot: { { name: auth::time value: 1604351628 type: int } } } Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_EXPIRED, lets retrieve a new one Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_RECEIVED, {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} Nov 2 21:14:08 main kamailio[1331]: INFO: <script>: AUTH_TOKEN_TIME, 1604351648 root@main:/etc/kamailio# kamcmd htable.dump server { entry: 11 size: 1 slot: { { name: auth::token value: {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzUxNjQ4LCJleHAiOjE2MDQzNTUyNDgsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604351648,"exp":1604355248,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" brandon(a)cryy.com "},"source":{"address":"34.94.123.121"}},"signature":"P52y8GIkL2FEeT51WbYicjIdjdWp-1FXpIRPLHFmlw4"}}} type: str } } } On Mon, Nov 2, 2020 at 12:44 PM Brandon Armstead <brandon(a)cryy.com> wrote: > Also to add one more thing, I added a third static variable into the mix > to test when using the "bad config" just setting server=>auth::name to > "brandon" and introducing the 3rd variable seems to just make kamailio hang > when doing kamcmd htable.dump server <---- (no results returned). > > On Mon, Nov 2, 2020 at 12:41 PM Brandon Armstead <brandon(a)cryy.com> wrote: > >> Daniel, >> >> In short kamcmd does not reflect the value set in $sht(server=>auth >> ::token) >> >> It alternates between being set and not being set (reflecting in kamcmd >> htable.dump >> >> root@main:/etc/kamailio# kamcmd htable.dump server >> >> { >> >> entry: 11 >> >> size: 1 >> >> slot: { >> >> { >> >> name: auth::token >> >> value: >> {"data":{"token":{"token":"eyJhbGciOiJIUzI1NiIsInR5cCI6IkpXVCJ9.eyJpc3MiOiJDUllZIiwiaWF0IjoxNjA0MzQ2NzgzLCJleHAiOjE2MDQzNTAzODMsIm1ldGhvZCI6InBvc3Q6YXV0aDp0b2tlbiIsImFjY291bnQiOnsiaWQiOiIxIiwiZmlyc3RuYW1lIjoiVGVzdCIsImxhc3RuYW1lIjoiQXJtc3RlYWQiLCJlbWFpbCI6ImJyYW5kb25AY3J5eS5jb20ifSwic291cmNlIjp7ImFkZHJlc3MiOiIzNC45NC4xMjMuMTIxIn19.aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ","header":{"alg":"HS256","typ":"JWT"},"payload":{"iss":"CRYY","iat":1604346783,"exp":1604350383,"method":"post:auth:token","account":{"id":"1","firstname":"Test","lastname":"Armstead","email":" >> brandon(a)cryy.com >> "},"source":{"address":"34.94.123.121"}},"signature":"aFGWqQg31oNnw10XfZ61Xm5Jncvt7RoNri2ZuDJ6yLQ"}}} >> >> type: str >> >> } >> >> } >> >> } >> >> >> In the kamailio logs it always shows AUTH_TOKEN_RECEIVED (correctly, with >> the correct auth token), kamcmd dump of htable alternates between execution >> of this route: htable:expired:server >> >> Exec 1: >> >> auth::time => [timestamp] >> >> Exec 2: >> >> auth::token => token >> >> Exec 3: >> >> auth::time => [timestamp] >> >> So on and so forth, [ kamcmd htable.dump server ] it does not correctly >> hold both $sht(server=>auth::time) and $sht(server=>auth::token) like it >> should, despite kamailio logs showing the HTTP_AUTH_TOKEN received message >> with the correct auth token (100% of the time). >> >> It's almost like there is a race condition when using the modules >> http_client_query in both init and expired event route. >> >> When I changed it to the resolved code block above separating the calls >> and introducing http_async_client inside htable:expired:server >> >> The result is 100% of the time auth token exists correctly, just like the >> logs 100% of the time show the auth token received regardless of >> event_route implementation. >> >> So somewhere somehow the problematic config causes the hash table to not >> store HTTP_AUTH_TOKEN it receives and logs. >> >

