- sr-users - kamailio.org

Kamailio connection issue
by Tomas Zanet 09 Mar '20

09 Mar '20

Hi All, In these days I'm facing an issue or a misconfiguration, I don't know, with our remote apps (SIP clients) and our cloud server in which there's a Kamailio version 4.4.4. Sometimes our app takes some seconds (about 10-12sec) to connect to the Kamailio socket (sips on 5061) and our app is stuck on connect sytem call. To debug this behaviour I put a rule in the iptable like this: sudo iptables -I INPUT -p tcp -m tcp --dport 5061 -m state --state NEW -j LOG --log-level 1 --log-prefix "New Connection " when everything goes well, there is one row which describes the new incoming connection Mar 4 07:08:40 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=15819 DPT=5061 WINDOW=65535 RES=0x00 CWR ECE SYN URGP=0 when our app gets stuck in the connect, there are many attempts and seems that Kamailio is not ready to accept this connection: Mar 4 07:11:54 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 CWR ECE SYN URGP=0 Mar 4 07:11:54 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:55 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:55 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:55 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:56 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:56 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:56 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:11:58 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:12:00 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=64 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 Mar 4 07:12:05 server-xip-01 kernel: New Connection IN=eth0 OUT= MAC=bc:76:4e:20:1f:fb:e8:ed:f3:3a:52:41:08:00 SRC=X.X.X.X DST=Y.Y.Y.Y LEN=48 TOS=0x00 PREC=0x00 TTL=45 ID=0 DF PROTO=TCP SPT=52900 DPT=5061 WINDOW=65535 RES=0x00 SYN URGP=0 When it happens our app takes more than 10 sec to connect to the Kamailio socket. I tried to increase the children configuration parameter in the cfg file because I thought there were not enough tcp receiver but with no luck. Here are some configuration parameters: children = 16 // was 8 before found this issue # 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 open_files_limit=8192 tcp_max_connections=8192 tls_max_connections=8192 Here is the output of the ulimit command: ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 31393 max locked memory (kbytes, -l) 64 max memory size (kbytes, -m) unlimited open files (-n) 65535 pipe size (512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 8192 cpu time (seconds, -t) unlimited max user processes (-u) 31393 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited In the kamailio log file I don't see anything wrong, I don't see messages like "[tcp_main.c:3913]: handle_new_connect(): maximum number of tls connections exceeded". Here is the output of kamctl ps command Process:: ID=0 PID=11725 Type=main process - attendant Process:: ID=1 PID=11727 Type=udp receiver child=0 sock=10.208.194.26:5061 Process:: ID=2 PID=11728 Type=udp receiver child=1 sock=10.208.194.26:5061 Process:: ID=3 PID=11729 Type=udp receiver child=2 sock=10.208.194.26:5061 Process:: ID=4 PID=11730 Type=udp receiver child=3 sock=10.208.194.26:5061 Process:: ID=5 PID=11731 Type=udp receiver child=4 sock=10.208.194.26:5061 Process:: ID=6 PID=11732 Type=udp receiver child=5 sock=10.208.194.26:5061 Process:: ID=7 PID=11733 Type=udp receiver child=6 sock=10.208.194.26:5061 Process:: ID=8 PID=11734 Type=udp receiver child=7 sock=10.208.194.26:5061 Process:: ID=9 PID=11735 Type=udp receiver child=8 sock=10.208.194.26:5061 Process:: ID=10 PID=11736 Type=udp receiver child=9 sock=10.208.194.26:5061 Process:: ID=11 PID=11737 Type=udp receiver child=10 sock=10.208.194.26:5061 Process:: ID=12 PID=11738 Type=udp receiver child=11 sock=10.208.194.26:5061 Process:: ID=13 PID=11739 Type=udp receiver child=12 sock=10.208.194.26:5061 Process:: ID=14 PID=11740 Type=udp receiver child=13 sock=10.208.194.26:5061 Process:: ID=15 PID=11741 Type=udp receiver child=14 sock=10.208.194.26:5061 Process:: ID=16 PID=11742 Type=udp receiver child=15 sock=10.208.194.26:5061 Process:: ID=17 PID=11743 Type=slow timer Process:: ID=18 PID=11744 Type=timer Process:: ID=19 PID=11745 Type=secondary timer Process:: ID=20 PID=11746 Type=MI FIFO Process:: ID=21 PID=11747 Type=USRLOC Timer Process:: ID=22 PID=11748 Type=USRLOC Timer Process:: ID=23 PID=11749 Type=USRLOC Timer Process:: ID=24 PID=11750 Type=USRLOC Timer Process:: ID=25 PID=11751 Type=ctl handler Process:: ID=26 PID=11752 Type=RTIMER SEC EXEC Process:: ID=27 PID=11753 Type=TIMER NH Process:: ID=28 PID=11754 Type=tcp receiver (generic) child=0 Process:: ID=29 PID=11755 Type=tcp receiver (generic) child=1 Process:: ID=30 PID=11756 Type=tcp receiver (generic) child=2 Process:: ID=31 PID=11757 Type=tcp receiver (generic) child=3 Process:: ID=32 PID=11758 Type=tcp receiver (generic) child=4 Process:: ID=33 PID=11759 Type=tcp receiver (generic) child=5 Process:: ID=34 PID=11760 Type=tcp receiver (generic) child=6 Process:: ID=35 PID=11761 Type=tcp receiver (generic) child=7 Process:: ID=36 PID=11762 Type=tcp receiver (generic) child=8 Process:: ID=37 PID=11763 Type=tcp receiver (generic) child=9 Process:: ID=38 PID=11764 Type=tcp receiver (generic) child=10 Process:: ID=39 PID=11765 Type=tcp receiver (generic) child=11 Process:: ID=40 PID=11766 Type=tcp receiver (generic) child=12 Process:: ID=41 PID=11767 Type=tcp receiver (generic) child=13 Process:: ID=42 PID=11768 Type=tcp receiver (generic) child=14 Process:: ID=43 PID=11769 Type=tcp receiver (generic) child=15 Process:: ID=44 PID=11770 Type=tcp main process Can anyone address me to find out this happens, please? Please let me know if you need further information... Thanks in advance, Best Regards Tomas Tomas Zanet R&D Software Designer tzanet(a)came.com CAME S.p.A.

4 5

Using DNS hostname with the Dispatcher module to determine online routes
by Thomas Cavill 09 Mar '20

09 Mar '20

Hi all, I am wondering if it is possible for the dispatcher module to monitor multiple endpoints that resolve from a single hostname using DNS. Can I configure a single DNS name in the dispatcher table which resolves to 3 IP addresses and have them monitored by OPTIONS pings, and then select the first one that is marked as active to route requests to? I haven't found much help online to know if this is possible/supported. i.e. Does the dispatcher module support monitoring of multiple results returned from DNS NAPTR records? Thanks, Tom

2 1

usrloc db_mode 2 and dropped contacts
by Alex Balashov 09 Mar '20

09 Mar '20

Hi, On 4.4.7, since switching to db_mode 2 for usrloc with the following settings: modparam("usrloc", "db_mode", 2) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "skip_remote_socket", 1) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "matching_mode", 0) I seem to run into a situation where, despite starting from zero local registrations and an empty DB `location` table, out of ~1700 registrations visible in the runtime usrloc table, about 60-80 never make it into the database. This is because they're being continuously UPDATE'd rather than INSERT'd, so there is some presumption that they should already be in the Postgres database. But they're not. This gap seems to very slowly grow over time. I understand that this option is intended to deal with a scenario like that: https://kamailio.org/docs/modules/5.3.x/modules/usrloc.html#usrloc.p.db_che… but the documentation doesn't say it needs to be on for db_mode 2 to work correctly, and it's off by default. I'm wondering what causes certain registrants to get into this state in the first place, where the registrar presumes that they've already been synced to the DB and only need to be update'd on renewal, where in fact they have not been synced. Causes I've ruled out: 1) The contacts being ignored due to non-local socket or other invalidation; 2) Contacts being explicitly deleted by an outside process. Any ideas appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

4 8

problem in sending out calls
by Mohammad Reza Keshavarzianpoor 06 Mar '20

06 Mar '20

Hi, I installed kamailio and siremis. I receive incoming calls from gateway and all is ok. but about sending out calls using dynamic routes I have problem, always I get "not found" error on calling out phone numbers. 1.I need to send my did number with call to approved by gateway 2.do drouting module needs to enabled by config file? 3. if any better solution than drouting, write please, all I need is every subscriber have its did number and route on outgoing calls too. Thanks

1 0

problem in sending out calls to gateway
by Mohammad Reza Keshavarzianpoor 06 Mar '20

06 Mar '20

Hi, I installed kamailio and siremis. I receive incoming calls from gateway and all is ok. but about sending out calls using dynamic routes I have problem, always I get "not found" error on calling out phone numbers. 1.I need to send my did number with call to approved by gateway 2.do drouting module needs to enabled by config file? 3. if any better solution than drouting, write please, all I need is every subscriber have its did number and route on outgoing calls too. Thanks

1 0

rtpengine failover in same SIP dialog with response Unknown call-ide
by Michal Popovic 06 Mar '20

06 Mar '20

Hi, we are facing the issue with automated failover in same SIP dialog in rtpengine module. This issue occurs after we upgraded version 4.4.7 to 5.3.2 and only for rtpengines that are in another datacenter with latency higher than about 100ms. configuration modparam("rtpengine", "rtpengine_sock", "1 == udp:10.0.0.1:2222 udp:10.0.0.2:2222") modparam("rtpengine", "rtpengine_sock", "1 == udp:10.0.0.3:2222 udp:10.0.0.4:2222") INVITE request from kamailio to 10.0.0.1:2222 3:ICE6:remove5:flagsl13:trust-address16:always-transcode17:asymmetric-codecs12:loop-protecte7:replacel6:origin18:session-connectione18:transport-protocol7:RTP/AVP8:rtcp-muxl5:demuxe7:call-id20:xxx:received-froml3:IP414:x.x.x.xe8:from-tag10:naluniuntp7:command5:offere RESPONSE 6:result2:oke ANSWER request is than send to 10.0.0.2:2222 with error 3:ICE5:force5:flagsl13:trust-address16:always-transcode17:asymmetric-codecs12:loop-protecte7:replacel6:origin18:session-connectione18:transport-protocol9:RTP/SAVPF8:rtcp-muxl6:accept5:offere7:call-id20:l63nam58lomphlpvli9o13:received-froml3:IP414:18.212.137.170e8:from-tag10:naluniuntp6:to-tag10:as297c65177:command6:answere RESPONSE d6:result5:error12:error-reason15:Unknown call-ide Only possible fix is to have only one setid with only one server. First RTPengine is not marked as failed in list of rtpengine servers after the first response.

2 5

uac_replace_from in 200 OK of BYE
by Grant Bagdasarian 06 Mar '20

06 Mar '20

Hello, We've been using the uac_replace_from function to anonymize the From header to anonymous(a)anonymous.invalid<mailto:anonymous@anonymous.invalid>. The uac module makes use of the dialog module for restoring the original URI. Everything works fine, except when the callee sends the BYE request (replaced uri is present in To header): the 200 OK received by the callee of the BYE suddenly contains the original URI in the To header. We're running Kamailio 5.1. Is this a known bug or limitation of the uac module? Any way to have this replaced correctly (preferably automatically by uac module)? Regards, Grant

3 4

Kamailio Websocket Module call hangup issue with freeswitch
by Murugan Pandian 06 Mar '20

06 Mar '20

Hello, We are using Kamailio as a register freeswitch as a Media server for SIP its working fine with out any issue. But when we use Websocket module (WebRTC) i can able to register and make call , Even call also connected from browser to PSTN but call will disconnect after 30 seconds look like ICE gathering issue can i know its NAT issue Proxy issue. Regard's Murugan Panidan

2 2

sip domain
by Joli Martinez 05 Mar '20

05 Mar '20

Hello, We have several SIP domains coming into our SBC. I need to build a Kamailio box that will 302 the call to the appropriate server based on the domain. If the call does not come in with a recognized domain I would like for it to 404 the call. Not sure how to do this. Can someone point me in the right direction?

5 5

Unexpected behavior on tmrec with monthly recurrence
by David Gonçalves 04 Mar '20

04 Mar '20

Hi, During some tests with the module tmrec, we stumble upon an unexpected behavior of the monthly recurrence with days of the week. The last week does not match our expected date. For example, the last Monday of March 2020 is day 30 (fifth Monday of the month) however, neither the 5MO or -1MO configuration on the bday parameter match with a timestamp configured to day 30. The same situation is verified on months with only 4 weeks. To easily explain the behavior we create this configuration: $var(ts)="1584355625"; # Monday, March 16, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|3MO||||","$(var(ts){s.int})")) { xlog("L_INFO","Epoch $var(ts) matches with 3MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 3MO\n"); } $var(ts)="1584960425"; # Monday, March 23, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|4MO||||","$(var(ts){s.int})")) { xlog("L_INFO","Epoch $var(ts) matches with 4MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 4MO\n"); } $var(ts)="1585565225"; # Monday, March 30, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|5MO||||","$(var(ts){s.int })")){ xlog("L_INFO","Epoch $var(ts) matches with 5MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with 5MO\n"); } $var(ts)="1584960425"; # Monday, March 23, 2020 10:47:05 if(tmrec_match("20200101T090000|PT8H|monthly||1|-1MO||||","$(var(ts){s.int })")){ xlog("L_INFO","Epoch $var(ts) matches with -1MO\n"); }else{ xlog("L_INFO","Epoch $var(ts) doesn't match with -1MO\n"); } The results were: Epoch 1584355625 matches with 3MO Epoch 1584960425 matches with 4MO Epoch 1585565225 doesn't match with 5MO Epoch 1584960425 matches with -1MO Are we configuring the tmrec_match wrongly? -- Cumprimentos / Best regards, *David Gonçalves* Research and Development Technician Phone: +351 256 370 980 Email: david.goncalves(a)itcenter.com.pt

2 5

Outbound calling with user based carrier
by Gaurav Bmotra 04 Mar '20

04 Mar '20

Hi Is it possible to configure user based carrier on kamailio for outbound calling ? Thank you Regards

2 1

Help me open ports for my kamailio
by Stefan Troplev 04 Mar '20

04 Mar '20

Hi, I've been struggling around with my kamailio configuration. I've managed to install kamailio on Ubuntu Server and configured 2 clients, I've registered them in Zoiper to my server. They can call each other, but are unable to talk. There is no audio heard. Ports opened on the router are 20000-30000 UDP and both TCP and UDP for 5060 and 5062. Are these ports correctly opened? Thank you. kamailio.cfg > #!KAMAILIO > #!define WITH_MYSQL > #!define WITH_AUTH > #!define WITH_USRLOCDB > #!define WITH_ACCDB > # > # Kamailio (OpenSER) SIP Server v5.2 - default configuration script > # - web: https://www.kamailio.org > # - git: https://github.com/kamailio/kamailio > # > # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> > # > # Refer to the Core CookBook at https://www.kamailio.org/wiki/ > # for an explanation of possible statements, functions and parameters. > # > # Note: the comments can be: > # - lines starting with #, but not the pre-processor directives, > # which start with #!, like #!define, #!ifdef, #!endif, #!else, > #!trydef, > # #!subst, #!substdef, ... > # - lines starting with // > # - blocks enclosed in between /* */ > # > # 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 > # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING > # > # *** 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 block 401 and 407 authentication replies execute: > # - define WITH_BLOCK401407 > # > # *** 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 > > ####### Include Local Config If Exists ######### > import_file "kamailio-local.cfg" > > ####### 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. > #!ifndef DBURL > #!define DBURL "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 ######### > > ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR > #!ifdef WITH_DEBUG > debug=4 > log_stderror=yes > #!else > debug=2 > log_stderror=no > #!endif > > memdbg=5 > memlog=5 > > log_facility=LOG_LOCAL0 > log_prefix="{$mt $hdr(CSeq) $ci} " > > /* number of SIP routing processes for each UDP socket > * - value inherited by tcp_children and sctp_children when not set > explicitely */ > children=8 > > /* uncomment the next line to disable TCP (default on) */ > # disable_tcp=yes > > /* number of SIP routing processes for all TCP/TLS sockets */ > # tcp_children=8 > > /* 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:10.0.0.10:5060 > listen=udp:192.168.1.127:5060 advertise PublicIP:5060 > /* 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 > > /* upper limit for TCP connections (it includes the TLS connections) */ > tcp_max_connections=2048 > > #!ifdef WITH_TLS > enable_tls=yes > > /* upper limit for TLS connections */ > tls_max_connections=2048 > #!endif > > ####### 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" > pstn.gw_port = "" desc "PSTN GW Port" > #!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 > > ####### Modules Section ######## > > /* set paths to location of modules */ > # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" > > #!ifdef WITH_MYSQL > loadmodule "db_mysql.so" > #!endif > > loadmodule "jsonrpcs.so" > loadmodule "kex.so" > loadmodule "corex.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 "counters.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 > > # ----------------- setting module-specific parameters --------------- > > > # ----- jsonrpcs params ----- > modparam("jsonrpcs", "pretty_format", 1) > /* set the path to RPC fifo control file */ > # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") > /* set the path to RPC unix socket control file */ > # modparam("jsonrpcs", "dgram_socket", > "/var/run/kamailio/kamailio_rpc.sock") > > # ----- ctl params ----- > /* set the path to RPC unix socket control file */ > # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") > > # ----- tm params ----- > # auto-discard branches from previous serial forking leg > modparam("tm", "failure_reply_mode", 3) > # default retransmission timeout: 30sec > modparam("tm", "fr_timer", 30000) > # default invite retransmission timeout after 1xx: 120sec > modparam("tm", "fr_inv_timer", 120000) > > # ----- rr params ----- > # set next param to 1 to add value to ;lr param (helps with some UAs) > modparam("rr", "enable_full_lr", 0) > # do not append from tag to the RR (no need for this script) > modparam("rr", "append_fromtag", 0) > > # ----- registrar params ----- > modparam("registrar", "method_filtering", 1) > /* uncomment the next line to disable parallel forking via location */ > # modparam("registrar", "append_branches", 0) > /* uncomment the next line not to allow more than 10 contacts per AOR */ > # modparam("registrar", "max_contacts", 10) > /* max value for expires of registrations */ > modparam("registrar", "max_expires", 3600) > /* set it to 1 to enable GRUU */ > modparam("registrar", "gruu_enabled", 0) > > # ----- acc params ----- > /* what special events should be accounted ? */ > modparam("acc", "early_media", 0) > modparam("acc", "report_ack", 0) > modparam("acc", "report_cancels", 0) > /* by default ww do not adjust the direct of the sequential requests. > * if you enable this parameter, be sure the enable "append_fromtag" > * in "rr" module */ > modparam("acc", "detect_direction", 0) > /* account triggers (flags) */ > modparam("acc", "log_flag", FLT_ACC) > modparam("acc", "log_missed_flag", FLT_ACCMISSED) > modparam("acc", "log_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) > /* enhanced DB accounting */ > #!ifdef WITH_ACCDB > modparam("acc", "db_flag", FLT_ACC) > modparam("acc", "db_missed_flag", FLT_ACCMISSED) > modparam("acc", "db_url", DBURL) > modparam("acc", "db_extra", > "src_user=$fU;src_domain=$fd;src_ip=$si;" > "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") > #!endif > > # ----- usrloc params ----- > /* enable DB persistency for location entries */ > #!ifdef WITH_USRLOCDB > modparam("usrloc", "db_url", DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", MULTIDOMAIN) > #!endif > > # ----- auth_db params ----- > #!ifdef WITH_AUTH > modparam("auth_db", "db_url", DBURL) > modparam("auth_db", "calculate_ha1", yes) > modparam("auth_db", "password_column", "password") > modparam("auth_db", "load_credentials", "") > modparam("auth_db", "use_domain", MULTIDOMAIN) > > # ----- permissions params ----- > #!ifdef WITH_IPAUTH > modparam("permissions", "db_url", DBURL) > modparam("permissions", "db_mode", 1) > #!endif > > #!endif > > # ----- alias_db params ----- > #!ifdef WITH_ALIASDB > modparam("alias_db", "db_url", DBURL) > modparam("alias_db", "use_domain", MULTIDOMAIN) > #!endif > > # ----- speeddial params ----- > #!ifdef WITH_SPEEDDIAL > modparam("speeddial", "db_url", DBURL) > modparam("speeddial", "use_domain", MULTIDOMAIN) > #!endif > > # ----- domain params ----- > #!ifdef WITH_MULTIDOMAIN > modparam("domain", "db_url", DBURL) > /* register callback to match myself condition with domains list */ > modparam("domain", "register_myself", 1) > #!endif > > #!ifdef WITH_PRESENCE > # ----- presence params ----- > modparam("presence", "db_url", DBURL) > > # ----- presence_xml params ----- > modparam("presence_xml", "db_url", DBURL) > modparam("presence_xml", "force_active", 1) > #!endif > > #!ifdef WITH_NAT > # ----- rtpproxy params ----- > modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") > > # ----- nathelper params ----- > modparam("nathelper", "natping_interval", 30) > modparam("nathelper", "ping_nated_only", 1) > modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) > modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") > > # params needed for NAT traversal in other modules > modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") > modparam("usrloc", "nat_bflag", FLB_NATB) > #!endif > > #!ifdef WITH_TLS > # ----- tls params ----- > modparam("tls", "config", "/etc/kamailio/tls.cfg") > #!endif > > #!ifdef WITH_ANTIFLOOD > # ----- pike params ----- > modparam("pike", "sampling_time_unit", 2) > modparam("pike", "reqs_density_per_unit", 16) > modparam("pike", "remove_latency", 4) > > # ----- htable params ----- > /* ip ban htable with autoexpire after 5 minutes */ > modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") > #!endif > > #!ifdef WITH_XMLRPC > # ----- xmlrpc params ----- > modparam("xmlrpc", "route", "XMLRPC"); > modparam("xmlrpc", "url_match", "^/RPC") > #!endif > > #!ifdef WITH_DEBUG > # ----- debugger params ----- > modparam("debugger", "cfgtrace", 1) > 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 { > > # per request initial checks > route(REQINIT); > > # NAT detection > route(NATDETECT); > > # CANCEL processing > if (is_method("CANCEL")) { > if (t_check_trans()) { > route(RELAY); > } > exit; > } > > # handle retransmissions > if (!is_method("ACK")) { > if(t_precheck_trans()) { > t_check_trans(); > exit; > } > t_check_trans(); > } > > # handle requests within SIP dialogs > route(WITHINDLG); > > ### only initial requests (no To tag) > > # authentication > route(AUTH); > > # record routing for dialog forming requests (in case they are routed) > # - remove preloaded route headers > remove_hf("Route"); > if (is_method("INVITE|SUBSCRIBE")) { > record_route(); > } > > # account only INVITEs > if (is_method("INVITE")) { > setflag(FLT_ACC); # do accounting > } > > # 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); > } > > # Wrapper for relaying requests > route[RELAY] { > > # enable additional event routes for forwarded requests > # - serial forking, RTP relaying handling, a.s.o. > if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); > } > if (is_method("INVITE|SUBSCRIBE|UPDATE")) { > if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); > } > if (is_method("INVITE")) { > if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); > } > > if (!t_relay()) { > sl_reply_error(); > } > exit; > } > > # Per SIP request initial checks > route[REQINIT] { > #!ifdef WITH_ANTIFLOOD > # flood detection from same IP and traffic ban for a while > # be sure you exclude checking trusted peers, such as pstn gateways > # - local host excluded (e.g., loop to self) > if(src_ip!=myself) { > if($sht(ipban=>$si)!=$null) { > # ip is already blocked > xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); > exit; > } > if (!pike_check_req()) { > xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); > $sht(ipban=>$si) = 1; > exit; > } > } > #!endif > if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { > # silent drop for scanners - uncomment next line if want to reply > # sl_send_reply("200", "OK"); > exit; > } > > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > exit; > } > > if(is_method("OPTIONS") && uri==myself && $rU==$null) { > sl_send_reply("200","Keepalive"); > exit; > } > > if(!sanity_check("17895", "7")) { > xlog("Malformed SIP message from $si:$sp\n"); > exit; > } > } > > # Handle requests within SIP dialogs > route[WITHINDLG] { > if (!has_totag()) return; > > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > route(DLGURI); > if (is_method("BYE")) { > setflag(FLT_ACC); # do accounting ... > setflag(FLT_ACCFAILED); # ... even if the transaction fails > } else if ( is_method("ACK") ) { > # ACK is forwarded statelessly > route(NATMANAGE); > } else if ( is_method("NOTIFY") ) { > # Add Record-Route for in-dialog NOTIFY as per RFC 6665. > record_route(); > } > route(RELAY); > exit; > } > > if (is_method("SUBSCRIBE") && uri == myself) { > # in-dialog subscribe requests > route(PRESENCE); > exit; > } > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > route(RELAY); > exit; > } else { > # ACK without matching transaction ... ignore and discard > exit; > } > } > sl_send_reply("404","Not here"); > exit; > } > > # Handle SIP registrations > route[REGISTRAR] { > if (!is_method("REGISTER")) return; > > if(isflagset(FLT_NATS)) { > setbflag(FLB_NATB); > #!ifdef WITH_NATSIPPING > # do SIP NAT pinging > setbflag(FLB_NATSIPPING); > #!endif > } > if (!save("location")) { > sl_reply_error(); > } > exit; > } > > # User location service > route[LOCATION] { > > #!ifdef WITH_SPEEDDIAL > # search for short dialing - 2-digit extension > if($rU=~"^[0-9][0-9]$") { > if(sd_lookup("speed_dial")) { > route(SIPOUT); > } > } > #!endif > > #!ifdef WITH_ALIASDB > # search in DB-based aliases > if(alias_db_lookup("dbaliases")) { > route(SIPOUT); > } > #!endif > > $avp(oexten) = $rU; > if (!lookup("location")) { > $var(rc) = $rc; > route(TOVOICEMAIL); > t_newtran(); > switch ($var(rc)) { > case -1: > case -3: > send_reply("404", "Not Found"); > exit; > case -2: > send_reply("405", "Method Not Allowed"); > exit; > } > } > > # when routing via usrloc, log the missed calls also > if (is_method("INVITE")) { > setflag(FLT_ACCMISSED); > } > > route(RELAY); > exit; > } > > # Presence server processing > route[PRESENCE] { > if(!is_method("PUBLISH|SUBSCRIBE")) return; > > if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { > route(TOVOICEMAIL); > # returns here if no voicemail server is configured > sl_send_reply("404", "No voicemail service"); > exit; > } > > #!ifdef WITH_PRESENCE > if (!t_newtran()) { > sl_reply_error(); > exit; > } > > if(is_method("PUBLISH")) { > handle_publish(); > t_release(); > } else if(is_method("SUBSCRIBE")) { > handle_subscribe(); > t_release(); > } > exit; > #!endif > > # if presence enabled, this part will not be executed > if (is_method("PUBLISH") || $rU==$null) { > sl_send_reply("404", "Not here"); > exit; > } > return; > } > > # IP authorization and user authentication > route[AUTH] { > #!ifdef WITH_AUTH > > #!ifdef WITH_IPAUTH > if((!is_method("REGISTER")) && allow_source_address()) { > # source IP allowed > return; > } > #!endif > > if (is_method("REGISTER") || from_uri==myself) { > # authenticate requests > if (!auth_check("$fd", "subscriber", "1")) { > auth_challenge("$fd", "0"); > exit; > } > # user authenticated - remove auth header > if(!is_method("REGISTER|PUBLISH")) > consume_credentials(); > } > # if caller is not local subscriber, then check if it calls > # a local destination, otherwise deny, not an open relay here > if (from_uri!=myself && uri!=myself) { > sl_send_reply("403","Not relaying"); > exit; > } > > #!else > > # authentication not enabled - do not relay at all to foreign networks > if(uri!=myself) { > sl_send_reply("403","Not relaying"); > exit; > } > > #!endif > return; > } > > # Caller NAT detection > route[NATDETECT] { > #!ifdef WITH_NAT > force_rport(); > if (nat_uac_test("19")) { > if (is_method("REGISTER")) { > fix_nated_register(); > } else { > if(is_first_hop()) { > set_contact_alias(); > } > } > setflag(FLT_NATS); > } > #!endif > return; > } > > # RTPProxy control and signaling updates for NAT traversal > route[NATMANAGE] { > #!ifdef WITH_NAT > if (is_request()) { > if(has_totag()) { > if(check_route_param("nat=yes")) { > setbflag(FLB_NATB); > } > } > } > if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; > > if(nat_uac_test("8")) { > rtpproxy_manage("co"); > } else { > rtpproxy_manage("cor"); > } > > if (is_request()) { > if (!has_totag()) { > if(t_is_branch_route()) { > add_rr_param(";nat=yes"); > } > } > } > if (is_reply()) { > if(isbflagset(FLB_NATB)) { > if(is_first_hop()) > set_contact_alias(); > } > } > #!endif > return; > } > > # URI update for dialog requests > route[DLGURI] { > #!ifdef WITH_NAT > if(!isdsturiset()) { > handle_ruri_alias(); > } > #!endif > return; > } > > # Routing to foreign domains > route[SIPOUT] { > if (uri==myself) return; > > append_hf("P-hint: outbound\r\n"); > route(RELAY); > exit; > } > > # PSTN GW routing > route[PSTN] { > #!ifdef WITH_PSTN > # check if PSTN GW IP is defined > if (strempty($sel(cfg_get.pstn.gw_ip))) { > xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); > return; > } > > # route to PSTN dialed numbers starting with '+' or '00' > # (international format) > # - update the condition to match your dialing rules for PSTN routing > if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; > > # only local users allowed to call > if(from_uri!=myself) { > sl_send_reply("403", "Not Allowed"); > exit; > } > > # normalize target number for pstn gateway > # - convert leading 00 to + > if (starts_with("$rU", "00")) { > strip(2); > prefix("+"); > } > > if (strempty($sel(cfg_get.pstn.gw_port))) { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); > } else { > $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" > + $sel(cfg_get.pstn.gw_port); > } > > route(RELAY); > exit; > #!endif > > return; > } > > # XMLRPC routing > #!ifdef WITH_XMLRPC > route[XMLRPC] { > # allow XMLRPC from localhost > if ((method=="POST" || method=="GET") > && (src_ip==127.0.0.1)) { > # close connection only for xmlrpclib user agents (there is a bug in > # xmlrpclib: it waits for EOF before interpreting the response). > if ($hdr(User-Agent) =~ "xmlrpclib") > set_reply_close(); > set_reply_no_connect(); > dispatch_rpc(); > exit; > } > send_reply("403", "Forbidden"); > exit; > } > #!endif > > # Routing to voicemail server > route[TOVOICEMAIL] { > #!ifdef WITH_VOICEMAIL > if(!is_method("INVITE|SUBSCRIBE")) return; > > # check if VoiceMail server IP is defined > if (strempty($sel(cfg_get.voicemail.srv_ip))) { > xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); > return; > } > if(is_method("INVITE")) { > if($avp(oexten)==$null) return; > > $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) > + ":" + $sel(cfg_get.voicemail.srv_port); > } else { > if($rU==$null) return; > > $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) > + ":" + $sel(cfg_get.voicemail.srv_port); > } > route(RELAY); > exit; > #!endif > > return; > } > > # Manage outgoing branches > branch_route[MANAGE_BRANCH] { > xdbg("new branch [$T_branch_idx] to $ru\n"); > route(NATMANAGE); > } > > # Manage incoming replies > onreply_route[MANAGE_REPLY] { > xdbg("incoming reply\n"); > if(status=~"[12][0-9][0-9]") { > route(NATMANAGE); > } > } > > # Manage failure routing cases > failure_route[MANAGE_FAILURE] { > route(NATMANAGE); > > if (t_is_canceled()) exit; > > #!ifdef WITH_BLOCK3XX > # block call redirect based on 3xx replies. > if (t_check_status("3[0-9][0-9]")) { > t_reply("404","Not found"); > exit; > } > #!endif > > #!ifdef WITH_BLOCK401407 > # block call redirect based on 401, 407 replies. > if (t_check_status("401|407")) { > t_reply("404","Not found"); > exit; > } > #!endif > > #!ifdef WITH_VOICEMAIL > # serial forking > # - route to voicemail on busy or no answer (timeout) > if (t_check_status("486|408")) { > $du = $null; > route(TOVOICEMAIL); > exit; > } > #!endif > } > # The Kamailio configuration file for the control tools. # # Here you can set variables used in the kamctl and kamdbctl setup # scripts. Per default all variables here are commented out, the control tools # will use their internal default values. ## your SIP domain SIP_DOMAIN=192.168.1.127 ## chrooted directory # $CHROOT_DIR="/path/to/chrooted/directory" ## database type: MYSQL, PGSQL, ORACLE, DB_BERKELEY, DBTEXT, or SQLITE # by default none is loaded # # If you want to setup a database with kamdbctl, you must at least specify # this parameter. DBENGINE=MYSQL ## database host DBHOST=localhost ## database port DBPORT=3306 ## database name (for ORACLE this is TNS name) DBNAME=kamailio # database path used by dbtext, db_berkeley or sqlite # DB_PATH="/usr/local/etc/kamailio/dbtext" ## database read/write user DBRWUSER="kamailio" ## password for database read/write user DBRWPW="kamailiorw" ## database read only user DBROUSER="kamailioro" ## password for database read only user DBROPW="kamailioro" ## database access host (from where is kamctl used) # DBACCESSHOST=192.168.0.1 ## database super user (for ORACLE this is 'scheme-creator' user) # DBROOTUSER="root" ## password for database super user ## - important: this is insecure, targeting the use only for automatic testing ## - known to work for: mysql # DBROOTPW="dbrootpw" ## database character set (used by MySQL when creating database) #CHARSET="latin1" ## user name column # USERCOL="username" # SQL definitions # If you change this definitions here, then you must change them # in db/schema/entities.xml too. # FIXME # FOREVER="2030-05-28 21:32:15" # DEFAULT_Q="1.0" # Program to calculate a message-digest fingerprint # MD5="md5sum" # awk tool # AWK="awk" # gdb tool # GDB="gdb" # If you use a system with a grep and egrep that is not 100% gnu grep compatible, # e.g. solaris, install the gnu grep (ggrep) and specify this below. # # grep tool # GREP="grep" # egrep tool # EGREP="egrep" # sed tool # SED="sed" # tail tool # LAST_LINE="tail -n 1" # expr tool # EXPR="expr" # Describe what additional tables to install. Valid values for the variables # below are yes/no/ask. With ask (default) it will interactively ask the user # for an answer, while yes/no allow for automated, unassisted installs. # # If to install tables for the modules in the EXTRA_MODULES variable. # INSTALL_EXTRA_TABLES=ask # If to install presence related tables. # INSTALL_PRESENCE_TABLES=ask # If to install uid modules related tables. # INSTALL_DBUID_TABLES=ask # Define what module tables should be installed. # If you use the postgres database and want to change the installed tables, then you # must also adjust the STANDARD_TABLES or EXTRA_TABLES variable accordingly in the # kamdbctl.base script. # Kamailio standard modules # STANDARD_MODULES="standard acc lcr domain group permissions registrar usrloc msilo # alias_db uri_db speeddial avpops auth_db pdt dialog dispatcher # dialplan" # Kamailio extra modules # EXTRA_MODULES="imc cpl siptrace domainpolicy carrierroute userblacklist htable purple sca" ## type of aliases used: DB - database aliases; UL - usrloc aliases ## - default: none ALIASES_TYPE="DB" ## control engine: RPCFIFO ## - default RPCFIFO CTLENGINE="RPCFIFO" ## path to FIFO file for engine RPCFIFO RPCFIFOPATH="/var/run/kamailio/kamailio_rpc.fifo" ## check ACL names; default on (1); off (0) # VERIFY_ACL=1 ## ACL names - if VERIFY_ACL is set, only the ACL names from below list ## are accepted # ACL_GROUPS="local ld int voicemail free-pstn" ## check if user exists (used by some commands such as acl); ## - default on (1); off (0) # VERIFY_USER=1 ## verbose - debug purposes - default '0' VERBOSE=1 ## do (1) or don't (0) store plaintext passwords ## in the subscriber table - default '1' # STORE_PLAINTEXT_PW=0 ## Kamailio START Options ## PID file path - default is: /var/run/kamailio/kamailio.pid PID_FILE=/var/run/kamailio/kamailio.pid ## Extra start options - default is: not set # example: start Kamailio with 64MB share memory: STARTOPTIONS="-m 64" # STARTOPTIONS= >

2 3

kamailio 5.2.5 core WARNING
by Andrew Chen 03 Mar '20

03 Mar '20

Hi all, Can someone help shed some light as to what this core WARNING means? root@ashmaintpsg51:/etc/kamailio # kamailio -c kamailio.cfg 0(2192) WARNING: <core> [core/ppcfg.c:220]: pp_ifdef_level_check(): different number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 2 Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Dynamically change the transport protocol from UDP to TCP to avoid fragmentation
by supreeth herle 03 Mar '20

03 Mar '20

Hello Everyone, I have an IMS setup using Kamailo and is working as expected when the 4G Core Network and IMS is running in a single VM. However, when running in two different VMs registration is successful but calling fails. As per the scripts in kamailio ims example folder for P-CSCF, uac module is used to generate OPTIONS message to achieve a kind of NATPINGING to UEs. The ping achieved in this way times out with 408 when IMS and Core Network is run in different VMs. I suspect that the increasing packets size could be the issue for UE not responding to OPTIONS from P-CSCF (I have verified that OPTIONS message from P-CSCF is reaching the eNB). In this aspect, I would like to switch the transport protocol to TCP when MTU is greater than 1300 (please guide me if my thinking of switching to TCP to tackle NAT issue is wrong). I have explored the following options: tcp_reuse_port=yes udp_mtu=1300 udp_mtu_try_proto=TCP But, when I use these options, any request message which hit 1300 MTU mark are not sent out at all (in my case it is the NOTIFY request from SCSCF) and Kamailio has a warning saying that binding to address failed as the address is already in use. Following is the small part of the log: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19712]: DEBUG: ims_registrar_scscf [registrar_notify.c:2275]: notification_event_process(): About to free notification Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_reply: Destination URI: <null> Request URI: <null> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: INFO: rr [rr_mod.c:515]: pv_get_route_uri_f(): No route header present. Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (10.4.128.21:6060) Route-URI: Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: ERROR: <script>: NOTIFY (tel:0198765432100 (172.24.15.30:6060) to tel:0198765432100, 4125089758_4268630712(a)192.168.101.2) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within DLG Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Within loose route Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: PCSCF MO_indialog: Destination URI: sip:172.24.15.10:5060 Request URI: sip: 192.168.101.2:5060 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Source IP and Port: (172.24.15.30:6060) Route-URI: sip:mo@172.24.15.30;lr=on;ftag=4125089761 Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Received IP and Port: (10.4.128.21:5060) Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: NOTICE: <script>: Contact header: <sip:scscf.ims.mnc001.mcc001.3gppnetwork.org:6060> Feb 27 14:31:15 epc-ims /usr/local/sbin/kamailio[19730]: WARNING: <core> [core/tcp_main.c:1298]: tcp_do_connect(): binding to source address 10.4.128.21:5060 failed: Address already in use [98] I could be assuming things here about huge packet sizes causing the issue, if anyone has experienced similar issue in NATed scenario please let me know some hints or pointer to solve this. Thanks in advance. Best Regards, Supreeth

1 1

push notification via lua script on tsilo module
by max 03 Mar '20

03 Mar '20

I am expert in voip solution such as Asterisk and familiar with Kamailio .in one of our project we use the linphone app as a client mobile and we have to solved registering issue when extension has been unregistered in background mode. so i try to implement this scenario according federico document and enable extra module like this: loadmodule "tsilo.so" loadmodule "htable.so" #loadmodule "app_lua.so" loadmodule "cfgutils.so" I must disable app_lua.so because i haven't suitable lua script .i think most parts is OK and i can see ts_store parameters: ts.dump { Size: 1024 R-URIs: { R-URI: sip:1003@192.168.1.169:25378;rinstance=6cb3b1cc5efabdfb Hash: -238368019 Transactions: { Transaction: { Tindex: 9977 Tlabel: 1825563922 } } } Stats: { RURIs: 1 Max-Slots: 1 Transactions: 1 } } by the way push notification feature is enable in linphone app . Now i need a script (lua) for handling this issue (push notification for app) when we fetch SEND PUSH route. Could you please advice me ? do you have any example of this script? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 2

Kamailio - Force UAC to re register
by Sharad Kumar 03 Mar '20

03 Mar '20

Hey guys, I am implementing one scenario where I want the help of very smart people. So let’s say one UAC 999 registers to Asterisk 1 server via kamailio and asterisk 1 server goes down. Is there any way I can move that registration to another asterisk server or is there any way I can force UAC to re register so that I can route the new Register to the available asterisk ? Any help will be really appreciated. Thank you

2 1

Siremis web interface issue -- siptrace module active
by Bugaian A. Vitalie 02 Mar '20

02 Mar '20

Hi list, there is an issue in latest git version of Siremis with showing sip method content. If I select it from mysql I see it all good: select * from sip_trace where method = 'INVITE'; If I click it from web interface and select needed method I get only first register from sip_trace table. You can reproduce by /siremis/sipadmin/sip_trace_list <http://sbc.pride.md/siremis/sipadmin/sip_trace_list> then click on the method to see details. I tested on opera and chrome. Same result, something wrong with javascript, its not passing correctly parameter for selecting. Please have a look let me know if I can help. Thanks. Regards Vitalie A. Bugaian

2 9

Strange behaviour on $tu
by Duarte Rocha 02 Mar '20

02 Mar '20

Greetings, I have a cenario with the following header To = "sip:111234567890;npdi@111.222.333.444" If i print $tu I got "sip:111234567890". If i print $hdr(To) I get "sip:111234567890;npdi@111.222.333.444". If i don't include the "npdi" or any other user parameters, it works fine Is this the intended behaviour? Am i doing something wrong here? Best Regards,

2 3

How to detect NAT during authenticated registration of clients which adjust the headers
by Awal Junanto 02 Mar '20

02 Mar '20

Hi, We are building a service where we need to detect NAT when the clients register to our server. We are struggling in analyzing NAT status of some clients which modify their IP addresses/ports in the headers according to the value of "received" parameter sent during "401 Unauthorized" response. Here's the flow: Client->Server REGISTER sip:... Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@192.168.0.1:42251;transport=TLS;ob> ... Server->Client SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.0.1:41157 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias;received=1.2.3.4 WWW-Authenticate: ... ... Client->Server REGISTER sip:... Via: SIP/2.0/TLS 1.2.3.4:6201 ;rport;branch=z9hG4bKPj30093e5d-550d-4d4c-a9a2-22c3bd1cda7e;alias Contact: <sip:user@ 1.2.3.4:6201;transport=TLS;ob> Authorization: ... ... By the time the client is authenticated, there is no way to detect whether the request was coming from a natted device or not by just analysing the Via or Contact headers. Thanks in advance.