2 1

http_async_client
by Brandon Armstead 03 Nov '20

03 Nov '20

kamailio 5.5.0-dev3 I'm unsure if this is expected behavior, so thought I would ask... When using http_async_query("URI", "ROUTENAME") request_route { route(AUTH); # this does not get executed below route(AUTH) xinfo("[$ci][$rm] we hit this line"); } route[AUTH] { t_newtran(); http_async_query("URI", "ROUTENAME"); } route[ROUTENAME] { # do stuff here to check authentication and return if(auth + registration) { append_hf("Path...."); msg_apply_changes(); # this fails and errors out about incorrect route # ^ invalid usage - not in request route or a reply save("location"); } if(auth + invite) { # route to pstn } } Expected behavior: to be able to obtain some processing / information gathering about authentication with the call and additional information possibly and return to normal routing. As-is now I've to move the actions all into ROUTENAME for kamailio to continue processing, it does not go back to request_route { }. # this does not get executed below route(AUTH) Is this expected behavior, or am I missing something? If I do catch route(AUTH) with a hash table and prevent its execution after first run, the avp's from the transaction created are not available in request_route that were set in ROUTENAME.

2 4

htable + event_routes + mod-init + htable:expired
by Brandon Armstead 02 Nov '20

02 Nov '20

Running 5.5.0-dev3 modparam("htable", "htable", "ipban=>size=8;autoexpire=300;dmqreplicate=1;") modparam("htable", "enable_dmq", 1) modparam("htable", "dmq_init_sync", 1) modparam("htable", "timer_procs", 4) modparam("htable", "timer_interval", 5) modparam("htable", "db_expires", 1) modparam("htable", "htable", "account=>size=4;dmqreplicate=1;") modparam("htable", "htable", "server=>size=4;autoexpire=15;") I'm attempting to retrieve an auth token in the init process and also refresh the token upon expiration in the htable:expired event. I'm seeing some behavior where every other execution of htable:expired the variable $sht(server=>auth::token) is set per kamcmd htable.dump server, however xinfo() reports that the token was retrieved in the variable $sht (server=>auth::token). Please see remaining event route config below: event_route[htable:mod-init] { # generate auth token into http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); } event_route[htable:expired:server] { # process expired htable, renew auth token xinfo("AUTH_TOKEN_EXPIRED, lets retrieve a new one"); http_client_query("https://www.cryy.com/api/auth/token", '{"email": " brandon(a)cryy.com", "password":"XXXX"}', "$var(result)"); sht_lock("server=>auth::token"); $sht(server=>auth::time) = $TS; $sht(server=>auth::token) = $var(result); sht_unlock("server=>auth::token"); xinfo("AUTH_TOKEN_RECEIVED, $sht(server=>auth::token)"); xinfo("AUTH_TOKEN_TIME, $sht(server=>auth::time)"); } I've tried both with locking and unlocking. Also one last thing worth mentioning is that on the alternation where kamcmd htable.dump server shows no auth token, $sht(server=>auth::time) is available, when the auth token is visible in kamcmd htable.dump server there is no sht(server=>auth::time) returned. Also just to be explicitly clear -- xinfo() always returns AUTH_TOKEN_RECEIVED correctly in both event routes. Perhaps I'm over looking something -- thank you for the help in advance. - Brandon

2 3

outgoing TCP connection ID
by Sergiu Pojoga 02 Nov '20

02 Nov '20

Hi there, Is there a PV that exposes the outgoing TCP connection ID? $conid seems to be for received only. *$conid - The TCP connection ID of the connection the current message arrived on for TCP, TLS, WS, and WSS. Set to $null for SCTP and UDP.* The objective is to check the status of the egress TCP connection like for e.g.: if(!tcp_conid_state(conid)) { Thanks. --Sergiu

2 4

Kamailio v5.3.7 Released
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, Kamailio SIP Server v5.3.7 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.6. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.7. Note that 5.3 is the second last stable branch, still officially maintained by Kamailio development team. The latest stable branch is 5.4, with v5.4.2 being release out of it. For more details about version 5.3.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/11/kamailio-v5-3-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.3.7
by Daniel-Constantin Mierla 02 Nov '20

02 Nov '20

Hello, I am planning to release Kamailio v5.3.7 sometime next week, likely on Monday or Tuesday, this being the usual prior notice for heads up to identify issues not reported yet on bug tracker as well as the commits you are aware of that were not backported yet. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 1