3 6

using kamailio as sip server
by Mohammad Reza Keshavarzianpoor 02 Mar '20

02 Mar '20

Hi, I'm new to Kamailio I need to have a kamalio as sip server which do a trunk to NGN Center(like a friend peer trunk in freepbx). I need to outgoing calls go through kamailio and forwarded to ngn(multi did values, a range of numbers ...) extensions will register on kamailio I need incoming calls forwarded to an extension or a pbx server, doesn't matter and both are good ... that server will have heavy traffic about 1000 concurrent calls... I tried a successful kamailio installation but I don't know how to config that for my purpose. even I tried installation of dsiprouter over a fresh debian 9 but stopped on pyspark installation. please give me your helps, advises, hints or ideas! best regards

2 1

Re: [SR-Users] dlg.end_dlg on RTP timeout
by Miteshkumar Thakkar 28 Feb '20

28 Feb '20

This is really helpful development. I am facing one issue, when I use this. RTPEngine is sending two "dlg.terminate_dlg" events for single call to kamailio with same call-id but reversed to-tag and from-tag in second event. I believe this is due to two separate RTP streams. Problem here is that - in first event kamailio terminates the call, hence when second event comes, kamailio says - call-id not found and send response back to RTPEngine. But still RTPEngine keeps sending that event message for three more times. Kamailio has to respond to it with call-id not found. Which is overhead. Not recommended for production environment. How can I avoid this? Any help is appreciated. Thank you Mitesh That's great, thank you! Cheers, Daniel On 06.09.18 11:15, Richard Fuchs wrote: >* This is now supported as per * >* https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701… <https://github.com/sipwise/rtpengine/commit/89084da8d820919b44a0244e16e6701…> * > >* Cheers * > >* On 2018-09-05 05:39, Daniel-Constantin Mierla wrote: * >> >>* There is the dlg.terminate_dlg rpc command that requires callid, * >>* from-tag and to-tag as parameters: * >> >>* * * >>* https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate… <https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dlg.r.terminate…> * >> >>* So it expects something like: * >> >>* <?xml version="1.0" encoding="UTF-8"?> * >>* <methodCall> * >>* <methodName>dlg.terminate_dlg</methodName> * >>* <params> * >>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>* <param><value><string>_FROM_TAG_VALUE__</string></value></param> * >>* <param><value><string>_TO_TAG_VALUE__</string></value></param> * >>* </params> * >>* </methodCall> * >> >>* I planned to make the from-tag and to-tag optional for quite some * >>* time, but didn't get the time for it yet. * >> >>* Cheers, * >>* Daniel * >> >>* On 05.09.18 08:52, Richard Fuchs wrote: * >>>* Yup that's exactly right. * >>> >>>* It would be fairly simple to implement an additional XMLRPC format * >>>* if there's a particular one that's more friendly towards Kamailio. * >>> >>>* Cheers * >>> >>>* On 2018-09-05 02:42, Daniel-Constantin Mierla wrote: * >>>> >>>>* Looking quickly at the readme of rtpengine application and digging * >>>>* a bit with google, it is something like rtpengine has to be started * >>>>* with * >>>> >>>>* -b http(s)://myrpcserver.ip/path -x 1 * >>>> >>>>* and then the xmlrpc request is going to be sent to that url, having * >>>>* a format like: * >>>> >>>>* <?xml version="1.0" encoding="UTF-8"?> * >>>>* <methodCall> * >>>>* <methodName>teardown</methodName> * >>>>* <params> * >>>>* <param><value><string>_CALLID_VALUE__</string></value></param> * >>>>* </params> * >>>>* </methodCall> * >>>> >>>>* Is it right? If yes, then I can try to make a sample config that * >>>>* could handle it using xmlops, xhttp and jsonrpcs modules. * >>>> >>>>* Cheers, * >>>>* Daniel * >>>> >>>> >>>>* On 05.09.18 08:24, Richard Fuchs wrote: * >>>>>* It does an XMLRPC callback. Currently there's two formats for it, * >>>>>* one is a sems sbc teardown request (using the from-tag), the other * >>>>>* is a generic "teardown" command using the call ID. * >>>>> >>>>>* Cheers * >>>>> >>>>>* On 2018-09-04 07:52, Daniel-Constantin Mierla wrote: * >>>>>> >>>>>>* Hello, * >>>>>> >>>>>>* what do you get from rtpengine on rtp timeout? An RPC call back * >>>>>>* or an http request? * >>>>>> >>>>>>* Cheers, * >>>>>>* Daniel * >>>>>> >>>>>> >>>>>>* On 04.09.18 12:48, Igor Olhovskiy wrote: * >>>>>>>* Hi all! * >>>>>>> >>>>>>>* Is there any way to end dialog in Kamailio on RTP Timeout event * >>>>>>>* on RTPEngine? * >>>>>>> >>>>>>>* Or only look at logs/redis database with external tool? * >>>>>>> >>>>>>>* Regards, Igor * >>>>>>> >>>>>>> >>>>>>>* _______________________________________________ * >>>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>>* 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 <http://www.asipto.com> * >>>>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>>>>> >>>>>> >>>>>>* _______________________________________________ * >>>>>>* Kamailio (SER) - Users Mailing List * >>>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>> >>>>> >>>>>* _______________________________________________ * >>>>>* Kamailio (SER) - Users Mailing List * >>>>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>>>* 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 <http://www.asipto.com> * >>>>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>>>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>>>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * >>> >>> >>>* _______________________________________________ * >>>* Kamailio (SER) - Users Mailing List * >>>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >>>* 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 <http://www.asipto.com> * >>* www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda> * >>* Kamailio World Conference -- www.kamailioworld.com <http://www.kamailioworld.com> * >>* Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com <http://www.asipto.com> * > > >* _______________________________________________ * >* Kamailio (SER) - Users Mailing List * >* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * >* 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 Kamailio World Conference -- www.kamailioworld.com Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com

2 11

Teams integration
by Bugaian A. Vitalie 28 Feb '20

28 Feb '20

Hello list, its been about one month for me playing with kamailio and I need some help to sort out a real life situation. I followed this guide https://skalatan.de/en/blog/kamailio-sbc-teams; great article, also got some inspiration from here https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/..articles look very similar. I manged to get my calls out to kamailio from TEAMS it works perfectly call gets connected and codec negotiation is fine, but I am facing issue geting the call inbound to TEAMS. My invite looks like this: ======================== U 217.26.163.205:5060 -> 52.114.75.24:5061 #22 INVITE sip:+37360844269@sip.pstnhub.microsoft.com:5061 SIP/2.0. Record-Route: <sip:sbc.pride.md:5061;transport=tls;lr>. Record-Route: <sip:217.26.163.205:5061;nat=yes;lr>. Via: SIP/2.0/UDP 217.26.163.205;branch=z9hG4bK7838.7417f50ed201fcada9609f5b7c4e520f.0. Via: SIP/2.0/UDP 192.168.169.102:5060 ;received=46.214.187.67;branch=z9hG4bK80e3f7e5cc50ea11806b6eeeb899592c;rport=5060. From: "+37379844267" <sip:+37379844267@sbc.pride.md>;tag=1604785394. To: "+37360844269" <sip:+37360844269@sip.pstnhub.microsoft.com:5061 ;user=phone>. Call-ID: 80E3F7E5-CC50-EA11-8069-6EEEB899592C(a)192.168.169.102. CSeq: 227 INVITE. sip:+37379844267@192.168.169.102:5060;gr=008A94E3-CA50-EA11-805B-6EEEB899592C;alias=46.214.187.67~5060~1Content-Type: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK. Max-Forwards: 69. User-Agent: SIPPER for PhonerLite. Session-Expires: 1800. Supported: 100rel, replaces, from-change, gruu, timer. P-Preferred-Identity: <sip:+37379844267@sbc.pride.md>. Content-Length: 362. Contact: <sip:+37379844267@sbc.pride.md:5061;user=phone;transport=tls>. . v=0. o=- 2307737351 1 IN IP4 217.26.163.205. s=SIPPER for PhonerLite. c=IN IP4 217.26.163.205. t=0 0. m=audio 36864 RTP/SAVP 8 101. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:owGy5+mMZNyO5+9lFmUUOK3WqpEsxJH0+jtWz =============================================== Anybody has a good invite exaple for Teams? Or do you see any issue with my invite? I do use : record_route_preset("sbc.pride.md:5061;transport=tls","217.26.163.205:5060 ;nat=yes"); add_rr_param(";r2=on"); before sending this out. Please let me know if you can help. Thanks. Vitalie.

3 7

Trouble with 302 Redirect
by Jim Leahy 27 Feb '20

27 Feb '20

Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks

5 4

Rtpproxy not work in 2 kamailio + rtpproxy setup
by Voip support 27 Feb '20

27 Feb '20

Dear Community, I have a problem with rtpproxy not changing the SDP IP in connection information in following scenario: SIP1 -> kamailio1 + rtpproxy1 -> kamailio2 + rtpproxy2 -> SIP2 The first kamailio put SDP IP correctly (from rtpproxy) then the second kamailio does not change SDP connection information. I think its related to SDP body which has: a=nortpproxy:yes line added by kamailio1. So i tried to remove the a=nortpproxy:yes line from SDP before calling rtpproxy_manage. But it looks like no difference in working - i think the SDP sent to rtpproxy is still having the line even when i removed it. The issue is not happening when i remove kamailio1 + rtpproxy1. In that case my IP is put correctly in SDP in INVITE and in 200 OK. In case of two kamailio with rtpproxy (kamailio1 and kamailio2) in signalling path i am getting: rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy for replies (SIP2 IP address in 200OK SDP is present instead of kamailio2 IP address). For request INVITE i am not getting the error but the IP in SDP doesn't change. kamailio2 config is now like below (but the attempt to remove a=nortpproxy:yes, did not work for me) # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } # if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; -- commented this out because sometimes rtpproxy was not used and i wanted to use it always # added below code to remove a=nortpproxy:yes from SDP before rtpproxy_manage but its still not working if( has_body("application/sdp") && (1 == sdp_get_line_startswith("$avp(nortpproxy)", "a=nortpproxy" )) ){ sdp_remove_line_by_prefix("a=nortpproxy:yes"); xlog("removed nortpproxy:yes from SDP"); } if(nat_uac_test("8")) { xlog("rtpproxy CO"); rtpproxy_manage("co","MY_PUBLIC_IP_HERE"); } else { xlog("rtpproxy COR"); rtpproxy_manage("cor"," MY_PUBLIC_IP_HERE "); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } Best regards, Tom PS. thanks for any hints to resolve the issue.

2 1

record_route_advertised_address causes null send socket error
by Daniel Greenwald 26 Feb '20

26 Feb '20

I am using record_route_advertised_address to insert a record route header with an SRV record. The header is added correctly and everything appears to be working fine however I get a CRITICAL log error when calling record_route_advertised_address('INTERNAL_SRV'); 2020-02-25T20:14:57.078235+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket 2020-02-25T20:14:57.078352+00:00,ip-10-0-16-53: CRITICAL: cbbcfe06-d181-467c-bbf4-78772aa0562f 2 16777053 INVITE:<core> [core/msg_translator.c:514]: lump_check_opt(): null send socket Any ideas? I'm considering using insert_hf() to manually add it and bypass whatever check is failing. Thoughts? Thanks

3 7

Access to CANCEL messages sent during parallel forking
by Marco Capetta 26 Feb '20

26 Feb '20

Hi All, I have a question regarding call forking and how to access failed branches: the ones for which kamailio sends out the CANCEL because on another one a 200OK was received. This is the scenario: - A calls B - 3 devices are registered on B, so a parallel forking is done to B1, B2 and B3 - B1, B2, and B3 all reply with 180 Ringing - B1 replies with 200OK - Kamailio CANCELs the branches with B2 and B3 What I need is access to each single CANCEL that Kamailio sends out to B2 and B3. In particular I need to know the branch-id and Via branch of all the cancelled branches. Additionally it would be perfect to have them inside the TM module, in order to have access also to internal AVPs. I tried with dedicated failure routes and also with per-branch failure routes, but I was never able to access to those values. I also didn't found a specific event-route that can help me solve this problem. Does anyone have an idea? Kamailio version 5.2.5 Thanks Regards Marco

3 5

Siremis v5.3.0 Released
by Daniel-Constantin Mierla 26 Feb '20

26 Feb '20

Siremis v5.3.0 is out – the open source web management interface for Kamailio SIP Server. This version comes with major changes and many new features, among them the removal of legacy flash charts (now uses pure JS), make the code compatible with PHP 7.x and web views to manage many new modules added up to Kamailio v5.3.x, including the replacement of MI control interface with JSONRPC. This is the first full release in the 5.x series of Siremis. Step by step installation tutorial, screenshots and changelog are available on the web at: * https://kb.asipto.com/siremis:install53x:main * https://kb.asipto.com/siremis:install32x:screenshots * https://github.com/asipto/siremis/blob/v5.3.0/ChangeLog The deployment of Siremis can be also done using Docker, see the guidelines at: * https://github.com/asipto/siremis/tree/master/misc/docker Siremis project is hosted on Github at: * https://github.com/asipto/siremis/ The web site for the project is at: * https://www.siremis.org Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Understanding set_rtp_engine with two sets
by Anthony Alba 25 Feb '20

25 Feb '20

There is a use case mentioned "This is useful if you have a set of RTP proxies that the caller must use, and another distinct set of RTP proxies that the callee must use. This is supported by all rtpengine commands except rtpengine_manage(). " How do you actually implement this - is it something like this? Say caller supposed to target Set 1and callee supposed to target Set 2: # offer/request route set_rtpengine_set(1, 2) rtpengine_offer() # cannot use rtpengine_manage() # callee sees set 2 in SDP, right? # reply_route set_rtpengine_set(2, 1) # reverse the order? Is this correct?? # OR set_rtpengine_set(1, 2) # keep order, module does autoreversal magic # caller sees Set 1 in SDP My question: in the reply route do you keep the same order (module is clever enough to autoreverse so caller sees Set 1) OR do you explicitly reverse the order and THAT means caller sees Set 1. Cheer Anthony

3 2

kamcli postgres support
by Daniel-Constantin Mierla 25 Feb '20

25 Feb '20

Hello, anyone in the community with Python knowledge using Postgres that can spend a bit of time to add support in kamcli? * https://github.com/kamailio/kamcli It should be mainly about the content of the code corresponding to db subcommand: * https://github.com/kamailio/kamcli/blob/master/kamcli/commands/cmd_db.py The target for it is to replace kamdbctl and has some command line variants for couple of its functions, now being implemented for MySQL. The rest of the code relies on sqlalchemy, which should have transparent support for both mysql and postgress (even more). Testing the other commands connecting to database with postgres would be good, though. With more and more diversity in the default shell by main Linux/Unix distros, we hope to be able to replace in the near future the kamctl and kamdbctl tools, which are more like bash scripting (a few issue being reported and fixed lately). Besides that, using python allows better input/parameter checking/validation as well as more options for formatted output. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 3

Database errors with usrloc db_mode=1 and dmq_usrloc
by Joel Serrano 25 Feb '20

25 Feb '20

Hello, I have a setup with 2 kamailio in active/idle, locations are shared via dmq_usrloc, but database is also used as a cache where other systems can access and query "real-time" data. The config we have is: # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 1) modparam("usrloc", "db_load", 0) modparam("usrloc", "db_insert_update", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "server_id_filter", 1) modparam("usrloc", "timer_procs", 2) modparam("usrloc", "timer_interval", 60) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) modparam("dmq_usrloc", "sync", 1) modparam("dmq_usrloc", "batch_msg_contacts", 50) # 50 contacts / message modparam("dmq_usrloc", "batch_size", 10000) # 10000 contacts / batch modparam("dmq_usrloc", "batch_usleep", 500000) # one batch every 500ms With this setup, we assume: 1- almost real-time updated location info in the database. 2- as long as one of the two Kam is up, location information will not be lost (as they will sync with each other via DMQ) 3- if both servers go down, locations are lost (as they will not reload from database on startup). Our tests work well with the above config, with one exception, on the idle node, we see a lot of (only happens on the node receiving location via dmq): Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1136]: db_update_ucontact_ruid(): sql use_table failed Feb 21 00:11:25 csbc02 csbc[13916]: ERROR: usrloc [ucontact.c:1657]: update_contact_db(): failed to update database I added debug logs to see if I could understand the reason: Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:478]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.2.1.181:5050 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:423]: usrloc_dmq_execute_action(): Received DMQ_UPDATE. Update contact info... Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:75]: add_contact(): aor: 1063362 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:76]: add_contact(): ci->ruid: uloc-1-5e502cd3-5629-1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:77]: add_contact(): aorhash: 928261667 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:115]: add_contact(): '1063362' Not found in usrloc Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:117]: add_contact(): Insert record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: usrloc [ul_callback.h:83]: run_ul_callbacks(): contact=0x7fe41ff41f40, callback type 1/15, id 0 entered Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:800]: dmq_ul_cb_contact(): Callback from usrloc with type=1 Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:826]: dmq_ul_cb_contact(): Contact received from DMQ... skip Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: <core> [db.c:481]: db_use_table(): invalid parameter value Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [ucontact.c:669]: db_insert_ucontact(): sql use_table failed Feb 21 14:38:34 csbc02 csbc[30796]: ERROR: usrloc [urecord.c:598]: insert_ucontact(): failed to insert in database Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:121]: add_contact(): Insert ucontact Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:124]: add_contact(): Release record Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: dmq_usrloc [usrloc_sync.c:126]: add_contact(): Unlock udomain Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: sl [sl.c:305]: send_reply(): reply in stateless mode (sl) Feb 21 14:38:34 csbc02 csbc[30796]: DEBUG: <core> [core/msg_translator.c:161]: check_via_address(): (10.2.1.181, 10.2.1.181, 0) Can anyone help me understand what's the problem? I have the server_id setting, and the modparam server_id_filter, so shouldn't the receiving node just ignore any database tasks? Thanks, Joel.

3 12

Using the To header to route a call
by Dovid Bender 24 Feb '20

24 Feb '20

Hi, We are fighting with a vendor and trying to explain to them that call routing should be based on the RURI on the INVITE and *not* the To header. I went through RFC 3261 and while there are many "indicators" and it seems like its screaming that we should be using the RURI, is anyone aware of any specific text (that I may have missed) where it says specifically call routing should be done based on the RURI and *NOT* the to header? TIA. Dovid https://tools.ietf.org/html/rfc3261#page-78 1) 8.1.1.1 Request-URI *The initial Request-URI of the message SHOULD be set to the value of the URI in the To field.* - Note that it says SHOULD but is NOT mandatory. Also this is for the *INITIAL* request. *This may or may not be the ultimate recipient of the request.* - Again this is the To field. The RURI is what we are supposed to follow when we want to know where the call should be delivered to. 2) 8.1.2 Sending the Request -Usage of the URI from the *To and From fields in the original request within subsequent requests is done for backwards compatibility with RFC 2543*, which used the URI for dialog identification. *In this specification, only the tags are used for dialog identification*. It is expected that mandatory reflection of the original To and From URI in mid-dialog requests will be deprecated in a subsequent revision of this specification. - While this is for subsequent requests (and not an original one) you see that the only point of the To field is to identify the dialog. 3) 8.2.2.1 To and Request-URI The To header field identifies the *original recipient of the request* designated by the user identified in the From field. - Note that the TO field identifies the ORIGINAL recipient, this **CAN** change. This is basically telling you that you *can't* rely on the To header for routing. *However, the Request-URI identifies the UAS that is to process the request*. - As you can see the Request-RU and *NOT* the TO header tells you where the request should go. 5) 12.2.1.1 Generating the Request a. The UAC uses the *remote target* and route set to build the *Request-URI and Route header field of the request*. - This is self explanatory. it shows that the RURI is using for routing and *NOT* the to header. b. The procedures in Section 8.1.2 will normally result in the request being *sent to the address* indicated by the topmost Route header field value or the *Request-URI* if no Route header field is present. Subject to certain restrictions, they allow the request to be sent to an alternate address (such as a default outbound proxy not represented in the route set). Again self explanatory. 6) 16.5 Determining Request Targets Each target in the set is *represented as a URI*. - This clearly shows that how we route is based on the RURI and *NOT* the to header. 7) 26.5 Privacy Thus it MAY be desirable for privacy reasons to create a To header field that differs from the Request-URI. - Again this shows that the to may not match up the RURI. Think of the To header as the contact in someones phone and the RURI as where we are calling. The to header may be bob(a)1.1.1.1 but the URI is where we are actually calling which is 12346(a)1.1.1.1 or bob_smith(a)1.1.1.1.1

3 2

Kamailio presence when account not equal extension
by Zhan Bazarov 24 Feb '20

24 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 3

CentOS 8 RPMs missing from OBS
by Anthony Alba 21 Feb '20

21 Feb '20

The 5.3.2-4 builds for CentOS 8 are not available here: https://download.opensuse.org/repositories/home:/kamailio:/v5.3.x-rpms/Cent… Is https://rpm.kamailio.org/centos/8/5.3/5.3.2/x86_64/ aka Pulp the official release location now? (With -0 builds) For CentOS 7 Pulp has 5.3.2-0 and OBS has 5.3.2-4. The version mismatch is confusing. Cheers Anthony

2 1

Kamailio presence when account not equal extension
by Zhan Bazarov 21 Feb '20

21 Feb '20

Hello, I'm using presence and pua modules, it works perfectly when username is equal to extension... But now I have case when username is not equal to the extension.. Like: extension: 1111 username: aythnbghhgt How I can implement this? Any ideas? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 1

acc_radius
by volga629 21 Feb '20

21 Feb '20

2 1

Understanding some usrloc modparams
by Joel Serrano 20 Feb '20

20 Feb '20

Hello, I have an installation using usrloc + db_mode=1. If I want on db_mode=1 or db_mode=2 the expired contacts to be cleaned up I have to enable the modparam db_timer_clean: https://www.kamailio.org/docs/modules/devel/modules/usrloc.html#usrloc.p.db… The docs say: "*Enable (1) or disable (0) cleaning of expired db records on timer basis for db_mode WRITE-BACK and WRITE-THROUGH. It uses the secondary timer process.*" I have the following doubts: 1- What does the reference "it uses *the* secondary timer process" mean? Is there a specific secondary timer just for this purpose? 2- Is #1 related to the modparam timer_procs? (so it uses one of those when enabled?) 3- Why/When would you need a timer_proces modparam with a value higher than "1"? Thanks, Joel.