Missing refresher parameter in the Session-Expires header and Min-SE header
by Jose Figueroa 01 Nov '20

01 Nov '20

Hello everybody, I am trying an INVITE with a SBCs but fails, finding out it seems the Session-Expires doesn't include the param refresher= and according to this closed issue is already included https://github.com/kamailio/kamailio/issues/288 but I cannot achive this, I read the sst module documentation but I don't see anything related this parameter. So, in summary the SBC expects the following: Session-Expires: 1800; refresher=uac; Min-SE: 600 I set only: modparam("sst", "enable_stats", 1); modparam("dialog", "dlg_flag", 5) modparam("sst", "sst_flag", 6); modparam("sst", "min_se". 1800); Now on the invite I the setflag(); 5 and 6 so I see now the header Session-Expires: 1800 but not the refresher=uac; and also I don't see how to set the Min-SE:600 Could you please point me to the right direction? Any hint would be very helpful. -- Jose Figueroa Software Engineer

1 0

Confusion about TCP worker ports
by Noah Mehl 30 Oct '20

30 Oct '20

Hey all, I’m a little stuck on an implementation of a set of dispatchers via TCP. There are some oddities about the behavior of the TCP source port of the Kamailio tcp worker/s, and maybe this is expected, but it doesn’t seem valid. For instance, I have a dispatcher: "RECORDS": [{ "SET": { "ID": 1, "TARGETS": [{ "DEST": { "URI": “sip:2.2.2.2:5060;transport=tcp", "FLAGS": "AP", "PRIORITY": 5 } }] } }] But when Kamailio sends an OPTIONS keep alive, the source port for the worker is 33940, and not 5060 (which is the TCP listen port), as received by Freeswitch: recv 447 bytes from tcp/[1.1.1.1]:33940 at 18:58:24.958720: ------------------------------------------------------------------------ OPTIONS sip:2.2.2.2:5060;transport=tcp SIP/2.0 Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bK1525.80a9e442000000000000000000000000.0 To: <sip:2.2.2.2:5060;transport=tcp> From: <sip:inbound-kamailio-01>;tag=3c52ba62ee4c4621b9660728159919d3-cda8066f CSeq: 10 OPTIONS Call-ID: 3aa18693487268dc-2790(a)1.1.1.1 Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (5.4.2 (x86_64/linux)) ------------------------------------------------------------------------ Also, I get weird debug messages when this tcp worker is spun up (specifically about Resource temporarily unavailable): 11(2790) DEBUG: dispatcher [dispatch.c:3340]: ds_ping_result_helper(): probe all, mode DS_PROBE_ALL 11(2790) DEBUG: dispatcher [dispatch.c:3383]: ds_ping_set(): probing set #1, URI sip:2.2.2.2:5060;transport=tcp 11(2790) DEBUG: dispatcher [dispatch.c:3414]: ds_ping_set(): Default ping_from: sip:inbound-kamailio-01 11(2790) DEBUG: dispatcher [dispatch.c:3424]: ds_ping_set(): Default outbound proxy: 11(2790) DEBUG: tm [uac.c:450]: t_uac_prepare(): next_hop=<sip:2.2.2.2:5060;transport=tcp> 11(2790) DEBUG: tm [uac.c:158]: dlg2hash(): hashid 21073 11(2790) DEBUG: <core> [core/tcp_main.c:1993]: tcp_send(): no open tcp connection found, opening new one 11(2790) DEBUG: <core> [core/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 2.2.2.2 11(2790) DEBUG: <core> [core/tcp_main.c:1175]: tcpconn_new(): on port 5060, type 2, socket -1 11(2790) DEBUG: <core> [core/tcp_main.c:1494]: tcpconn_add(): hashes: 2712:0:0, 1 11(2790) DEBUG: <core> [core/tcp_main.c:2886]: tcpconn_1st_send(): pending write on new connection 0x7f24e64c1e18 sock 8 (-1/447 bytes written) (err: 11 - Resource temporarily unavailable) 11(2790) DEBUG: tm [uac.c:678]: send_prepared_request_impl(): uac: 0x7f24e65285a8 branch: 0 to 2.2.2.2:5060 11(2790) DEBUG: <core> [core/onsend.c:50]: run_onsend(): required parameters are not available - ignoring 27(2806) DEBUG: <core> [core/tcp_main.c:3792]: handle_ser_child(): read response= 7f24e64c1e18, 5, fd 46 from 11 (2790) 27(2806) DEBUG: <core> [core/io_wait.h:375]: io_watch_add(): DBG: io_watch_add(0x56490f0f8060, 46, 2, 0x7f24e64c1e18), fd_no=37 27(2806) DEBUG: <core> [core/io_wait.h:782]: io_watch_chg(): DBG: io_watch_chg (0x56490f0f8060, 46, 0x1, 0xffffffff) fd_no=38 called 27(2806) DEBUG: <core> [core/io_wait.h:600]: io_watch_del(): DBG: io_watch_del (0x56490f0f8060, 46, -1, 0x0) fd_no=38 called 27(2806) DEBUG: <core> [core/tcp_main.c:4457]: handle_tcpconn_ev(): sending to child, events 1 27(2806) DEBUG: <core> [core/tcp_main.c:4127]: send2child(): selected tcp worker idx:0 proc:19 pid:2798 for activity on [tcp:1.1.1.1:5060], 0x7f24e64c1e18 19(2798) DEBUG: <core> [core/tcp_read.c:1749]: handle_io(): received n=8 con=0x7f24e64c1e18, fd=8 19(2798) DEBUG: <core> [core/tcp_read.c:1548]: tcp_read_req(): content-length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:620]: parse_msg(): SIP Reply (status): 19(2798) DEBUG: <core> [core/parser/msg_parser.c:622]: parse_msg(): version: <SIP/2.0> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:624]: parse_msg(): status: <200> 19(2798) DEBUG: <core> [core/parser/msg_parser.c:626]: parse_msg(): reason: <OK> 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1525.80a9e442000000000000000000000000.0>; state=6 19(2798) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <33940>; state=16 19(2798) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 19(2798) DEBUG: <core> [core/parser/msg_parser.c:498]: parse_headers(): Via found, flags=2 19(2798) DEBUG: <core> [core/parser/msg_parser.c:500]: parse_headers(): this is the first via 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1mB9HryQ8ZBFc 19(2798) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 19(2798) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [59]; uri=[sip:2.2.2.2:5060;transport=tcp] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:174]: get_hdr_field(): to body (39)[<sip:2.2.2.2:5060;transport=tcp>], to tag (13)[1mB9HryQ8ZBFc] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <10> <OPTIONS> 19(2798) DEBUG: <core> [core/receive.c:319]: receive_msg(): --- received sip message - reply - call-id: [3aa18693487268dc-2790(a)1.1.1.1] - cseq: [10 OPTIONS] 19(2798) DEBUG: <core> [core/parser/msg_parser.c:185]: get_hdr_field(): content_length=0 19(2798) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Are these SIP messages expected to come from other ports than the listen port (5060 in this case)? Also, if the worker source port is not 5060, shouldn’t the SIP message get updated with the correct port? In the case of OPTIONS, Freeswitch is replying correctly to the source port: 33940. However, in the case of an in dialog BYE, Freeswitch is NOT replying to the source port of the Kamailio messages, but only to port 5060. Here is an example (relayed from web sockets to freeswitch over TCP) INVITE (as received from Freeswitch): recv 1481 bytes from tcp/[1.1.1.1]:33940 at 16:56:47.920698: ------------------------------------------------------------------------ INVITE sip:991012@sip.domain <sip:991012@sip.domain>.com SIP/2.0 Record-Route: <sip:1.1.1.1;transport=tcp;r2=on;lr;nat=yes> Record-Route: <sip:1.1.1.1:5061;transport=tls;r2=on;lr;nat=yes> Via: SIP/2.0/TCP 1.1.1.1;branch=z9hG4bKd408.3f53e940ccb20c1033df4b3a8ebd8a34.0;i=1 Via: SIP/2.0/TLS 172.22.199.110:55304;received=5.5.5.5;rport=39518;branch=z9hG4bKPj5Css6JomCt9Cli2cCINbXi4FbPM5wewG;alias Max-Forwards: 69 From: "Noah Mehl" <sip:5135555555@inbound-jail>;tag=s3i3y2tiOCgnUId5TD4Vp0UChf9GyEy9 To: <sip:991012@inbound-jail> Contact: <sip:74895612@172.22.199.110:54887;transport=tls;alias=5.5.5.5~39518~3> Call-ID: 5aoRBMBHahxqSLzrIpFnlfRz.UcGsmfq CSeq: 27271 INVITE Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: replaces, norefersub, gruu User-Agent: Blink Pro 4.6.0 (MacOSX) Content-Type: application/sdp Content-Length: 528 v=0 o=- 3812979407 3812979407 IN IP4 5.5.5.5 s=Blink Pro 4.6.0 (MacOSX) t=0 0 m=audio 50016 RTP/SAVP 113 0 101 c=IN IP4 5.5.5.5 a=rtcp:50017 a=rtpmap:113 opus/48000/2 a=fmtp:113 useinbandfec=1 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:UhHq6hth9HqALmiJ3AEeoGkixObBzkLURG60wJKT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:VKYaSCVwgvXCPaRvudTrgLORhWmOA7wyDJVeGjcu a=sendrecv a=oldmediaip:172.22.199.110 a=oldmediaip:172.22.199.110 ------------------------------------------------------------------------ This doesn’t seem valid, as the top Via doesn’t have a port specified? For reference, just rebuilt form the 5.4 branch today: commit 62dff5b8b157236cae7defe64291a6e4a8ae27b5 (upstream/5.4) Author: Kamailio Dev <kamailio.dev(a)kamailio.org> Date: Wed Oct 28 20:16:28 2020 +0100 modules: readme files regenerated - modules ... [skip ci] Thanks! ~Noah