4 6

Active-Active Registrar setup with users behind NAT persistence in event of server failure
by Jonathan Hunter 19 Feb '20

19 Feb '20

Hi Guys, I have a quick question wondered if anyone could provide some suggestions please?! We have a setup of two registrars in EC2 (kamailio 4.4 running on debian VMs) which act in an active/active fashion with a number of devices behind NAT where we use the DMQ dmq_t_replicate function to send the registration details between each server and append the PATH parameter so the registrar the device registered with is always associated with the client device. This all works fine, the only issue is persistency should we have a failure of one of the registrars, be it process or server and recovering the correct PATH detail so the end device can still be reached. I have tried this with separate databases with each registrar, which works when using cache and db but I ideally want the same database for the two servers and not change the architecture at this time unless it is a necessary. I have also tested with dmq_usrloc but this again provides the usrlocation details of the notification servers in the group(not path), again which I understand but I wondered if there is anything else I could try or its just a question of in the event of a failure half the registrations will be effected until it recovers and they re-register again if I dont want to have separate databases (location tables) or I add more network elements into the flow, such as HA pair or load balancer but just want to make sure I have exhausted all options or consider the risk before I look to do this. Many thanks! Jon

2 2

IMS Third party Register issue
by Lanvin ADON 19 Feb '20

19 Feb '20

Hi All, I'm facing an issue with the third party registration in the ISC interface between the SCSCF and the AS. The AS is triggered for deregistration but not for registration. Attached are logs for a register that not works and a deregister that works Regards Lanvin ADON

2 3

Documentation Broken Link
by Sam Ware 19 Feb '20

19 Feb '20

On the http://www.kamailio.org/w/documentation/ page, the Docbook tutorials <http://sip-router.org/docbook/sip-router/branch/master/> link is broken. -- Sam D Ware

2 2

xlog() format change from 5.3 to 5.4-nightly
by Joel Serrano 19 Feb '20

19 Feb '20

Hello, I've noticed that the output format of xlog() changes from 5.3 to 5.4-nightly. v5.3 xlog(): Feb 13 11:56:35 csbc01 csbc[27333]: NOTICE: {1 24727 REGISTER b1d374f6-9b8c53ff(a)A.B.C.D} <script>: [log] auth successful v5.4-nightly xlog(): Feb 13 11:56:35 csbc01 csbc[27333]: NOTICE: {1 24727 REGISTER b1d374f6-9b8c53ff(a)A.B.C.D} <script>: xlog_helper(): [log] auth successful Is there a way to avoid adding the extra "*xlog_helper():*" string? We have a lot of scripts that parse the K logs for different reasons and I would like to avoid having to update them all to match the new format. I've tried adding *modparam("xlog", "prefix", "")* but it only affects the "<script>:" part and not the "xlog_helper():" part. Thanks, Joel.

3 4

ds_is_from_list doesn't accept groupid from a variable?
by Björn Bylander 14 Feb '20

14 Feb '20

Hi, According to https://kamailio.org/docs/modules/5.3.x/modules/dispatcher.html#dispatcher.…, ds_is_from_list should accept an integer variable for the group id parameter. The following simple example doesn¹t work however: $var(foo) = 5; ds_is_from_list($var(foo)); Kamailio logs ³function ds_is_from_list: parameter 1 is not constant². Is this a bug or have I misinterpreted the documentation somehow? Thanks in advance, Björn Bylander

2 1

Call-ID Uniqueness and safety?
by Mark Boyce 14 Feb '20

14 Feb '20

Hi all Over coffee we’ve been discussing the uniqueness and safety of call-id this morning. Lots of things like RTPEngine recording use call-id to generate filenames (yes it does escape them first :-). Not to mention dialog tracking (also uses from/to tags) etc. Which got us thinking. As call-id is determined by the first SIP packet that normally comes from a UA, should we care if this UA was broken or worse malicious? What would happen if they were repeated? For example we may struggle to determine a recordings owner in RTPEngine (file name format is callid+random.wav). For security should we be rejecting requests where we’ve seen the call-id before? All theoretical over coffee, as it was that or start working! Just wondering what peoples thoughts were? Cheers Mark

4 7

ACC module
by Voip support 13 Feb '20

13 Feb '20

Dear Community, I have setup ACCDB module to save missed / failed and successful calls. My issue is that i am getting duplicated entries for INVITE / BYE in acc database table. For example: id;method;from_tag;to_tag;callid;sip_code;sip_reason;time;src_ip;dst_user;dst_domain;src_user;src_domain;cdr_id 231838;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231839;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231840;INVITE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:28;22.33.44.55;nlb124#78019989897056;11.11.11.11;asterisk;172.16.0.35;0 231841;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 231842;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 231844;BYE;21-7FC1D85;1914472180948750;KKyIdaaJceAu7pzN(a)172.16.0.35;200;OK;2020-02-11 19:16:36;22.33.44.55;;190.196.153.130;asterisk;172.16.0.35;0 Following way: # ----- 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") modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$rd;src_ip=$si") # our lines modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_on_failed", 1) #!endif Then in routing config i set setflag(FLT_ACC FLT_ACCMISSED FLT_ACCFAILED in several places. My question is - each setflag cause an entry to be saved in DB? How can i debug and resolve the issue of duplicates. I am trying to use a CDR script found somewhere on siremis sites to do the accounting but it's making trouble because due to duplicates in acc table i am getting cdrs unique key violations. Secondary question is - if a BYE is not received / lost is there any way to send a check triggered by kamailio to each caller and callee side to check if dialog is still active. I would not like to use session timers (as endpoints need to support it). Best regards, Tom

2 7

URI Mapping in Presence?
by Mark Boyce 12 Feb '20

12 Feb '20

Hi All Looking for the best way to achieve mapping from requested uri to actual uri in presence subscription. The scenario I’ve been presented with is that there is a direct relation between extension number and sip user but they are not the same thing. So for example; Extension 10 is mapped to SIP user sip-SomethingLong Extension 11 is mapped to SIP user sip-SomethingEvenLonger When extension 10 subscribes to 11 we end up with something like this; From: “Extension 10" <sip:sip-SomethingLong@ourpbx.com:5060>;tag=12345678 To: <sip:11@ourpbx.com:5060>;tag=987654321-c8bd I *think* what I need to end up with in presence db is; presently_uri = sip:sip-SomethingEvenLonger@ourpbx.com to_user = 10 (Is that right?) However I can’t see a standard way to achieve this. Manage_subscribe will let me override the watcher uri but not the watched. Anyone see a simple way to achieve this? Thanks Mark

2 2

Generating provisional responses from kamailio instance
by George Diamantopoulos 12 Feb '20

12 Feb '20

Hello all, I would like to use rtpengine's media playback capabilities in an early media scenario. The use case is as follows: * Kamailio receives an incoming INVITE * During INVITE processing, kamailio config script determines that an early media message should be played back and then the call should fail with 4xx * To that effect, Kamailio sends out a locally-generated 183 Progress and instructs rtpengine to playback an audio file * After the audio file has finished playing, kamailio responds with 4xx to the INVITE Is this possible at all? Thanks! BR, George

3 2

Kamailio crashes after Contact expires event route
by Sergiu Pojoga 12 Feb '20

12 Feb '20

Hi there, After below even route is triggered, Kamailio crashes with the following in logs: Feb 12 08:03:59 kamailio[24671]: INFO: <script>: Received HTTP/RPC request from IP=127.0.0.1 Feb 12 08:03:59 kamailio[24671]: WARNING: <script>: Contact myusername(a)sip.mydomain.net has expired Feb 12 08:03:59 kamailio[24671]: INFO: drouting [drouting.c:483]: rpc_reload(): RPC command received! Feb 12 08:03:59 kamailio[24675]: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 59 Feb 12 08:03:59 kamailio[24614]: ALERT: <core> [main.c:745]: handle_sigs(): child process 24671 exited by a signal 11 Feb 12 08:03:59 kamailio[24614]: ALERT: <core> [main.c:748]: handle_sigs(): core was not generated Feb 12 08:03:59 kamailio[24614]: INFO: <core> [main.c:770]: handle_sigs(): terminating due to SIGCHLD Feb 12 08:03:59 kamailio[24675]: INFO: <core> [main.c:825]: sig_usr(): signal 15 received event_route[usrloc:contact-expired] { xlog("L_WARN", "Contact $ulc(exp=>aor) has expired\n"); # Remove the sip: from the front of the Received URI $var(exp_received) = $(ulc(exp=>received){s.substr,4,0}); sql_query("db", "UPDATE `dr_gateways` SET address = '127.0.0.1' WHERE address = '$var(exp_received)'", "rz"); sql_result_free("rz"); jsonrpc_exec('{"jsonrpc": "2.0", "method": "drouting.reload", "id": 1}'); } If I comment out the sql and jsonrpc_exec lines above - crash does not happen. Let me know if core dump is needed. Thanks, --Sergiu

2 1

RTP Engine in docker
by Voip support 12 Feb '20

12 Feb '20

Dear Community, I would like to use rtpengine but had a very hard time to do the compilation under ubuntu 16.04 and 18.04. On Ubuntu 18.04 after already compiled the rtpengine i was unable to install the deb packages. Many different errors occurred. I finally tried to install RTP engine on debian 10 and i was able to install it. I am thinking of 2 use scenarios: - handling many concurrent calls like using rtpproxy for normall traffic - make WebRTC to legacy RTP transcoding (convert WebRTC SDP to legacy SDP to use with non webrtc compliant sip server) For the first scenario i imagine that it would be far better to run in-kernel mode because of performance. For the second scenario i think userspace daemon should be fine ( i expect not much traffic maximum 50-100 calls). However my question is what is the correct order of installing the deb packages. Which packages do i really need. For running rtpengine in docker could i use Debian 10 OS and compile rtpengine and install just " ngcp-rtpengine-daemon_6.2.0.0+0~mr6.2.0.0_amd64.deb" ? (the in this case the host running docker can be any linux distribution?) If i would like to run rtpengine in docker in kernel mode - is it possible or i need to use same linux distribution in host and docker container (because of the kernel match?) Best regards, Tom

5 6

Dispatcher error with rtpptoxy
by Gaurav Bmotra 12 Feb '20

12 Feb '20

hi i m using kamailio 5.2.1 in Kubernetese on Host network it was working file but now the following issue i n getting call connecting but audio both way issue here is the logs that got -------------------------------- 2(14) ERROR: {1 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 4(16) NOTICE: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578636040;method=INVITE;from_tag=88c92018;to_tag=2c3c0a16;call_id=EQEiv5Mjs-eV1OCGv6Potw..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 4(16) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:477]: acc_db_request(): failed to insert into database 6(18) ERROR: {2 2 INVITE EQEiv5Mjs-eV1OCGv6Potw..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) NOTICE: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578636041;method=BYE;from_tag=88c92018;to_tag=2c3c0a16;call_id=EQEiv5Mjs-eV1OCGv6Potw..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 8(20) ERROR: {2 3 BYE EQEiv5Mjs-eV1OCGv6Potw..} acc [acc.c:477]: acc_db_request(): failed to insert into database 3(15) ERROR: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 9(21) NOTICE: acc [acc.c:279]: acc_log_request(): ACC: call missed: timestamp=1578636141;method=INVITE;from_tag=6b032a77;to_tag=;call_id=LH6sikNzpP9rXXMJvH20wQ..;code=408;reason=Request Timeout;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.70.57 9(21) ERROR: db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) 9(21) ERROR: <core> [db_query.c:243]: db_do_insert_cmd(): error while submitting query 9(21) ERROR: acc [acc.c:477]: acc_db_request(): failed to insert into database 9(21) NOTICE: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} <script>: RTF_DISPATCH: INVITE 300 [LH6sikNzpP9rXXMJvH20wQ..] 9(21) NOTICE: {1 2 INVITE LH6sikNzpP9rXXMJvH20wQ..} <script>: RTF_DISPATCH: WARNING sip: 49.14.97.169:63353 is broken and marked as inactive 3(15) ERROR: {1 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) ERROR: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} rtpproxy [rtpproxy.c:2586]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy 8(20) NOTICE: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} acc [acc.c:279]: acc_log_request(): ACC: transaction answered: timestamp=1578638074;method=INVITE;from_tag=b9979f2a;to_tag=c2952a71;call_id=2gY1N8nGCT02Ut7iAEpKWg..;code=200;reason=OK;src_user=200;src_domain=testcalls.aone.one;src_ip=157.39.9.195;dst_ouser=300;dst_user=300;dst_domain=106.78.71.32 8(20) ERROR: {2 2 INVITE 2gY1N8nGCT02Ut7iAEpKWg..} db_mysql [km_dbase.c:126]: db_mysql_submit_query(): driver error on query: Unknown column 'src_user' in 'field list' (1054) ------------------------------- plz help me thank you -- *Regards:* Gaurav Kumar

3 2

Same Via Tag for INVITE and ACK on S/L loadbalancer
by Sebastian Damm 11 Feb '20

11 Feb '20

Hi, I stumbled upon an interop problem with a carrier. We have the following scenario: Gateway --> Loadbalancer --> Carrier The loadbalancer generates a Via header for each request. But since it is stateless, the Via tag is generated for each request. As a consequence, the Via tag in the ACK differs from the one in the INVITE. And one carrier doesn't handle those ACKs if the Via tag differs. Is there a way to force the creation of a "deterministic" Via branch tag? For example, building it from a hash over call-id and from-tag or something like that? Thanks in advance Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

5 10

Different REGISTER expires based on User-Agent
by mayamatakeshi 11 Feb '20

11 Feb '20

Hi, is it possible to reply with a different expires for a REGISTER depending on some condition like the User-Agent? I am thinking maybe I can use: $ulc(profile=>expires) = 600 (but my kamailio dev machine is out of service and I cannot test this right now).

5 8

Kamailio v5.1.10 Released
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, Kamailio SIP Server v5.1.10 stable release is out. This is a maintenance release of the old stable branch, 5.1, that includes fixes since the release of v5.1.9. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.10, to v5.2.x or 5.3.x series. For more details about version 5.1.10 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/02/kamailio-v5-1-10-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is an old stable branch, v5.1.10 being the last planned release out of it, to mark the end of official maintenance. The latest stable branches are now 5.2 and 5.3, at this time with v5.3.2 being the latest stable release. 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 World Conference - April 27-29, 2020 -- www.kamailioworld.com

1 0

Planning the release of v5.1.10
by Daniel-Constantin Mierla 10 Feb '20

10 Feb '20

Hello, I am considering to release Kamailio v5.1.10 on Thursday, Jan 30, 2020, to mark the end of official packaging from branch 5.1. At this moment the last two stable branches are 5.2 and 5.3. If anyone is aware of some commits for fixes that should be backported to branch 5.1, write to sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 2

asterisk kamailio mobile network
by Gintaras Kazakevicius 07 Feb '20

07 Feb '20

Hello, need help. Call from mobile to asterisk: Mobile network provider <trunk> Kamailio <-> Asterisk --- working Call from asterisk to mobile Asterisk <-> Kamailio <trunk> Mobile network provider --- not working How to configure? Maybe you have working example of this solution?

2 1

DIAMETER (server & avp helper)
by Steve Bucklin 07 Feb '20

07 Feb '20

Anyone familiar with: ims_diameter_server cdp cdp_avp I have set Kamailio up to connect with SMSC using Diameter. Initially just want to capture requests in JSON and examine. My trace tells me that I am sending a good message for DIRECT_DEBITING (436). I know the ocs module does not support this - only trying to "take apart" the message. When I get the message, I experience the following: Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: ERROR: cdp [diameter_avp.c:645]: AAAUngroupAVPS(): hss3g_ungroup_avps: source buffer to short!! Cannot read the whole AVP header! Feb 7 13:07:58 ip-172-31-45-179 /sbin/kamailio[18048]: CRITICAL: cdp [diameter_avp.c:702]: AAAUngroupAVPS(): AVP:<> I suspect this is due to vendor, but not experienced here. The docs seem to suggest the 3gpp is supported, and the diameter xml is set for 0 & 10415? Steve

1 0

Broken link on https://www.kamailio.org/wiki/cookbooks/devel/selects
by Björn Bylander 07 Feb '20

07 Feb '20

Hi, In the section "Complete Select List" on X there's a link to http://sip-router.org/docbook/sip-router/branch/master/select_list/select_l… which doesn't exist anymore. Can anyone tell me where the current list is located? Thanks in advance, Björn Bylander

2 1

Best practice to figure out one way rtp delay
by Karsten Horsmann 06 Feb '20

06 Feb '20

Hi List, I have a new setup with two Kamailios installations. One serves siptrunks from the internet and one is for internal routing. So far so good. At the end I have an 3rd party b2bua that receives and sends calls via the both Kamailios. For testing purposes I setup an freeswitch with beep and then echo application to the caller. My call flow are like this Call 1 to kam1 sbc then internal kam2 and b2bua. B2bua make then new call no 2 vice versa to PSTN freeswitch with echo. After this b2bua bridges the calls together. This generates for the caller an beep and echo. The interesting thing is now, that Call 1 gets an hearable delay of 1 second. But only in the rtp steam from me to the caller. The second calls seems equal of timing. Since is a really new complete setup of hardware and stuff, the question of its working before is answered with an no. Now my question to you guys. How can I get an measurable method to finding the delaying parts (could be network, servers, applications etc). Only capturing on one place don't did the trick for me. Thanks for your hints Cheers Karsten

3 4

Binary Websocket messages
by Andre Pires dos Santos 06 Feb '20

06 Feb '20

Hello guys, we are new to kamailio so sorry basic or wrong points. We started using kamailio 5.3.2 and rtpengine as our sip to webrtc solution and SIP.js as webrtc client. We noticed that most of the times webrtc client receives UTF-8 messages. But sometimes webrtc client receives binary messages. When we get binary messages SIP.js cannot parse the message. I couldn't find how to force kamailio to send only UTF-8 messages. Is it possible? We have other issues but I think it will be better for the list to break other points in different messages. Thanks! Santos

2 8

IMS RX AAR issues
by Pafel 06 Feb '20

06 Feb '20

Hello, I am using kamailio 5.3 + rtpengine module + sipwise rtpengine in a IMS test environment. I have and issue with ims_qos module and Rx AAR command. PCSCF is configured with #!define WITH_RX ##!define WITH_RX_REG #!define WITH_RX_CALL I have 2 test phones connected. Both of them use ipv6. PCSCF has ipv4 and ipv6 address. All other nodes - SCSCF, ICSCF use ipv4. Simple scenario UE1 calls UE2. UE2 is replying with 183 session progress and then RX AAR is sent by PCSCF to PCRF/PGW. The problem is that AVPs Media-Sub-Component Flow-Description include ipv4 address of the proxy and ipv6 address of the UE. Example: permit out 17 from 192.168.1.1 34998 to fd18::1:2dca:3303:1286:257d 50010 permit in 17 from fd18::1:2dca:3303:1286:257d 50010 to 192.168.1.1 34998 But this is not acceptable for the PGW (different ip versions). The SDP between UE and PCSCF is rewritten by rtpengine/sipwise. Looking at the ims_qos module I found that the address of the proxy is taken from the session original request SDP media ip/port. Having in mind that rtpengine/sipwise proxy is rewritting it is there any way I can set the correct ipv6 address and port of the proxy? I've tried to get the rewritten media address from SDP (in logs) but without success: if(sdp_get_line_startswith("$avp(cline)", "c=")) { xlog("L_INFO", "SDP media ip: $(avp(cline){s.select,2, })\n"); } if(sdp_get_line_startswith("$avp(mline)", "m=")) { xlog("L_INFO", "SDP media port: $(avp(mline){s.select,1, })\n"); } I am always getting the ipv4 media address and port of the proxy. Any idea how I can set and use the rewritten by the rtpengine/sipwise media ip and proxy for the RX AAR? Thanks in advance, Pavel Siderov

1 1

Re: [SR-Users] Pre-built Debian Package for Kemix Module?
by Yufei Tao 05 Feb '20

05 Feb '20

Hi Daniel and Victor, Thank you very much for the very quick and helpful response, as always! Best regards, Yufei

1 0

Pre-built Debian Package for Kemix Module?
by Yufei Tao 05 Feb '20

05 Feb '20

Hi, I am trying to use KEMI lua that needs kemix module (for KSR.kx.<...>). I wonder if there is any pre-built Debian package that include the kemix module, something like kamailio-<...>-modules, that I can install? Or do I need to build from source with kemix in included_modules? On a related question, for the kamailio-<...>-modules packages how do I find out what modules are included in each of these package? E.g. things like kamailio-lua-modules, kamailio-mysql-modules etc. are easy to guess, while kamailio-nth, kamailio-utils-modules etc. are not so easy. Thank you! Yufei

3 3

config file packaging
by Sergey Safarov 05 Feb '20

05 Feb '20

Hi At present time kamailio config file packaged with kamailio binaries. So projects that use kamailio cannot simple package "/etc/kamailio/kamailio.cfg" file. This leads to packages conflict on RPM dists. As options we can separate "kamailio default config" and "kamailio binaries" to packages. 1) kamailio-cfg-default 2) kamailio-base This allow not install "kamailio-default" config when project provide own config file. For backward compatibility we create "kamailio" empty package that have two dependencies "kamailio-cfg-default" and "kamailio-base". What you think?