6 19

Double port in TO header
by David Villasmil 30 Oct '20

30 Oct '20

Hello guys, This happened to me a long time ago but i totally forgot the reason. I using a simple dispatcher, and when relaying out to the destination, kamailio is adding an additional :5060 port to the To header: To: <sip:910609738@172.31.6.1:5080:5060> The Destination is actually 172.16.6.1:5080, i don't know why Kamailio is adding that :5060 The dispatcher is if (!ds_select_dst($var(dispatcher_outbound), "4")) { xlog("[DISPATCH]: ds_select_dst FAILED!\n"); send_reply("404", "No destination"); exit; } else { xlog("[DISPATCH]: ds_select_dst was succesful\n"); } xlog("L_DBG", "[DISPATCH]: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); Any ideas? I totally forgot... and it's late, i guess... Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 8

Re: [SR-Users] Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 30 Oct '20

30 Oct '20

Perfect, that's what I was looking for. :) Thanks! > This is what I have always done: > > # export PATH=$PATH:/usr/pgsql-${MAJOR_VER}/bin > # cd src/modules/db_postgres > # make install > > The `pg_config` utility in /usr/pgsql-${MAJOR_VER}/bin ensures that the proper header include path is supplied. > > -- Alex This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

1 0

Building db_postgres with PostgreSQL headers and libs in non-standard location
by Chaigneau, Nicolas 29 Oct '20

29 Oct '20

Hello, I'd like to build Kamailio module db_postgres, with the PostgreSQL headers and libs in a non-standard location. I'm trying to figure out how to do that, if it's possible, without having to edit db_postgres/Makefile. I see in the Makefile that it's trying to find the files relative to $(LOCALBASE): # use standard know paths # libpq-fe.h locations DEFS +=-I$(LOCALBASE)/include -I$(LOCALBASE)/pgsql/include \ -I$(SYSBASE)/include/pgsql -I$(SYSBASE)/include/postgresql \ -I$(SYSBASE)/include/postgresql/8.0 LIBS +=-L$(LOCALBASE)/lib -L$(LOCALBASE)/pgsql/lib \ -L$(LOCALBASE)/lib/pgsql -lpq However, LOCALBASE (which defaults to /usr/local) is used for many things other than PostgreSQL, so changing that is not a great idea, I think... Any idea ? Regards, Nicolas. This message contains information that may be privileged or confidential and is the property of the Capgemini Group. It is intended only for the person to whom it is addressed. If you are not the intended recipient, you are not authorized to read, print, retain, copy, disseminate, distribute, or use this message or any part thereof. If you receive this message in error, please notify the sender immediately and delete all copies of this message.