2 2

How to implement simple presence in kamailio active-active cluster.
by Zhan Bazarov 05 Feb '20

05 Feb '20

Hello! We have next setup, which I've desribed in: http://sip-router.1086192.n5.nabble.com/Bad-request-when-INVITE-relayed-to-… Presence works fine in setup with single kamailio, without any loaded presence modules... So, I want to something wtih SUBSCRIBE and NOTIFY methods... But when I making a relay on second kamailio instance I receive 407 proxy auth required from this instance... So, its how I'm trying to implement this: part of code from route[LOCATION] case -2: if (is_method("SUBSCRIBE|NOTIFY")) { $ru = "sip:" + $tU + "@" + "ip_of_second_kam:5078"; t_relay("ip_of_second_kam", 5078); exit; } else { send_reply("405", "Method Not Allowed"); exit; } Any ideas how I can do that? Thanks. -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

1 0

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hi Lanvin, (please keep the sr-users list in CC) the modules works on my machine: henning@linux:~/repositories/kamailio> killall kamailio ; ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.188.25 [192.168.188.25]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 192.168.188.25 [192.168.188.25]:5060 Aliases: henning@linux:~/repositories/kamailio> ps aux |grep kamailio henning 15948 0.1 0.1 166776 14900 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d henning 15949 0.0 0.0 166776 4224 ? S 15:33 0:00 ./src/kamailio -f etc/kamailio.cfg -L src/modules/ -Y /tmp -a no -E -e -d [...] Here the diff to the default cfg in git master: diff --git a/etc/kamailio.cfg b/etc/kamailio.cfg index c0e618719..dbc911fb0 100644 --- a/etc/kamailio.cfg +++ b/etc/kamailio.cfg @@ -309,6 +309,14 @@ loadmodule "pike.so" loadmodule "debugger.so" #!endif + +loadmodule "ims_dialog" +loadmodule "ims_usrloc_pcscf" +loadmodule "ims_ipsec_pcscf" +loadmodule "ims_registrar_pcscf" +modparam("ims_dialog", "dlg_flag", 4) + + Cheers, Henning Am 04.02.20 um 11:55 schrieb Lanvin ADON: > > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > #loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > loadmodule "ims_registrar_pcscf" > > > > > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:27 pcscf kamailio[17616]: 0(17618) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:27 pcscf kamailio[17616]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17620]: 0(17622) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17620]: ERROR: error while initializing > modules > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:28 pcscf kamailio[17623]: 0(17625) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:28 pcscf kamailio[17623]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17626]: 0(17628) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17626]: ERROR: error while initializing > modules > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't > bind ims_ipsec_pcscf > Feb 4 11:53:29 pcscf kamailio[17629]: 0(17631) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing module > ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 11:53:29 pcscf kamailio[17629]: ERROR: error while initializing > modules > > > > > > > > > On Tue, Feb 4, 2020 at 11:46 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello Lanvin, > > try to load it before and see if it starts then. > > Cheers, > > Henning > > -- > > Henning Westerholt – https://skalatan.de/blog/ > > Kamailio services – https://gilawa.com <https://gilawa.com/> > > *From:* Lanvin ADON <lanvin.adon(a)gmail.com > <mailto:lanvin.adon@gmail.com>> > *Sent:* Tuesday, February 4, 2020 11:45 AM > *To:* Henning Westerholt <hw(a)skalatan.de <mailto:hw@skalatan.de>> > *Subject:* Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hello Henning > > ims_ipsec_pcscf is loaded after. > > loadmodule "ims_dialog" > loadmodule "ims_usrloc_pcscf" > loadmodule "ims_registrar_pcscf" > #!ifdef WITH_IPSEC > loadmodule "ims_ipsec_pcscf" > #!endif > > Thanks > > Lanvin ADON > > On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de > <mailto:hw@skalatan.de>> wrote: > > Hello, > > just a guess, but do you load the ims_ipsec_pcscf module > before the ims_registrar_pcscf module? > > 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 > *Lanvin ADON > *Sent:* Tuesday, February 4, 2020 8:54 AM > *To:* sr-users(a)lists.kamailio.org > <mailto:sr-users@lists.kamailio.org> > *Subject:* [SR-Users] P-CSCF not starting- Kamailio 5.3 > > Hi All, > > I'm installing an IMS core with Kamailio 5.3 release. The > P-CSCF is not starting with the following error message. > > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> > [core/sr_module.c:638]: find_mod_export_record(): export of > <bind_ims_ipsec_pcscf> not found (flags 0) > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: > ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: > mod_init(): can't bind ims_ipsec_pcscf > Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> > [core/sr_module.c:849]: init_mod(): Error while initializing > module ims_registrar_pcscf > (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) > Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while > initializing modules > > For the configuration file, I'm using the example files in the > source. > > > > -- > > Regards > > Lanvin ADON > > > > -- > > Cordialement, > > Lanvin ADON > > > > -- > > */ > /* > > Cordialement, > > Lanvin ADON -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Re: [SR-Users] P-CSCF not starting- Kamailio 5.3
by Henning Westerholt 04 Feb '20

04 Feb '20

Hello Lanvin, try to load it before and see if it starts then. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Lanvin ADON <lanvin.adon(a)gmail.com> Sent: Tuesday, February 4, 2020 11:45 AM To: Henning Westerholt <hw(a)skalatan.de> Subject: Re: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hello Henning ims_ipsec_pcscf is loaded after. loadmodule "ims_dialog" loadmodule "ims_usrloc_pcscf" loadmodule "ims_registrar_pcscf" #!ifdef WITH_IPSEC loadmodule "ims_ipsec_pcscf" #!endif Thanks Lanvin ADON On Tue, Feb 4, 2020 at 11:06 AM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello, just a guess, but do you load the ims_ipsec_pcscf module before the ims_registrar_pcscf module? 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 Lanvin ADON Sent: Tuesday, February 4, 2020 8:54 AM To: sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] P-CSCF not starting- Kamailio 5.3 Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON -- Cordialement, Lanvin ADON

1 0

P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

2 1

parallel forking in combination with rtpengine
by michael.willow＠web.de 04 Feb '20

04 Feb '20

Hi everbody, I encountered an issue with my Kamailio in combination with Rtpengine. I wondered if anybody got a solution or a hint as this issue is only in combination with special parallel forked calls. I sent a call from my SIP gateway via a Kamailio to another SIP provider. The other SIP provider or the B-user seems to use a parallel forking as I receive 183 Session Progress and 180 Ringing with different to-tags. My kamailio is using a Rtpengine to relay the media from my SIP gateway to the other SIP provider. As Kamailio receives two different to-tags that include a SDP body rtpengine_manage makes Rtpengine to change RTP port. Some short example (cut down to the important messages): a-gateway kamailio b-provider | INVITE Port 16890 | INVITE Port 20400 | | ------------------------> | --------------------------> | | | | | 180 to-tag 1 / no SDP | 180 to-tag 1 / no SDP | | <------------------------ | <------------------------- | | | | | 183 to-tag 2 / Port 20408 | 183 to-tag 2 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20400 ----> Port 46298 | | | | | | | | 180 to-tag 2 / Port 20408 | 180 to-tag 0 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | 200 OK to-tag 1 / Port 20408 | 200 OK to-tag 1 / Port 46298 | | <------------------------ | <------------------------- | | | | | | | | RTP | RTP | | Port 16890 <---- Port 20408 | Port 20400 <---- Port 46298 | | | | | | | | RTP | RTP | | Port 16890 ----> Port 20408 | Port 20418 ----> Port 46298 | Kamailio receives INVITE from a-gateway with rtp port 16890 and rtpengine rewrites it with port 20400 to b-provider. b-provider sends 180 ringing from to-tag 1 and no SDP which is just passed through kamailio to a-gateway. b-provider sends 183 progress from to-tag 2 and SDP with port 46298 to kamailio. rtpengines rewrites the sdp port to 20408 which is passed to a-gateway. b-provider start sending RTP via the negotiated ports which is passed correctly to a-gateway. b-provider sends 180 ringing from to-tag 2 with the same SDP. Rewritten and passed to a-gateway. b-provider sends 200 OK on Invite. This 200 Ok is from to-tag 1 but including the same SDP endpoint as to-tag 2 used before. Media from b-provider via rtpengine to a-gateway is passed the same way as before. Media from a-gateway to rtpengine is passed the same way as before. Media from rtpengine is passed with a different source port to b-provider. As this port was not announced in the callflow b-provider blocks the traffic. Any ideas how to fix this issue or force rtpengine to stay on the same port? Thanks Michael

1 0

P-CSCF not starting- Kamailio 5.3
by Lanvin ADON 04 Feb '20

04 Feb '20

Hi All, I'm installing an IMS core with Kamailio 5.3 release. The P-CSCF is not starting with the following error message. Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Feb 4 07:01:32 pcscf kamailio[13878]: 0(13880) ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/pcscf/lib64/kamailio/modules/ims_registrar_pcscf.so) Feb 4 07:01:32 pcscf kamailio[13878]: ERROR: error while initializing modules For the configuration file, I'm using the example files in the source. -- Regards Lanvin ADON

1 0

Call merging issues on cellular network
by Gaurav Bmotra 03 Feb '20

03 Feb '20

Hi I m using kamailio 5.3 I m trying to merge the call on cellular network but is not working. As on WiFi it is working fine . Any one can help me Thank you

2 1

Kamailio TLS server hello without any client certificate request
by Patrick Murphy 03 Feb '20

03 Feb '20

Hey guys - I'm using the following settings on my Kamailio config: modparam("tls", "low_mem_threshold2", 1024) modparam("tls", "tls_force_run", 1) modparam("tls", "tls_disable_compression", 1) modparam("tls", "ssl_max_send_fragment", 4096) modparam("tls", "renegotiation", 0) modparam("tls", "low_mem_threshold1", 2048) modparam("tls", "require_certificate", 1) When a client attempts to connect to Kamailio, the server hello it receives does *not* contain the client certificate request. And I've noticed that kamailio starts with: tls [tls_domain.c:694]: set_verification(): TLSs<default>: No client certificate required and no checks performed tls [tls_domain.c:320]: fill_missing(): TLSs<default>: require_certificate=0 Complete version output: kamailio 5.1.1 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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 ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. Any hints?

2 2

Dialog module - default_timeout behavior
by Gholamreza Sabery 03 Feb '20

03 Feb '20

Hi, According to the documentation default_timeout is "The default dialog timeout (in seconds), in the absence of a custom value provided in an AVP". There is also a "send_bye" option which can automatically send a BYE message on dialog timeout. Suppose that a user sends an INVITE and the call is successful (200). He talks more than default_timeout seconds (so there is not BYE for more than default_timeout seconds). This means that when the user actually sends BYE, the dialog is already terminated by the dialog module? Regards

3 4

sqlops multiple databases
by Ali Taher 03 Feb '20

03 Feb '20

Hello, Is there a way in sqlops module to use multiple postgres databases based on routing logic ? Here is what I'm using right now: modparam("sqlops","sqlcon","cb=>postgres://postgres:postgres@localhost:5432/db1") Thanks Ali Taher

2 1

Bad request when INVITE relayed to another kamailio
by Zhan Bazarov 31 Jan '20

31 Jan '20

We have Kamailio-cluster via route53(round-robin) some-domain.net we have two kamailio with public IP's phone1 is registered on kam1 phone2 is registered on kam2 when we are calling from phone1 to phone2 callflow looks: phone1 => kam1 => asterisk => kam1 => t_relay(address of second kamailio:5078) => kam2 => phone2 it works perfectly, but in case when we are using polycom as phone2 - we are getting 404 response from polycom... *Invite from second kamailio * 2020/01/20 10:31:21.799327 10.199.240.19:5078 -> 37.17.41.5:49811 INVITE sip:jyu3xsfkrz6c5qn@10.3.0.116;transport=tcp SIP/2.0 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.191:5078;r2=on;lr;nat=yes;rtp=bridge> Record-Route: <sip:10.199.240.135:5078;lr> Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135> Contact: <sip:asterisk@10.199.240.179:7060;alias=10.199.240.179~7060~1> Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 CSeq: 22619 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Supported: timer, replaces, norefersub Session-Expires: 1800 Min-SE: 90 P-Asserted-Identity: "Penny" <sip:1015@10.199.240.179> Max-Forwards: 68 User-Agent: Awesome Calling Platform 3.0 Content-Type: application/sdp Content-Length: 463 *Response from POLYCOM * 2020/01/20 10:31:22.054766 37.17.41.5:49811 -> 10.199.240.19:5078 SIP/2.0 400 Bad Request Via: SIP/2.0/TCP some-domain.net:5078;branch=z9hG4bK9ca3.93c2345f3eb1d4b0e1244e722a9bfb6e.0 Via: SIP/2.0/UDP some-domain.net:5078;rport=5078;received=10.199.240.135;branch=z9hG4bK9ca3.89b66f1dd6e86a5922180b8ed8475072.0 Via: SIP/2.0/UDP 10.199.240.179:7060;received=10.199.240.179;rport=7060;branch=z9hG4bKPj269365e0-798d-404b-bb77-2ad78472905c From: "Penny" <sip:1015@10.199.240.179>;tag=ba402508-a640-409f-ba30-dffdfe499f43 To: <sip:jyu3xsfkrz6c5qn@10.199.240.135>;tag=8BC58304-83D9B045 CSeq: 22619 INVITE Call-ID: c3a406ca-9ac7-423d-9697-06d0603f48d5 Record-Route: <sip:10.199.240.19:5078;transport=tcp;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.19:5078;r2=on;lr;nat=yes;rtp=bridge>, <sip:10.199.240.235:5078;lr> User-Agent: PolycomVVX-VVX_450-UA/5.8.0.12848 Accept-Language: en Content-Length: 0 Any ideas how to fix it? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

3 6

$vn vs $var
by Alex Balashov 31 Jan '20

31 Jan '20

Hi, I must have missed the discussion about it somewhere, but what is the inspiration for the new $vn variable container? According to the documentation, the only difference is that it can hold $null values, whereas variables initialised in $var to $null are coerced into an integer value of 0. But why was this a concern? Why is it necessary? Is it not confusing to users? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

rtpengine relay ip and port in avp
by Pafel 31 Jan '20

31 Jan '20

Thanks for the info Yuri! I tried several scenarios but without success. My sip proxy has ipv4 and ipv6 address. The other nodes use ipv4. When there is a request for a ipv6 UAC/customer comming to the proxy the rtpengine/sipwise is rewriting the sdp with the correct ipv6 address of the proxy. So I am trying to get the ipv6 address of the proxy from the sdp (which is always the same ipv6 address) and the port (which is always different). I've enabled modparam("rtpengine", "write_sdp_pv", "$avp(sdp)") And I have in request section: rtpengine_manage($avp(rtpproxy_offer_flags)); set_body("$avp(sdp)", "application/sdp"); xlog("L_INFO", "Request SDP $avp(sdp))\n"); And the problem is that I am always getting the ipv4 address and port but not the rewritten by the proxy ipv6 address and port. Any idea? Regards, Pavel

1 0

Re: [SR-Users] modifying via header from public IP to private IP
by vinod mn 31 Jan '20

31 Jan '20

Thank's david in my kamailio configuration I am using *listen=udp:192.168.131.69:5060 <http://192.168.131.69:5060/> advertise 221.20.20.12:5060 <http://221.20.20.12:5060/>* (as mentioned by you) and when the kamailio sends an invite to sip-gateway I am seeing this sip-gateway ip is *192.168.132.67 * and this gateway don't have any public-ip assigned .. Via: *SIP/2.0/UDP 221.20.20.12:5060 <http://221.20.20.12:5060/>;* branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 I want private ip instead of public ip in the first via -- Thanks and regards Vinod.M.N

3 4

kamailio
by Genildo Nascimento 30 Jan '20

30 Jan '20

Someone managed installed kamailio with IPV6

4 5

modifying via header from public IP to private IP
by vinod mn 30 Jan '20

30 Jan '20

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. Please let me know, is it possible to replace. [image: image.png] the via headers that kamailio is sending to sip-gateway is sip-phone ip:192.168.113.112 Via: SIP/2.0/UDP *221.20.20.12:5060 <http://221.20.20.12:5060>* ;branch=z9hG4bKaa6a.14c65f5675077ae7b0df70b02516979b.0 Via: SIP/2.0/UDP 192.168.113.112 :50386;received= 192.168.113.112 ;branch=z9hG4bK-d87543-550b984016109617-1--d87543-;rport=50386 how to modify the first via header to have private-IP instead of public-IP.. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

2 1

How to make a distinction between sip and tel uri
by Duarte Rocha 29 Jan '20

29 Jan '20

Greetings, I have a client that sends me two P-Asserted-Identity headers, one with SIP URI and one with tel URI. The pseudo variable $ai returns the topmost P-Asserted-Identity header and I only need the one with the SIP URI. How should i proceed ? Is there a way to delete the one with tel URI? the header removal functions i find only work with header name. I could also work with something different than $ai for this scenario. Best Regards,

2 1

Re-write contact header
by egemen ulus 29 Jan '20

29 Jan '20

Hi everyone, I have asterisk farm and kamailio which located front of asterisks farm as a proxy. When someone send a invite from outside to my kamailio (without record route), i forward it to my Asterisk farm. Then My one of Asterisk reply it as 200 ok. Then my kamailio forward this 200 ok to remote side with record route. But in my 200 ok message, contact header contains my Asterisk IP address. That is why remote side tries to send ACK to Asterisk. How to rewrite my contact header as my kamailio (SBC) address ? Actually I tried fixed nated contact and advertise ip but I couldn't make it worked. I am really new on that. Kamailio cfg file is attached. Please help! Egemen Get Outlook for Android<https://aka.ms/ghei36>

3 4

Kamailio TLS
by Himala Gamage 28 Jan '20

28 Jan '20

Hello All, Kamailio newbie here, I'm trying to set up TLS on Kamailio and having some issues, Kamailio Running on Debian 10. This what I get when trying to install TLS apt-get install -y kamailio-tls-modules The following packages have unmet dependencies: kamailio-tls-modules : Depends: libcurl3 (>= 7.16.2) but it is not installable E: Unable to correct problems, you have held broken packages. Can I install the TLS package with libcurl4? If not, any other workaround? Thanks, In advance Himala

2 4

SIP Redirect CANCEL handling
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I'm using Kamailio as sip redirect server where SBC forwards the calls to Kamailio which sends the routing back to SBC in 3xx response. Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the scenario: the client sends CANCEL request to SBC which responds with 200 OK and 487 Request Terminated and then sends CANCEL request to Kamailio. I'm not sure how Kamailio should handle the CANCEL request here, should it send only 200 OK back to SBC or also should send 487 request terminated? Regards, Ali Taher

3 7

get first occurrence value of a param in a string
by Ali Taher 28 Jan '20

28 Jan '20

Hello, I need to get the value of first tgrp param in this string : $var(x) = <sip:xxxxxxxxx;tgrp=0001000;trunk-context=xx.xx.xx.xx@xx.xx.xx.xx:5060;transport=UDP;user=phone;tgrp=237> I'm trying $var(y) = $(var(x){param.value,tgrp}); but it's getting the last value of tgrp which is "237>". Noting that first tgrp is not always in the second index , some other parameters can be added to the string. How to get the value of first occurrence of tgrp param ? Regards, Ali Taher

1 0

centos 7 repo is broken
by Aidar Kamalov 27 Jan '20

27 Jan '20

Hello # yum install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirror.reconn.ru * extras: mirror.docker.ru * nux-dextop: mirror.li.nux.ro * updates: dedic.sh Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.6-7.el7.centos will be installed --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Processing Dependency: /usr/bin/python23 for package: kamailio-5.2.6-7.el7.centos.x86_64 --> Finished Dependency Resolution Error: Package: kamailio-5.2.6-7.el7.centos.x86_64 (kamailio) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest -- Aydar A. Kamalov

2 1

Kamailio on Docker
by Jack R 27 Jan '20

27 Jan '20

Hi Please advise have you tried deploying Kamailio (just plain SIP server) with mysql on Docker. I was trying the same, using Kamailio & mysql images on https://hub.docker.com however there are two issues that I was facing: 1. I ran Wireshark on server & inside container I am getting 403 Not relaying and SIP client MircoSIP does not connect. Request reaches the server and the container however it gives 403 Not relaying. I thought to print myuri value from Kamailio.cfg, to understand the issue better however then comes the second issue as below: 2. This is a bigger issue: I am not able to generate any logs -no syslogs, no docker logs (very limited logs get generated). Hence not able to troubleshoot anything. Please advise and help. Thanks and Regards, Jack.

2 1

Kamailio/RTC dinner at Fosdem 2020
by Daniel-Constantin Mierla 27 Jan '20

27 Jan '20

Hello, Fosdem 2020 is approaching, there will be a talk from Henning about Kamailio, many other friends and related projects are presenting in the RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, Homer, Janus, Jitsi, CGRateS, Linphone, ...): * https://fosdem.org/2020/schedule/track/real_time_communications/ I plan to be at the event and wondering if there are enough interested participants for having the traditional Kamailio/RTC dinner. If yes, Torrey will help to book a place to accommodate us (first choice could be the same restaurant like the last year, if available). Reply if you want to join the dinner and say how many other people are joining you. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

6 6

CSeq values in dialog OPTIONS keepalives
by Alex Balashov 24 Jan '20

24 Jan '20

Hi, I'd like to reopen an old thread of discussion: Using Kamailio 4.4.x latest, when I enable dialog OPTIONS keepalives on both sides of a call - ka-dst and ka-src dialog params, that is - I get OPTIONS pings sent to one side with a CSeq value like this: CSeq: 0 OPTIONS and the other side: CSeq: (CSeq of e2e ACK - 1) OPTIONS The 0 CSeq is acceptable to most UAs--at least, among the small number I've tested--and they answer 200 OK, but the second one results in a '500 Server Internal Error'. I have read the prior literature on this: https://lists.kamailio.org/pipermail/sr-users/2012-May/073069.html https://lists.kamailio.org/pipermail/sr-users/2018-April/101096.html and I (mostly) understand the rationale of keeping this mechanism stateless. I tried turning on `track_cseq_updates` to see if it might change the behaviour, but it does not. Anyway, a mechanism that deliberately elicits 500 responses from the endpoint is going to raise some eyebrows and just looks a bit unprofessional. I'm wondering if there is a better way to go. My questions are: 1. While I understand the reasoning here: "it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept)." Would it be such a problem to use a CSeq value that is (last highest known CSeq observed) + ($RANDOM % LARGE_VALUE)? 2. In the case that `ka-dst` and `ka-src` are both enabled, why is there an inconsistency in the behaviour with respect to the upstream and downstream side (CSeq value of 0 vs. CSeq value of <= CSeq(ACK))? 3. At the risk of inviting some baroque state-keeping that is runtime-dependent, could there be an implementation where the CSeq of genuine in-dialog requests from the UA is modified in-flight by Kamailio, taking advantage of its being in the middle of in-dialog requests, to the appropriate next highest value? I ask this because, if I understood the limited documentation for `track_cseq_updates` correctly, https://kamailio.org/docs/modules/5.3.x/modules/dialog.html#dialog.p.track_… "Enable the callbacks for tracking if CSeq number needs to be updated. It is the case when the INVITE has to be authenticated to downstream provider using uac_auth() from uac module. This is done only for requests in downstream direction. The CSeq difference is stored in $dlg_var(cseq_diff), be sure this variable is not overwritten via config operation." it seems like the door to this method might already be open? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 6

TLS and TCP port question
by Cyrille Demaret 24 Jan '20

24 Jan '20

Hi, Im trying to implement kamailio as a Teams SBC as described here : <https://skalatan.de/en/blog/kamailio-sbc-teams> https://skalatan.de/en/blog/kamailio-sbc-teams (thanks to the author, I think hes here) Im using Kamailio 5.2.1 and I still have a problem to have the SBC active on the Microsoft side. Im correctly receiving the OPTIONS packet from Microsoft but I have some doubt on my OK reply. Heres the dialog captured in sngrep using the siptrace module : The OPTIONS packet I receive from Microsoft contains a Contact header with a different port as the original packet. So, Kamailio send the 200 OK back to this port. However, when I capture packets using tcpdump, I can see that theres no connection to 52.114.76.76:5061 : 15:01:35.466926 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 1, win 2053, length 0 15:01:35.467383 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 1:189, ack 1, win 2053, length 188 15:01:35.467423 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], ack 189, win 237, length 0 15:01:35.478535 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [.], seq 1:2881, ack 189, win 237, length 2880 15:01:35.478771 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 2881:3017, ack 189, win 237, length 136 15:01:35.498028 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 2881, win 2053, length 0 15:01:35.551569 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3017, win 2052, length 0 15:01:35.765965 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 189:282, ack 3017, win 2052, length 93 15:01:35.767719 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3017:3259, ack 282, win 237, length 242 15:01:35.800634 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [P.], seq 282:825, ack 3259, win 2051, length 543 15:01:35.802362 IP MY_IP.sip-tls > 52.114.76.76.3008: Flags [P.], seq 3259:3656, ack 825, win 245, length 397 15:01:35.863823 IP 52.114.76.76.3008 > MY_IP.sip-tls: Flags [.], ack 3656, win 2050, length 0 If I check the Kamailio log file : Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=613 a=26 n=sl_send_reply Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/msg_translator.c:162]: check_via_address(): (52.114.76.76, 52.114.76.76, 0) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} sl [sl_funcs.c:500]: sl_run_callbacks(): execute callback for event type 1 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace.c:1364]: trace_sl_onreply_out(): trace off... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2225]: tcpconn_send_put(): send from reader (17837 (25)), reusing fd Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2460]: tcpconn_do_send(): sending... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2494]: tcpconn_do_send(): after real write: c= 0x7f81740edb30 n=397 fd=8 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/tcp_main.c:2495]: tcpconn_do_send(): buf= Jan 24 15:01:35 mysbc kamailio[17788]: #027#003#003#001...M...t.[.._..gu..-3.8D#027#022..J.>..a..H.AË.. ..)om.Oh.O.1.A..'.Fa...c.9..M}...#004O..#004 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:498]: pipport2su(): the port string is 5061 Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/proxy.c:264]: mk_proxy(): doing DNS lookup... Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} siptrace [siptrace_hep.c:302]: trace_send_hep2_duplicate(): setting up the socket_info Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) exec: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=614 a=2 n=exit Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:353]: receive_msg(): request-route executed in: 1018 usec Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) Jan 24 15:01:35 mysbc kamailio[17788]: 25(17837) DEBUG: {1 1 OPTIONS fcde6307-1d3b-42fd-91bd-4547744fa00c} <core> [core/receive.c:457]: receive_msg(): cleaning up So, it seems that Kamailio is reusing the existing tls connection and then sends my 200 OK to the port 3008 and not to the port 5061 as asked in the contact header. Is my understanding ok? Is that a normal behavior? Thanks for your help. Best regards, Cyrille

1 1

Kamailio v5.2.6 released
by Henning Westerholt 24 Jan '20

24 Jan '20

Hello, Kamailio SIP Server v5.2.6 stable release is out. This is a maintenance release of the previous stable branch, 5.2, that includes fixes since the release of v5.2.5. There are no changes necessary to database schema or configuration language structure if you are upgrade from an earlier installation of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.6. For more details about version 5.2.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-2-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/> Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Implicit branch creation and private variables questions
by George Diamantopoulos 24 Jan '20

24 Jan '20

Hello all, I'm trying to accomplish the following scenario, and some questions have been raised during testing: I'd like to be able to fork serially to a number of downstream destinations in case of failure, but also try several hosts which are available per destination network before failing over to the next one. To that end, I'm using something similar to the following: request_route { ... $vn(some_var) = "some_value"; $vn(some_other_var) = "some_other_value"; ... $avp(provider_order) = "last_provider"; $avp(provider_order) = "first_provider"; ... t_set_fr(120000, 2000); route(PROVIDER_SELECTION); ... } route[PROVIDER_SELECTION] { if ( is_avp_set("$avp(provider_order)") ) { t_on_failure("PROVIDER_SERIAL"); if ( $avp(provider_priority) == "first_provider" )

4 6

Parallel forking and rtpengine handling
by Sebastian Damm 24 Jan '20

24 Jan '20

Hi, our scenario is the following: We have two clients registered to our Kamailio server, one with a TLS capable phone, one via websocket. Now, when a call comes in, the call is forked and is sent out to both clients. rtpengine handling is done in the branch route, so there are two offers, and we use the "via-branch" parameter. Now, when one branch answers the call, what happens to the other branch? I there a way to delete the other branch? How and in which route? Or does Kamailio do this automatically? I'm happy for a hint. Regards, Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH

5 8

Re: [SR-Users] TLS verification error LetsEncrypt
by Social Boh 24 Jan '20

24 Jan '20

I'm not sure but with let's encrypt you can create only server certificate, not client certificate so you can't require and verify client certificate. Regards --- I'm SoCIaL, MayBe El 24/01/2020 a las 09:01, Bugaian A. Vitalie escribió: > Ok, thanks. > > But my question is still about why verification fails/or what should > be chked to make it work. Not how to disable it. > > Thanks. > > Vitalie. > > On Fri, Jan 24, 2020 at 2:54 PM Social Boh <social(a)bohboh.info > <mailto:social@bohboh.info>> wrote: > > Hello, > > changing: > > [client:default] > #method = TLSv1.2+ > verify_certificate = yes > > require_certificate = yes > > with > > [client:default] > #method = TLSv1.2+ > verify_certificate = no > require_certificate = no > > --- > I'm SoCIaL, MayBe > > El 24/01/2020 a las 08:46, Bugaian A. Vitalie escribió: >> Hello list, >> >> I have tried to setup my tls config tish LetsEncrypt following >> this post: >> >> https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ >> >> My tls config looks like this: >> >> >> [server:default] >> method = TLSv1.2+ >> verify_certificate = no >> require_certificate = no >> private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem >> certificate = >> /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem >> ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> server_name = sbc.example.net <http://sbc.example.net> >> server_id = sbc.example.net <http://sbc.example.net> >> >> #ca_list = /usr/local/etc/fullchain.pem >> #ca_list = /usr/local/etc/kamailio/tls/cacert.pem >> #crl = /usr/local/etc/kamailio/tls/crl.pem >> >> >> # --- >> # This is the default client domain profile. >> # Settings in this domain will be used for all outgoing >> # TLS connections that do not match any other >> # client domain in this configuration file. >> # We require that servers present valid certificate. >> # >> [client:default] >> #method = TLSv1.2+ >> verify_certificate = yes >> require_certificate = yes >> >> =================================== >> My ca_list has all certificates from >> cat /etc/ssl/certs/ca-certificates.crt >> >> /etc/letsencrypt/live/sbcc.example.net/ca_list.pem >> <http://sbcc.example.net/ca_list.pem> >> >> I keep getting certificate validation failed see bellow: >> >> an 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6371]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: >> tls_err_ret(): TLS write:error:1416F086:SSL >> routines:tls_process_server_certificate:certificate verify failed >> Jan 24 08:39:56 sbc.example.net <http://sbc.example.net> >> /usr/local/sbin/kamailio[6370]: ERROR: <core> >> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: >> error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) >> >> ===================== >> >> What params should I change or where to look for a solution on >> this one? >> >> Thanks. >> >> Vitalie A. Bugaian. >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org> >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >

2 1

TLS verification error LetsEncrypt
by Bugaian A. Vitalie 24 Jan '20

24 Jan '20

Hello list, I have tried to setup my tls config tish LetsEncrypt following this post: https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ My tls config looks like this: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /etc/letsencrypt/live/sbc.example.net-0001/privkey.pem certificate = /etc/letsencrypt/live/sbc.example.net-0001/fullchain.pem ca_list = /etc/letsencrypt/live/sbc.example.net-0001/ca_list.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem server_name = sbc.example.net server_id = sbc.example.net #ca_list = /usr/local/etc/fullchain.pem #ca_list = /usr/local/etc/kamailio/tls/cacert.pem #crl = /usr/local/etc/kamailio/tls/crl.pem # --- # This is the default client domain profile. # Settings in this domain will be used for all outgoing # TLS connections that do not match any other # client domain in this configuration file. # We require that servers present valid certificate. # [client:default] #method = TLSv1.2+ verify_certificate = yes require_certificate = yes =================================== My ca_list has all certificates from cat /etc/ssl/certs/ca-certificates.crt >> /etc/letsencrypt/live/ sbcc.example.net/ca_list.pem I keep getting certificate validation failed see bellow: an 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6371]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474421f68 r: 0x7f0474422028 (-1) Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS write:error:1416F086:SSL routines:tls_process_server_certificate:certificate verify failed Jan 24 08:39:56 sbc.example.net /usr/local/sbin/kamailio[6370]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f0474401cb8 r: 0x7f0474401d78 (-1) ===================== What params should I change or where to look for a solution on this one? Thanks. Vitalie A. Bugaian.

1 0

Releasing Kamailio v5.2.6
by Henning Westerholt 24 Jan '20

24 Jan '20

Hello, after the release of Kamailio 5.3.2 it is time to also release another minor version for the branch 5.2. I plan to release Kamailio 5.2.6 next Friday, 24th January 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Cheers, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 1

Ideas to reject a call due to RTPEngine failure when handling the 200Ok
by Patrick Wakano 23 Jan '20

23 Jan '20

Hello list, Hope you all doing well! I have a setup of Kamailio + RTPEngine (so no B2BUA), and I've come to a situation which I can't really figure out the best decision..... The case is a late SDP negotiation that the 200Ok with the SDP offer fails to engage the RTPEngine (for example no RTPEngine available). Also, the same problem can happen with a normal SDP negotiation, in case the INVITE goes through normally but then when engaging the SDP answer for the 200Ok, the RTPengine answer request fails for any reason... In these cases, I can't generate a proper SDP for the 200OOk so the call will connect normally, but the media path will be broken... So for such cases, I am thinking it is reasonable to fail the call. However, failing a call during the 200Ok stage is tricky, at this point we can't convert the 200Ok to a 50X. I tried and Kamailio doesn't allow and also RFC wise this is a no no for a SIP proxy.... I could completely remove the SDP body and forward the 200Ok. This will force a failed SDP negotiation and leave it for the endpoints the task to terminate the call. Is that acceptable? Has anyone faced such situation? Any idea is much appreciated! Kind regards, Patrick Wakano

2 2

uac_replace_from and uac_auth fails to authenticate.
by Kjeld Flarup 23 Jan '20

23 Jan '20

I have a setup where I have a fallback to a GSM number I look up the GSM number and provider information in a database and sets the headers. dlg_manage(); $du = "sip:" + $dbr(ra=>[0,0]); $tu = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); $ru = "sip:"+$rU+"@"+$dbr(ra=>[0,0]); uac_replace_from("sip:"+$dbr(ra=>[0,1])+"@EXTERNALIP"); After this the call goes to a failure_route to do uac_auth() Now my problem is that this works with the providers Asterisk server. But if the call is send to the providers Kamailio server, authentication is rejected. Removing uac_replace_from makes the call accepted on the Kamailio server The only possible problem I can see is that the first INVITE without authentication, has correct From header. But the second with the nonce and auth, uses the wrong From header. Thus two different From headers in the same SIP dialog. Unfortunately uac_replace_from is not allowed in failure_route, so I could test if this is the problem. Is the two different From headers a problem, and how could that be fixed. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

5 6

ims_ipsec_pcscf mandatory for ims_registrar_pcscf ?
by Hamid Hashmi 23 Jan '20

23 Jan '20

According to Module documentation ims_ipsec_pcscf is not mandatory [cid:9c92ff73-4978-4837-b541-a38e15525140] but mod: ims_registrar_pcscf is giving error on ims_ipsec_pcscf. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:253]: mod_init(): Successfully bound to TM module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:632]: find_mod_export_record(): found export of <ul_bind_ims_usrloc_pcscf> in module ims_usrloc_pcscf [/opt/lib64/kamailio/modules/ims_usrloc_pcscf.so] Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:264]: mod_init(): Successfully bound to PCSCF Usrloc module Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/sr_module.c:638]: find_mod_export_record(): export of <bind_ims_ipsec_pcscf> not found (flags 0) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: ims_registrar_pcscf [ims_registrar_pcscf_mod.c:268]: mod_init(): can't bind ims_ipsec_pcscf Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: ERROR: <core> [core/sr_module.c:849]: init_mod(): Error while initializing module ims_registrar_pcscf (/opt/lib64/kamailio/modules/ims_registrar_pcscf.so) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: cdp_avp [cdp_avp_mod.c:225]: cdp_avp_destroy(): Destroying module cdp_avp Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: WARNING: <core> [core/mem/q_malloc.c:487]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tcpops [tcpops_mod.c:156]: mod_destroy(): TCP keepalive module unloaded. Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:84]: tm_shutdown(): start Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:87]: tm_shutdown(): emptying hash table Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:89]: tm_shutdown(): removing semaphores Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:91]: tm_shutdown(): destroying tmcb lists Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: tm [t_funcs.c:94]: tm_shutdown(): done Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/shm.c:288]: shm_destroy_manager(): destroying memory manager: q_malloc Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/q_malloc.c:1185]: qm_shm_lock_destroy(): destroying the shared memory lock Jan 3 02:19:09 CSCFNode2 [Proxy-CSCF][833]: DEBUG: <core> [core/mem/pkg.c:97]: pkg_destroy_manager(): destroying memory manager: q_malloc There is no option to use ims_registrar_pcscf without ims_ipsec_pcscf. [cid:ec21fdae-7d7e-437c-a7ea-6345f36a1428] Regards Hamid R. Hashmi

2 3

rtpengine relay ip and port in avp
by Pafel 23 Jan '20

23 Jan '20

Hello, I am using kamailio 5.3, rtproxy module and sipwise rtpengine. Is there any way I can get the rewritten / relay address and port by the rtpproxy/rtpengine. In avp or some other variable? Regards, Pavel Siderov

2 2

Param.valueat not returning special chars
by Duarte Rocha 21 Jan '20

21 Jan '20

Hello, I've run into a problem while using parameters parsing tools from Kamailio. When i request the value of a parameter it discards some characters like "" for example. Here is an example : $var(Parameters) = "kamailio=\"version5\";sip=hey"; xerr("String Parameter = $var(Parameters)"); xerr("Parameter on position 1 = $(var(Parameters){param.name,0}) and has value = $(var(Parameters){param.valueat,0})"); This code returns : ERROR: <script>: String Parameter = kamailio="version5";sip=hey ERROR: <script>: Parameter on position 0 = kamailio and has value = version5 Is there anything i can do to get the original string? I'm running Kamailio 5.2.0. Best Regards, Duarte Rocha

1 0

Printing xavp
by Ovidiu Sas 21 Jan '20

21 Jan '20

What's the best way to print the entire content of an xavp? The xavp_params_implode() doesn't print the entire stack of an xavp, only the top layer: The following code produces "c=d;e=f;" $xavp(x=>e) = "f"; $xavp(x[0]=>c) = "d"; $xavp(x[1]=>e) = "z"; $xavp(x[1]=>c) = "y"; xavp_params_implode("x", "$var(out)"); Is there a way to print everything inside $xavp(x)? Thanks, Ovidiu

3 10

Re: [SR-Users] Get destination IP from forwarded invite
by CUMHUR KARAHAN 21 Jan '20