2 2

www.kamailio.org maintenance on 29th October 2020, 10:00 CET
by Henning Westerholt 29 Oct '20

29 Oct '20

Hello, there will be a short planned maintenance on www.kamailio.org<http://www.kamailio.org> on 29th October 2020 at 10:00 CET. Relevant services including main homepage and documentation wiki will be unavailable for about 10-15 minutes. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Kamailio unresponsive with Dialog+DMQ
by Patrick Wakano 28 Oct '20

28 Oct '20

Hello list, Hope all are doing well! We are running load tests in our Kamailio server, that is just making inbound and outbound calls and eventually (there is no identified pattern) Kamailio freezes and of course all calls start to fail. It does not crash, it just stops responding and it has to be killed -9. When this happens, SIP messages are not processed, dmq keepalive fails (so the other node reports as down), dialog KA are not sent, but Registrations from UAC seem to still go out (logs from local_route are seen). We don't have a high amount of cps, it is max 3 or 4 per sec, and it gets around 1900 active calls. We are now using Kamailio 5.2.8 installed from the repo on a CentOS7 server. Dialog has KA active and DMQ (with 2 workers) is being used on an active-active instance. >From investigation using GDB as pasted below, I can see UDP workers are stuck on a lock either on a callback from t_relay... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e95ffb0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e95ffb0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_UPDATE, dlg=0x7ffb362ea3c8, needlock=1, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a61f28e in dlg_on_send (t=0x7ffb36c98120, type=16, param=0x7fff2e9601e0) at dlg_handlers.c:739 #7 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36c98198, type=16, trans=0x7ffb36c98120, params=0x7fff2e9601e0) at t_hooks.c:260 #8 0x00007ffb2ef286d0 in run_trans_callbacks (type=16, trans=0x7ffb36c98120, req=0x7ffb742f27e0, rpl=0x0, code=-1) at t_hooks.c:287 #9 0x00007ffb2ef38ac1 in prepare_new_uac (t=0x7ffb36c98120, i_req=0x7ffb742f27e0, branch=0, uri=0x7fff2e9603e0, path=0x7fff2e9603c0, next_hop=0x7ffb742f2a58, fsocket=0x7ffb73e3e968, snd_flags=..., fproto=0, flags=2, instance=0x7fff2e9603b0, ruid=0x7fff2e9603a0, location_ua=0x7fff2e960390) at t_fwd.c:381 #10 0x00007ffb2ef3d02d in add_uac (t=0x7ffb36c98120, request=0x7ffb742f27e0, uri=0x7ffb742f2a58, next_hop=0x7ffb742f2a58, path=0x7ffb742f2e20, proxy=0x0, fsocket=0x7ffb73e3e968, snd_flags=..., proto=0, flags=2, instance=0x7ffb742f2e30, ruid=0x7ffb742f2e48, location_ua=0x7ffb742f2e58) at t_fwd.c:811 #11 0x00007ffb2ef4535a in t_forward_nonack (t=0x7ffb36c98120, p_msg=0x7ffb742f27e0, proxy=0x0, proto=0) at t_fwd.c:1699 #12 0x00007ffb2ef20505 in t_relay_to (p_msg=0x7ffb742f27e0, proxy=0x0, proto=0, replicate=0) at t_funcs.c:334 or loose_route... #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1bce08 in futex_get (lock=0x7ffb35217b90) at ../../core/futexlock.h:108 #2 0x00007ffb2b1bec44 in bcast_dmq_message1 (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>, incl_inactive=0) at dmq_funcs.c:156 #3 0x00007ffb2b1bf46b in bcast_dmq_message (peer=0x7ffb35e8bf38, body=0x7fff2e9629d0, except=0x0, resp_cback=0x7ffb2a8a0ab0 <dlg_dmq_resp_callback>, max_forwards=1, content_type=0x7ffb2a8a0a70 <dlg_dmq_content_type>) at dmq_funcs.c:188 #4 0x00007ffb2a6448fa in dlg_dmq_send (body=0x7fff2e9629d0, node=0x0) at dlg_dmq.c:88 #5 0x00007ffb2a64da5d in dlg_dmq_replicate_action (action=DLG_DMQ_STATE, dlg=0x7ffb363e0c10, needlock=0, node=0x0) at dlg_dmq.c:628 #6 0x00007ffb2a62b3bf in dlg_onroute (req=0x7ffb742f11d0, route_params=0x7fff2e962ce0, param=0x0) at dlg_handlers.c:1538 #7 0x00007ffb2e7db203 in run_rr_callbacks (req=0x7ffb742f11d0, rr_param=0x7fff2e962d80) at rr_cb.c:96 #8 0x00007ffb2e7eb2f9 in after_loose (_m=0x7ffb742f11d0, preloaded=0) at loose.c:945 #9 0x00007ffb2e7eb990 in loose_route (_m=0x7ffb742f11d0) at loose.c:979 or t_check_trans: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2a5ea9c6 in futex_get (lock=0x7ffb35e78804) at ../../core/futexlock.h:108 #2 0x00007ffb2a5f1c46 in dlg_lookup_mode (h_entry=1609, h_id=59882, lmode=0) at dlg_hash.c:709 #3 0x00007ffb2a5f27aa in dlg_get_by_iuid (diuid=0x7ffb36326bd0) at dlg_hash.c:777 #4 0x00007ffb2a61ba1d in dlg_onreply (t=0x7ffb36952988, type=2, param=0x7fff2e963bf0) at dlg_handlers.c:437 #5 0x00007ffb2ef285b6 in run_trans_callbacks_internal (cb_lst=0x7ffb36952a00, type=2, trans=0x7ffb36952988, params=0x7fff2e963bf0) at t_hooks .c:260 #6 0x00007ffb2ef286d0 in run_trans_callbacks (type=2, trans=0x7ffb36952988, req=0x7ffb3675c360, rpl=0x7ffb742f1930, code=200) at t_hooks.c:28 7 #7 0x00007ffb2ee7037f in t_reply_matching (p_msg=0x7ffb742f1930, p_branch=0x7fff2e963ebc) at t_lookup.c:997 #8 0x00007ffb2ee725e4 in t_check_msg (p_msg=0x7ffb742f1930, param_branch=0x7fff2e963ebc) at t_lookup.c:1101 #9 0x00007ffb2eee44c7 in t_check_trans (msg=0x7ffb742f1930) at tm.c:2351 And the DMQ workers are here: #0 0x00007ffb74e9bbf9 in syscall () from /lib64/libc.so.6 #1 0x00007ffb2b1d6c81 in futex_get (lock=0x7ffb35217c34) at ../../core/futexlock.h:108 #2 0x00007ffb2b1d7c3a in worker_loop (id=1) at worker.c:86 #3 0x00007ffb2b1d5d35 in child_init (rank=0) at dmq.c:300 Currently I will not be able to upgrade to latest 5.4 version to try to reproduce the error and since 5.2.8 has already reached end-of-life, maybe is there anything I can do on the configuration to avoid such condition? Any ideas are welcome! Kind regards, Patrick Wakano