21 Jan '20

Hi Joel, Thank you for your response. In fact, I want to get Request-line URI's parameters of the outgoing INVITE that is leaving the first Kamailio box, briefly. Unfortunately "du - destination URL" or "sndto(name)" pseudo-variables indicate the next hope's information. And in this case, the next hope's information is not same as the ClientB's information. As I tried to tell before, our setup is like that: ClientA -> Kamailio1 -> Kamailio2 -> ClientB I need the request line URI parameters of the invite message sent by Kamailio1 to Kamailio2. And these parameters contain ClientB's ip and port information such as 10.252.180.51:36945 as you can see below: INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE Call-ID: 1wF-vjzGIX ... And, your question: no, we don't use the dispatcher mechanism. We just check the DB to find out Kamailio server which the number registered into and send the messages to this server basically. Thanks for all... -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of sr-users-request(a)lists.kamailio.org Sent: 17 Ocak 2020 Cuma 14:00 To: sr-users(a)lists.kamailio.org Subject: sr-users Digest, Vol 176, Issue 17 Send sr-users mailing list submissions to sr-users(a)lists.kamailio.org To subscribe or unsubscribe via the World Wide Web, visit https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users or, via email, send a message with subject or body 'help' to sr-users-request(a)lists.kamailio.org You can reach the person managing the list at sr-users-owner(a)lists.kamailio.org When replying, please edit your Subject line so it is more specific than "Re: Contents of sr-users digest..." Today's Topics: 1. Get destination IP from forwarded invite (CUMHUR KARAHAN) 2. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 3. Re: Get destination IP from forwarded invite (Joel Serrano) 4. Re: Kamalio + RTP Engine + SIP Client behind NAT (Nuno Miguel Reis) 5. Re: Kamailio/RTC dinner at Fosdem 2020 (Federico Santulli) 6. Re: http_client interface (Daniel-Constantin Mierla) 7. Re: Kamailio stop to process incoming SIP traffic via TCP. (adeykunov) 8. Question about xhttp module kamailio 5.2.x (Jimmy Rodley) 9. Trouble with 302 Redirect (Jim Leahy) 10. Re: Kamalio + RTP Engine + SIP Client behind NAT (Richard Fuchs) 11. Re: http_client interface (Fred Posner) 12. Re: Trouble with 302 Redirect (Alex Balashov) 13. UCaaS Platform (Dario Vargas) 14. Re: [commercial announcement] (Henning Westerholt) ---------------------------------------------------------------------- Message: 1 Date: Thu, 16 Jan 2020 11:45:15 +0000 From: CUMHUR KARAHAN <cumhur.karahan(a)consultant.turkcell.com.tr> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Get destination IP from forwarded invite Message-ID: <319901a90d17413292ad249eb56df2ec(a)consultant.turkcell.com.tr> Content-Type: text/plain; charset="utf-8" Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: image001.gif URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/37a3c32b/…> ------------------------------ Message: 2 Date: Thu, 16 Jan 2020 08:23:21 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <4686712f-d865-c6c9-11ac-649966b867b1(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 15/01/2020 13.39, Nuno Miguel Reis wrote: > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 <http://10.10.10.10:5000> when the call establishes > with a public server running kamailio + rtpengine, the actual RTP > arrives from the home router public IP on a natted port, let's admit > it's 100.100.100.100:65100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100%3a65100&umid=39581E1E-9C53-E505-96A6-E057F22 > BD18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-193c6dfa2da7d1ac0c > 93d1a002e73781a0a22d9e>, event though, RTPENGINE assumes that RTP > stream is coming from > 100.100.100.100 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f100.100.100.100&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&au > th=cbd2d006abdb2241308511f72807fe0bc6594b4a-9e365df6d282d7bf1e79e91e70 > 73c5a94aed6d82>:5000.I'm using the '--sip-source' with RTPENGINE to > make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. Rtpengine does this automatically, unless the `asymmetric` flag is used. You should see a log message about a `confirmed peer address` as soon as the first RTP is received on a port. Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/483ec5d0/…> ------------------------------ Message: 3 Date: Thu, 16 Jan 2020 07:23:28 -0800 From: Joel Serrano <joel(a)textplus.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Get destination IP from forwarded invite Message-ID: <CAMtXxQmmRb-ieA9T+0ofxAxiOd23w_0nHos=nZOy9-JNRahnfA(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hello, If I understand correctly your email, you want the IP/Host of the destination of the outgoing INVITE that is leaving a Kamailio box? First of all, how are you forwarding the request? Are you using dispatcher module? Here are a couple of options: 1- https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#du_-_destinat… (although this depends on your config, it might not be set) 2- https://www.kamailio.org/wiki/cookbooks/5.3.x/core#onsend_route + https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#sndto_name If you can describe a little more how you are doing things we might be able to give you more information. Joel. On Thu, Jan 16, 2020 at 3:53 AM CUMHUR KARAHAN < cumhur.karahan(a)consultant.turkcell.com.tr> wrote: > Hi all, > > > > I’m a newbie for Kamailio and working on a feature needs end-users’ > destination IP in the first transferred/forwarded invite message. > > > > Basically, our system/call flow is like: > > ClientA -> Kamailio1 -> Kamailio2 -> ClientB > > > > Invite from the clientA: > > > > *tls:212.252.34.165:32980 > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2f212.252.34.165%3a32980&umid=39581E1E-9C53-E505-96A6-E057F22B > D18D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-d4b503e38b32fd4dee8 > 59434f30f8c5f3e832496>* > > *tls:172.21.193.100:5061 <http://172.21.193.100:5061>* > > *INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 > <http://sip:905364167881@testsip.tvoip.turkcell.com.tr:443> SIP/2.0* > > *Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 20 INVITE* > > *…* > > > > Needed Ip (Client B’s IP 10.252.180.51 for this case) is in the INVITE > row of the forwarded message. > > > > Invite to the Kamailio2 : > > > > *udp:172.21.193.100:5060 <http://172.21.193.100:5060>* > > *udp:172.21.193.98:5060 <http://172.21.193.98:5060>* > > *INVITE sip:905364167881@**10.252.180.51**:36945;transport=tls > SIP/2.0* > > *Record-Route: > <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes>* > > *Record-Route: > <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f > 5.f572;nat=yes>* > > *Via: SIP/2.0/UDP > gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0 > 593548557.1;i=f* > > *Via: SIP/2.0/TLS > 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;r > port=32980* > > *From: <sip:905368601286@testsip.tvoip.turkcell.com.tr > <sip%3A905368601286(a)testsip.tvoip.turkcell.com.tr>>;tag=WbINy0ypx* > > *To: sip:905364167881@testsip.tvoip.turkcell.com.tr > <sip%3A905364167881(a)testsip.tvoip.turkcell.com.tr>* > > *CSeq: 21 INVITE* > > *…* > > > > When the first invite message receives, the dialog variables or > sel(msg…) parsed values belong to this first received message, as far > as I understand during the tests. However I need the forwarded invite message’s variables. > Even after t_relay in RELAY route, it looks the message I have is the > first one. So, I cannot reach this later IP in the forwarded invite > message. As far as I understand, bear with me here please, this IP > manipulation executes in t_relay function and in the C side of > Kamailio. So, I cannot find a way to do that in the our native scripts. > > > > Is there any way to get it without diving C code’s abyss? Do you have > any idea about it? I would appreciate it, if I can get any help or any clue. > > > > Thanks for all... > > > > S. Cumhur Karahan > > > > > <https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=htt > p%3a%2f%2fturkcell.li%2fncfYh&umid=39581E1E-9C53-E505-96A6-E057F22BD18 > D&auth=cbd2d006abdb2241308511f72807fe0bc6594b4a-bbcefdc4443d64fef54812 > 6bd3c5f932d722a0a4> > > Bu elektronik posta ve onunla iletilen butun dosyalar sadece > gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel > kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu > elektronik postanin icerigini aciklamaniz, kopyalamaniz, > yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. > > TURKCELL bu mesajin icerdigi bilgilerin doğruluğu veya eksiksiz oldugu > konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin > ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve > saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca > gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir > > Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. > ------------------------------ > > This e-mail and any files transmitted with it are confidential and > intended solely for the use of the individual or entity to whom they > are addressed. If you are not the intended recipient you are hereby > notified that any dissemination, forwarding, copying or use of any of > the information is strictly prohibited, and the e-mail should > immediately be deleted. > > TURKCELL makes no warranty as to the accuracy or completeness of any > information contained in this message and hereby excludes any > liability of any kind for the information contained therein or for the > information transmission, reception, storage or use of such in any way > whatsoever. The opinions expressed in this message belong to sender > alone and may not necessarily reflect the opinions of TURKCELL. > > This e-mail has been scanned for all known computer viruses. > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 92 bytes Desc: not available URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/682b8d5d/…> ------------------------------ Message: 4 Date: Thu, 16 Jan 2020 17:29:53 +0000 From: Nuno Miguel Reis <nmreis(a)student.dei.uc.pt> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <CAGTw3V7Y8pHD+a8psPL7Tah=HTgrmvVRPhPBKB9VCuoujX_Nnw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi again. Thanks for all the help and suggestions. I realized the issue happens if using kernel forwarding only. If I change rtpengine to start at userspace without the kernel module enabled everything works fine as expected. Do you have any hints on why this could be happening with the kernel module? I'm running rtpengine like this: usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 --listen-ng= 127.0.0.1:2223 --tos=184 --sip-source kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 --listen-ng=127.0.0.1:2223 --tos=184 --no-fallback --sip-source when using kernel module, I'm also setting this rules on chain INPUT under iptables: -A INPUT -p udp -m udp --dport 5000:16383 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 16384:32768 -j RTPENGINE --id 0 -A INPUT -p udp -m udp --dport 49152:65535 -j RTPENGINE --id 0 Looking forward to hear from you guys. Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra On Thu, Jan 16, 2020 at 1:23 PM Richard Fuchs <rfuchs(a)sipwise.com> wrote: > On 15/01/2020 13.39, Nuno Miguel Reis wrote: > > Hi guys. > > I'm replacing a environment where I was using kamailio + freeswitch by > another where I'm adding rtpengine to the mix. > One of the issues I'm having now is when I have a SIP Client behind NAT: > > When I send the INVITE from the SIP Client, the SDP is using the > private LAN IP address + the advertised media port, let's admit it's > 10.10.10.10:5000 when the call establishes with a public server > running kamailio + rtpengine, the actual RTP arrives from the home > router public IP on a natted port, let's admit it's > 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. > I'm using the '--sip-source' with RTPENGINE to make it use the > received ip address instead of the private IP coming in the SDP but > I'm not finding anything to make RTPENGINE adapt an start sending the > RTP FLOW to the port where he starts receiving the RTP flow from, > discarding the SDP media port information. > > Rtpengine does this automatically, unless the `asymmetric` flag is used. > You should see a log message about a `confirmed peer address` as soon > as the first RTP is received on a port. > > Cheers > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/d61a6989/…> ------------------------------ Message: 5 Date: Thu, 16 Jan 2020 20:45:04 +0100 From: Federico Santulli <kam.list(a)nhm.it> To: miconda(a)gmail.com, "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Cc: "business(a)lists.kamailio.org" <business(a)lists.kamailio.org> Subject: Re: [SR-Users] Kamailio/RTC dinner at Fosdem 2020 Message-ID: <54CDD921-48D2-47F0-935E-0B181F037774(a)nhm.it> Content-Type: text/plain; charset=us-ascii Hi Daniel, We would like to join the dinner, 3 people from US. Kind regards. Federico Santulli NHM - S.R.L. Via Raffaello Sanzio, 88 81031 Aversa (CE) Italy > Il giorno 9 gen 2020, alle ore 09:52, Daniel-Constantin Mierla <miconda(a)gmail.com> ha scritto: > > Hello, > > Fosdem 2020 is approaching, there will be a talk from Henning about > Kamailio, many other friends and related projects are presenting in > the RTC Devroom (Giacomo Vacca and Federico Cabiddu, Wazo, Asterisk, > Homer, Janus, Jitsi, CGRateS, Linphone, ...): > > * https://fosdem.org/2020/schedule/track/real_time_communications/ > > I plan to be at the event and wondering if there are enough interested > participants for having the traditional Kamailio/RTC dinner. If yes, > Torrey will help to book a place to accommodate us (first choice could > be the same restaurant like the last year, if available). > > Reply if you want to join the dinner and say how many other people are > joining you. > > Cheers, > Daniel > > -- > Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- > www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, > 2020, in Berlin -- > https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www. > kamailioworld.com&umid=39581E1E-9C53-E505-96A6-E057F22BD18D&auth=cbd2d > 006abdb2241308511f72807fe0bc6594b4a-ac4b2b5016612d36d9bf3bf438b6b436b1 > f1ae8e > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ Message: 6 Date: Thu, 16 Jan 2020 22:56:03 +0100 From: Daniel-Constantin Mierla <miconda(a)gmail.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Fred Posner <fred(a)palner.com> Subject: Re: [SR-Users] http_client interface Message-ID: <4ab3422e-c3d6-7d85-fb3e-53b09bfa6c1a(a)gmail.com> Content-Type: text/plain; charset=utf-8 The libcurl seems to offer an option for that as well: - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html So it can be a small patch to the module and support it in the future. If you look for a solution on current stable, maybe one of the embedded scripting languages can help, you can execute inline a small script in Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the http client from lua to do more complex http api queries (e.g., with specific headers, ...). Cheers, Daniel On 13.01.20 17:00, Fred Posner wrote: > I'd never had a reason to before, but on a deployment I need http > messages to go out a specific interface. Curl has an --interface > option, but I don't see how to do the same with http_client. Unless > I'm missing something... > > Anyone ever do something like this? > -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=www.kamai… ------------------------------ Message: 7 Date: Thu, 16 Jan 2020 09:49:40 -0700 (MST) From: adeykunov <adeykunov(a)intermedia.net> To: sr-users(a)lists.sip-router.org Subject: Re: [SR-Users] Kamailio stop to process incoming SIP traffic via TCP. Message-ID: <1579193380553-0.post(a)n5.nabble.com> Content-Type: text/plain; charset=us-ascii Daniel, Got the same issue on 5.3.1 with openssl1.1, debian9. After 3 working days of tests (about ~30-50 wss clients), suddenly we've got a lot of connections stucked in CLOSE_WAIT state. Kamailio called sig_alarm_abort() when we try to reboot. Thanks, Andrey -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html ------------------------------ Message: 8 Date: Thu, 16 Jan 2020 22:02:40 +0000 (UTC) From: Jimmy Rodley <jimmyrodley(a)yahoo.com> To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question about xhttp module kamailio 5.2.x Message-ID: <443353431.8467378.1579212160245(a)mail.yahoo.com> Content-Type: text/plain; charset="utf-8" Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/f581f9a0/…> ------------------------------ Message: 9 Date: Thu, 16 Jan 2020 15:54:54 -0500 From: Jim Leahy <rickfromsunnyvale(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] Trouble with 302 Redirect Message-ID: <CAH3xi0-AEEj_bwLYd4OzSH96Cz8=qyrdsNWTjWnf5Sm1efrvhg(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Hi All, I'm having some issues trying to properly send a '302 Moved Temporarily' reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server when UAC2 is in DND. Here are the pertinent parts of my config: route { # drop some things right away route(DROP); # perform sanity check route(SANITYCHECK); # handle registrations if(is_method("REGISTER")) { route(REGISTRATION); exit; } # handle invites if(is_method("INVITE")) { route("INVITE"); exit; } route(RELAY); } route[INVITE] { # add this proxy to the record-route so it stays in the loop record_route(); # if we can find this user in the location DB... if(lookup("location")) { # relay the invite to the new dURI t_on_failure("FOURDIGITFAIL"); t_on_reply("FOURDIGITREPLY"); route(RELAY); # go back to the main route return; } onreply_route[FOURDIGITREPLY] { if(t_check_status("4[0-9]{2}")) { t_reply("302", "Moved Temporarily"); } } I'm having issues with the t_reply in the onreply_route. It successfully sends the reply, but I can't seem to set the Contact header properly. I'm trying to set it to the location of the user's mailbox on the voicemail sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: remove_hf("Contact"); insert_hf("Contact: sip:1000@vmpbx.domain.com"); t_reply("302", "Moved Temporarily"); But as I understand it, that just changes the header in the reply from UAC2, not the reply I'm sending to UAC1 Then I tried: append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); t_reply("302", "Moved Temporarily"); But I can't use append_to_reply in an onreply block. How can I change the Contact header in the reply that's sent? Am I just going about this the wrong way? Any help would be appreciated. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/66fc8ba2/…> ------------------------------ Message: 10 Date: Thu, 16 Jan 2020 17:10:06 -0500 From: Richard Fuchs <rfuchs(a)sipwise.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Kamalio + RTP Engine + SIP Client behind NAT Message-ID: <0b704a3c-afc4-3ca0-8d8b-38e2d70bcae9(a)sipwise.com> Content-Type: text/plain; charset="utf-8"; Format="flowed" On 16/01/2020 12.29, Nuno Miguel Reis wrote: > Hi again. > > Thanks for all the help and suggestions. I realized the issue happens > if using kernel forwarding only. If I change rtpengine to start at > userspace without the kernel module enabled everything works fine as > expected. > Do you have any hints on why this could be happening with the kernel > module? > > I'm running rtpengine like this: > > usersapece: $ rtpengine -f -L 7 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 --sip-source > > kernel: $ rtpengine -f -L 7 --table=0 --interface=100.100.100.100 > --listen-ng=127.0.0.1:2223 <http://127.0.0.1:2223> --tos=184 > --no-fallback --sip-source > The kernel module receives its instructions from the userspace daemon, so there's no reason there should be a difference. Can you post 1) logs and 2) the flags you use for your offers/answers? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200116/70d3b6de/…> ------------------------------ Message: 11 Date: Thu, 16 Jan 2020 17:11:23 -0500 From: Fred Posner <fred(a)palner.com> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] http_client interface Message-ID: <9808e784-3d56-d100-4c46-b136cc09ef08(a)palner.com> Content-Type: text/plain; charset=utf-8 On 1/16/20 4:56 PM, Daniel-Constantin Mierla wrote: > The libcurl seems to offer an option for that as well: > > - https://curl.haxx.se/libcurl/c/CURLOPT_INTERFACE.html > > So it can be a small patch to the module and support it in the future. > > If you look for a solution on current stable, maybe one of the embedded > scripting languages can help, you can execute inline a small script in > Lua/Python/Perl with app_lua_run() & equivalents. In the past I used the > http client from lua to do more complex http api queries (e.g., with > specific headers, ...). > > Cheers, > Daniel > Thank you Daniel, I implemented a work around (little proxy in go) but would love if the option was added for the future. Fred Posner fred(a)qxork.com https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com ------------------------------ Message: 12 Date: Thu, 16 Jan 2020 17:31:43 -0500 From: Alex Balashov <abalashov(a)evaristesys.com> To: sr-users(a)lists.kamailio.org Subject: Re: [SR-Users] Trouble with 302 Redirect Message-ID: <20200116223143.GA857@typhoon> Content-Type: text/plain; charset=us-ascii Try doing this in a failure_route[]; onreply_route[] isn't really meant to transform replies, just to passively observe or drop them. Also, you'll want to use append_to_reply(): https://kamailio.org/docs/modules/5.3.x/modules/textops.html#textops.f.appe… Note that the conditions that give rise to the invocation of a failure_route are broader and somewhat more abstract than any given SIP reply -- e.g. transaction reply timeouts. So, you would want to structure your failure_route like this: failure_route[REDIRECT_HANDLE] { if(t_is_canceled()) exit; if(t_branch_timeout()) { # Handle the timeout case... exit; } # Use $T_rpl to access reply code ($rs) and reply reason if needed # ($rs), per: # # https://www.kamailio.org/wiki/cookbooks/5.3.x/pseudovariables#t_rpl_pv if($T_rpl($rs) == 486) { append_to_reply("Contact: <sip:...>\r\n"); t_reply("302", "Moved Temporarily"); exit; } # Some other case. } -- Alex On Thu, Jan 16, 2020 at 03:54:54PM -0500, Jim Leahy wrote: > Hi All, > I'm having some issues trying to properly send a '302 Moved Temporarily' > reply to the calling UAC (UAC1) when the remote UAC (UAC2) returns a 486 > status (Do Not Disturb). The goal is to redirect UAC1 to a voicemail server > when UAC2 is in DND. > Here are the pertinent parts of my config: > > route { > # drop some things right away > route(DROP); > > # perform sanity check > route(SANITYCHECK); > > # handle registrations > if(is_method("REGISTER")) { > route(REGISTRATION); > exit; > } > > # handle invites > if(is_method("INVITE")) { > route("INVITE"); > exit; > } > > route(RELAY); > } > > route[INVITE] { > # add this proxy to the record-route so it stays in the loop > record_route(); > > # if we can find this user in the location DB... > if(lookup("location")) { > # relay the invite to the new dURI > t_on_failure("FOURDIGITFAIL"); > t_on_reply("FOURDIGITREPLY"); > route(RELAY); > # go back to the main route > return; > } > > onreply_route[FOURDIGITREPLY] { > if(t_check_status("4[0-9]{2}")) { > t_reply("302", "Moved Temporarily"); > } > } > > I'm having issues with the t_reply in the onreply_route. It successfully > sends the reply, but I can't seem to set the Contact header properly. I'm > trying to set it to the location of the user's mailbox on the voicemail > sever (ie sip:1000@vmpbx.domain.com), but nothing I try works. I've tried: > remove_hf("Contact"); > insert_hf("Contact: sip:1000@vmpbx.domain.com"); > t_reply("302", "Moved Temporarily"); > > But as I understand it, that just changes the header in the reply from > UAC2, not the reply I'm sending to UAC1 > > Then I tried: > append_to_reply("Contact: sip:1000@vmpbx.domain.com" ); > t_reply("302", "Moved Temporarily"); > > But I can't use append_to_reply in an onreply block. > How can I change the Contact header in the reply that's sent? Am I just > going about this the wrong way? Any help would be appreciated. Thanks > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2…, https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=http%3a%2… ------------------------------ Message: 13 Date: Fri, 17 Jan 2020 00:58:49 -0500 From: Dario Vargas <dariovargas.gcs(a)gmail.com> To: sr-users(a)lists.kamailio.org Subject: [SR-Users] UCaaS Platform Message-ID: <CAHqy4U_jecJALyvoZM6p6oGB5VyC_kZN=Sj3_vqjEhkO-tmORw(a)mail.gmail.com> Content-Type: text/plain; charset="utf-8" Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/943422d8/…> ------------------------------ Message: 14 Date: Fri, 17 Jan 2020 07:46:13 +0000 From: Henning Westerholt <hw(a)skalatan.de> To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org>, Dario Vargas <dariovargas.gcs(a)gmail.com> Subject: Re: [SR-Users] [commercial announcement] Message-ID: <2dcadf18-240a-1eb4-4a46-89d1d84c6007(a)skalatan.de> Content-Type: text/plain; charset="utf-8" Hello Dario, this type of content is not appropriate for the sr-users lists, please post it here. Please use our dedicated business list for this kind of commercial announcements. https://lists.kamailio.org/cgi-bin/mailman/listinfo/business Regards, Henning Am 17.01.20 um 06:58 schrieb Dario Vargas: commercial announcement _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio services - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… Kamailio Merchandising - https://ddei3-0-ctp.trendmicro.com:443/wis/clicktime/v1/query?url=https%3a%… -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20200117/fa884666/…> ------------------------------ Subject: Digest Footer _______________________________________________ sr-users mailing list sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ------------------------------ End of sr-users Digest, Vol 176, Issue 17 *****************************************