3 6

Kamailio v5.4.2 Released
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, Kamailio SIP Server v5.4.2 stable release is out. This is a maintenance release of the latest stable branch, 5.4, that includes fixes since the release of v5.4.1. 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 v5.4.1. For more details about version 5.4.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/10/kamailio-v5-4-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Releasing Kamailio v5.4.2
by Daniel-Constantin Mierla 27 Oct '20

27 Oct '20

Hello, I am considering to release Kamailio v5.4.2 sometime next week, likely on Tuesday, October 27, 2020. This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

1 1

KEMI and Native mixing
by Marcin Kowalczyk 26 Oct '20

26 Oct '20

Hi, Is it possible to mix both KEMI (lua) and Native configs in one kamailio instance? So some blocks are called from native script and some others from KEMI (lua)? Marcin

6 5

Kamailio-5.x and Asterisk16.X Integration Walkthrough Videos or Wikis
by Lorem Ipsum 26 Oct '20

26 Oct '20

Hi there, After reading about Kamailio, i like this project but i am having trouble finding the wikis or step by step guide to successfully integrate asterisk with kamailio for simple use case where phones register to the asterisk pbx via kamailio and make/receive calls to ITSP, there is no proper documentation on 2 products integration, really appreciated your guidance and if somebody could provide me the step by step guide. Looking forward to the guides.

2 1

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