2 2

TLS Profiles : server_name and server_id query
by mahesh b 21 Jan '20

21 Jan '20

Hi All, Am using kamailio 5.1.9 version >From http://www.kamailio.org/docs/modules/5.1.x/modules/tls.html , i understand " The value for server_id can be any string, being used to match TLS client config profile, overriding the match on ip:port and server_name. " Which suggests that the server_name and server_id can be different for a tls profile IN my tls.cfg i have [client:default] verify_certificate = no require_certificate = no [client:10.211.160.172:5071] method = TLSv1.2 verify_certificate = yes require_certificate = yes private_key = /ssl/sip/profile1/server.key certificate = /ssl/sip/profile1/server.crt ca_list = /ssl/sip/profile1/ca.crt cipher_list = RSA verify_depth = 9 server_id = profile1 server_name = btip.176.com And in sar.cfg i am doing, before sending the request out $xavp(tls=>server_name)="btip.176.com"; $xavp(tls[0]=>server_id)="profile1"; But if i see the logs , its picking client default profile for tls handshake, Am i missing anything in tls.cfg or sar.cfg ? if i see the code in tls_lookup_cfg() while (p) { if(srvid && srvid->len>0) { LM_DBG("comparing addr: [%s:%d] [%s:%d] -- id: [%.*s] [%.*s]\n", ip_addr2a(&p->ip), p->port, ip_addr2a(ip), port, p->server_id.len, ZSW(p->server_id.s), srvid->len, ZSW(srvid->s)); if(p->server_id.s && p->server_id.len==srvid->len && strncasecmp(p->server_name.s, srvid->s, srvid->len)==0) { LM_DBG("TLS config found by server id\n"); return p; } .... Why is that there is a strncasecmp between p->server_name.s and srvid->s ?? the strncasecmp should be between p->server_id.s and srvid->s right ? Regards, Mahesh.B

2 1

Re: [SR-Users] [EXT] Re: Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 21 Jan '20

21 Jan '20

Hi Henning, It turned out that I didn't know how an e.164 lookup is initiated particularly the required format of the calling number. Reading through the source code, the function call: is_e164() showed I was missing the leading + char. I also noticed that I missed the LM_ERR in my log files eg: Jan 10 10:50:11 imskamailio /usr/local/sbin/kamailio[24552]: ERROR: enum [enum.c:840]: enum_query(): R-URI user '<1234567>' is not an E164 number In our case we are looking at standing up a ENUM server that can be queried for specific user characteristics to complete a call, so now we append the '+' when we want to check a caller or callee: enum_pv_query("+$rU", "enumsvc.e164.arpa.", "sip+video") enum_pv_query("+$fU", " enumsvc.e164.arpa", "+sip+video") All is good now. Thanks! Martin From: Henning Westerholt <hw(a)skalatan.de> Sent: Saturday, January 18, 2020 6:47 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; Woscek, Martin W. <mwoscek(a)mitre.org> Subject: [EXT] Re: [SR-Users] Kamailio 5.2.2 IMS -enum_query question Hello Martin, do you see any errors in the log file during the time that you execute the enum_query? Have you tried to test it with activated debug logging? Cheers, Henning Am 10.01.20 um 17:57 schrieb Woscek, Martin W.: Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com Kamailio Merchandising - https://skalatan.de/merchandising/

1 0

Re: [SR-Users] Kamailio service error
by Voip support 21 Jan '20

21 Jan '20

I think you need to capture w tcpdump on kamailio port and then check what packet is causing this. tcmdump -f "port 5060" -w file.pcap Then open file in wireshark and try figure out ( i think not valid sip packet coming on 5060 port). wt., 21 sty 2020 o 11:26 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I think you need to capture w tcpdump on kamailio port and then check what > packet is causing this. > > tcmdump -f "port 5060" -w file.pcap > > Then open file in wireshark and try figure out ( i think not valid sip > packet coming on 5060 port). > > wt., 14 sty 2020 o 08:54 Alex Balashov <abalashov(a)evaristesys.com> > napisał(a): > >> It would appear that an incoming SIP message has one or more >> grammatically invalid headers that confuse the parser? >> >> — >> Sent from mobile, with due apologies for brevity and errors. >> >> > On Jan 14, 2020, at 2:37 AM, Ratanraj Singh <ratanraj.singh2(a)gmail.com> >> wrote: >> > >> > >> > Hi All, >> > I am troubling with kamailio service error. Can someone help on this >> issue? >> > >> > >> > <image.png> >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >

1 0

Possible bug in Presence module !?
by SamyGo 21 Jan '20

21 Jan '20

Hi, I'm experiencing a strange behavior with the Kamailio presence module. I've a Polycom phone which takes the the XML tag "version" very seriously and so as soon it SUBSCRIBEs to a parking lot a NOTIFY is sent from server with "version=1" - When the Parking lot/extension gets busy it receives the second NOTIFY with "version=1" whereas it should've been 2. So it fails to light up the BLF icons. Now, any further notify correctly increments the version tag and Polycom blf works fine until its SUBSCRIBE expires and the first cycle of NOTIFY is exchanged. I can tell its a bug because for each NOTIFY the *event_route[tm:local-request]* prints the attribute $subs(version) and it correctly shows the sequence 1,2,3...onwards, However the sngrep/network trace tells it differently, the version shows up as 1,1,2,3...onwards *version:* kamailio 5.2.3 (x86_64/linux) c36229 flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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 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: c36229 compiled on 22:20:40 Jan 18 2020 with gcc 4.9.2 *MODULES:*#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_mwi.so" loadmodule "presence_xml.so" loadmodule "presence_dialoginfo.so" loadmodule "presence_reginfo.so" loadmodule "pua.so" loadmodule "pua_rpc.so" loadmodule "pua_dialoginfo.so" modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 60) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) modparam("pua", "outbound_proxy", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 1) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) modparam("presence", "presentity_table", "presentity") modparam("presence", "active_watchers_table", "active_watchers") modparam("presence", "watchers_table", "watchers") modparam("presence", "db_update_period", 60) modparam("presence", "db_table_lock_type", 0) modparam("presence", "local_log_level", 3) modparam("presence", "notifier_processes", 10) modparam("presence", "force_delete", 0) modparam("presence", "subs_db_mode", 2) modparam("presence", "db_table_lock_type", 0) modparam("presence", "expires_offset", 60) modparam("presence", "send_fast_notify", 1) modparam("presence", "clean_period", 30) modparam("presence", "publ_cache", 0) modparam("presence", "min_expires_action", PRESENCE_MIN_EXPIRES_ACTION) modparam("presence", "min_expires", PRESENCE_MIN_EXPIRES) modparam("presence", "max_expires", PRESENCE_MAX_EXPIRES) modparam("presence", "sip_uri_match", 1) modparam("presence", "waitn_time", 1) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", "sip:MY_HOST_IP:MY_HOST_PORT") modparam("presence_xml", "force_dummy_presence", 1) modparam("presence_xml", "force_active", 1) modparam("presence_xml", "db_url", DBURL) modparam("presence_dialoginfo", "force_single_dialog", 1) modparam("presence_dialoginfo", "force_dummy_dialog", 1) #!endif Looking to find some answers, Thanks, Sammy

2 1

Kamalio + RTP Engine + SIP Client behind NAT
by Nuno Miguel Reis 20 Jan '20

20 Jan '20

Hi guys. I'm replacing a environment where I was using kamailio + freeswitch by another where I'm adding rtpengine to the mix. One of the issues I'm having now is when I have a SIP Client behind NAT: When I send the INVITE from the SIP Client, the SDP is using the private LAN IP address + the advertised media port, let's admit it's 10.10.10.10:5000 when the call establishes with a public server running kamailio + rtpengine, the actual RTP arrives from the home router public IP on a natted port, let's admit it's 100.100.100.100:65100, event though, RTPENGINE assumes that RTP stream is coming from 100.100.100.100:5000. I'm using the '--sip-source' with RTPENGINE to make it use the received ip address instead of the private IP coming in the SDP but I'm not finding anything to make RTPENGINE adapt an start sending the RTP FLOW to the port where he starts receiving the RTP flow from, discarding the SDP media port information. In my previous setup with freeswitch alone, freeswitch did this transparently. How do you guys solve this issue in RTPENGINE? Regards, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

5 9

rfc: moving lib srutils to core
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, during the last developers meeting in Dusseldorf we had some discussions about simplifying the building process and internal dependencies/libraries. The src/lib/srutils contains now mainly the uuid and json functions, which are used by very common modules such as registrar/usrloc, tm, dialog, jsonrpcs, dmq*, ... There is a part related to the time recurrence matching, used by drouting, tmrec, but the code is not that significant in size, so I am planning to relocate the lib code to the core and get rid of dependencies to internal libsrutils to many modules. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Ann: Registration to Kamailio World 2020 is open!
by Daniel-Constantin Mierla 20 Jan '20

20 Jan '20

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (April 27-29, 2020, in Berlin, Germany) is now open! More details are available at: * https://www.kamailioworld.com/k08/registration/ Call for presentations is still open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 9-11, 2020, Berlin - www.asipto.com Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Kamailio 5.2.2 IMS -enum_query question
by Woscek, Martin W. 18 Jan '20

18 Jan '20

Hi, I am trying to invoke unum lookups from calls to enum_query() ,is_from_user_enum() and enum_pv_query(). However we are not seeing any enum lookups over the wire using wireshark. In the Kamailio.cfg the following are set: loadmodule "enum" modparam("enum", "domain_suffix", "mydnsserver.org.") modparam("enum", "i_enum_suffix", " mydnsserver.org.") sample use in our Kamailio.cfg file: if(!enum_query("mydnsserver.org.")) { xlog("....); #do something } else { Xlog("....") #do something else } I do see NAPTR lookups being done by the other Kamailio IMS processes initially when it starts up so I know it has connectivity to the DNS server, but none for the invoked enum queries. We are running Kamailio 5.2.2. Thanks, Martin

2 1

Question about xhttp module kamailio 5.2.x
by Jimmy Rodley 18 Jan '20

18 Jan '20

Hi, I am on kamailio 5.2.4 and want to use the xhttp module for certain management operations. The request involves some DB queries . The DB is remote and via ODBC and might have some delay.The documentation on the xhttp module states about latency http://kamailio.org/docs/modules/devel/modules/xhttp.html mentioning that the request is handled in an asynchronous manner. However, the JSONRPC-S module documentation says that this module implements the support for asynchronous RPC commands only for HTTP and HTTPS transports. https://www.kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idm104941… I am confused about the process under whose context the xttp request . Can there be a dedicated process to handle certain management requests which do some custom instructions like certain DB operations or initiate certain http requests ? Thanks,Rupesh

2 1

http_client interface
by Fred Posner 17 Jan '20

17 Jan '20

I'd never had a reason to before, but on a deployment I need http messages to go out a specific interface. Curl has an --interface option, but I don't see how to do the same with http_client. Unless I'm missing something... Anyone ever do something like this? -- Fred Posner fred(a)qxork.com https://qxork.com Direct/SMS: +1 (336) 439-3733 Need Fred? Call Fred. 336-HEY-FRED Matrix: @fred:matrix.lod.com

3 4

UCaaS Platform
by Dario Vargas 17 Jan '20

17 Jan '20

Take the LEAD in Unified Communications as a Service! We can help you get started in the next Telco Disruptive Business. White Label Hosted Cloud Platform as a Service (CPaaS) will enable you to offer all of the below products under your own brand - Hosted Contact Center (CCaaS) Hosted Unified Communications (UCaaS) Hosted PBX (Full Feature Set) SIP Trunk Ai Chat Bots Caller Profile Solutions (Display your clients' corporate identity on cell phones) to improve phone answer ratio. Video Surveillance (VSaaS integrated with AWS) Web Conferencing Virtual FAX Business SMS Software developed in the USA to have all the routing features required to connect with US and Canadian carriers for inbound and outbound (LRN, LNP, TF, SMS, Push, Video etc). Use our existing contracts and pricing for telco services or BYOC (bring your own carrier model). Alternately we can provide you with introductions to the carriers of choice below to get started - Nuwave.com - Number 1 in TF 8XX Inbound Voipinvite.com - 1000s of MPS and CPS for SMS and Voice Broadcast Voxbone.com - World Wide DiDs - Low Monthly minimum Peerlessnetwork.com - Outbound / Inbound Business calling Centurylink.com / Level 3 - USA DiDs and outbound and inbound - minimum volumes required Verizon.com - Outbound and Inbound enterprise business calls - minimum volumes required Dario Vargas info(a)dnl-switch.com

2 1

Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

11 46

Get destination IP from forwarded invite
by CUMHUR KARAHAN 16 Jan '20

16 Jan '20

Hi all, I'm a newbie for Kamailio and working on a feature needs end-users' destination IP in the first transferred/forwarded invite message. Basically, our system/call flow is like: ClientA -> Kamailio1 -> Kamailio2 -> ClientB Invite from the clientA: tls:212.252.34.165:32980 tls:172.21.193.100:5061 INVITE sip:905364167881@testsip.tvoip.turkcell.com.tr:443 SIP/2.0 Via: SIP/2.0/TLS 100.64.13.139:56993;branch=z9hG4bK.GoiTxA4Ee;rport From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 20 INVITE ... Needed Ip (Client B's IP 10.252.180.51 for this case) is in the INVITE row of the forwarded message. Invite to the Kamailio2 : udp:172.21.193.100:5060 udp:172.21.193.98:5060 INVITE sip:905364167881@10.252.180.51:36945;transport=tls SIP/2.0 Record-Route: <sip:gearbox11.turkcell.tgc:5060;r2=on;lr;did=7f5.f572;nat=yes> Record-Route: <sip:testsip05.tvoip.turkcell.com.tr:443;transport=tls;r2=on;lr;did=7f5.f572;nat=yes> Via: SIP/2.0/UDP gearbox11.turkcell.tgc:5060;branch=z9hG4bK67ad.4e8e863f2475ab8d3053af0593548557.1;i=f Via: SIP/2.0/TLS 100.64.13.139:56993;received=212.252.34.165;branch=z9hG4bK.1SYkXD1Me;rport=32980 From: <sip:905368601286@testsip.tvoip.turkcell.com.tr>;tag=WbINy0ypx To: sip:905364167881@testsip.tvoip.turkcell.com.tr CSeq: 21 INVITE ... When the first invite message receives, the dialog variables or sel(msg...) parsed values belong to this first received message, as far as I understand during the tests. However I need the forwarded invite message's variables. Even after t_relay in RELAY route, it looks the message I have is the first one. So, I cannot reach this later IP in the forwarded invite message. As far as I understand, bear with me here please, this IP manipulation executes in t_relay function and in the C side of Kamailio. So, I cannot find a way to do that in the our native scripts. Is there any way to get it without diving C code's abyss? Do you have any idea about it? I would appreciate it, if I can get any help or any clue. Thanks for all... S. Cumhur Karahan [http://s.turkcell.com.tr/SiteAssets/Genel/mail-imza.jpg] <http://turkcell.li/ncfYh> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses.

2 1

Negative ACK issue
by Michael Broughton 15 Jan '20

15 Jan '20

Hello, Long time Kam/Ser user, first time poster. I'm running into a problem with one of our telco providers when we make a call that ends up being not in service or some other error. In this case our ACK's are not working and the phone line stays open for a period of time until something times out on their end. They claim the issue is that our negative ACK message is dropping one of the Via headers. This is the only case I can find in our setup where Kamailio does this. But it does drop the first Via, which is the first hop in our internal network. I don't understand why this is a problem for them, and I'm still trying to get a reasonable explanation out of them. Technically, I don't see why it would be a problem. This behaviour is not an issue with our other telco providers. Strangely enough, it is also not an issue for this provider when we make the calls over their MPLS network (we are switching to the internet). My question is, can this behaviour be changed in Kamailio somehow? Is there a way for it to keep all the Via headers for negative ACK's? Or, do I just need to poke them harder to fix their issues? My setup: Source -> Kamailio -> Firewall (NAT, SIP ALG) -> Telco Provider I hope I have provided enough information. Thanks! Michael

4 9

Kamailio Slack
by Abdirahman A. Osman 15 Jan '20

15 Jan '20

Hello, Is the Kamailio Slack open for sr-users , if yes where can I find the invite link? Thanks Abdirahman

3 2

Re: [SR-Users] sr-users Digest, Vol 176, Issue 11
by Sam Ware 14 Jan '20

14 Jan '20

Alex, I get the following error ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [ ERROR: bad config file (1 errors) CRITICAL: <core> [core/cfg.y:3537]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 462, column 23-40: Can't get from cache: $(rU{s.select,0,#) David, I did try escaping and get the same error. $avp(techprefix) = $(rU{s.select,0,\#}); additionally, I think I initially tried it double quoted. $avp(techprefix) = $(rU{s.select,0,"#"}); Resulted in: ERROR: <core> [core/pvapi.c:1127]: pv_parse_spec2(): wrong char [ ERROR: bad config file (1 errors) CRITICAL: <core> [core/cfg.y:3537]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 462, column 23-42: Can't get from cache: $(rU{s.select,0,"#") Here is the entire block. if ($rU =~ "[\*\#]") { xlog("L_INFO", "[$ci] -> $rU appears to have a tech prefix, processing...\n"); if ($rU =~ "\*") { $avp(techprefix) = $(rU{s.select,0,*}); ## Reset the RURI user to the non-tech prefix'ed value $rU = $(rU{s.select,1,*}); ## Overwrite the message source variable with "sourceip-techprefix" $var(messagesource) = $_s($si-$avp(techprefix)); } else if ($rU =~ "\#") { $avp(techprefix) = $(rU{s.select,0,#)}); ## Reset the RURI user to the non-tech prefix'ed value $rU = $(rU{s.select,1,#}); ## Overwrite the message source variable with "sourceip-techprefix" $var(messagesource) = $_s($si-$avp(techprefix)); } } On Sat, Jan 11, 2020 at 5:00 AM <sr-users-request(a)lists.kamailio.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > Today's Topics: > > 1. Re: Kamailio in Docker Compile Error (Sergey Safarov) > 2. Re: Kamailio in Docker Compile Error (Sam Ware) > 3. Kamailio 5.2.2 IMS -enum_query question (Woscek, Martin W.) > 4. String Transformation Question (Sam Ware) > 5. Re: String Transformation Question (Alex Balashov) > 6. Re: String Transformation Question (David Villasmil) > > > > ---------- Forwarded message ---------- > From: Sergey Safarov <s.safarov(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 14:38:02 +0300 > Subject: Re: [SR-Users] Kamailio in Docker Compile Error > please try kamailio-ci:5.3.2-alpine > I will fix error about kamailio-ci:5.3.1-alpine. > > On Fri, Jan 10, 2020 at 1:53 PM Asgaroth <00asgaroth00(a)gmail.com> wrote: > >> Hi, >> >> I see that the kamailio-ci:5.3.1-alpine tag exists, but was getting an >> error when trying to pull it 2 days ago, 5.3.0- alpine works fine. >> >> Unfortunately I cannot post the actual docker pull error right now (in >> airport, on my phone) >> >> On Fri 10 Jan 2020, 08:32 Sergey Safarov, <s.safarov(a)gmail.com> wrote: >> >>> yum -y install yum-utils >>> yum-config-manager --add-repo http://rpm.kamailio.org/centos/kamailio.repo >>> yum install kamailio >>> >>> >>> Also you can use this docker image >>> https://hub.docker.com/repository/docker/kamailio/kamailio-ci >>> >>> On Fri, Jan 10, 2020 at 10:58 AM Daniel-Constantin Mierla < >>> miconda(a)gmail.com> wrote: >>> >>>> Hello, >>>> >>>> try to install which and pkg-config tools. >>>> >>>> Cheers, >>>> Daniel >>>> On 10.01.20 02:21, Sam Ware wrote: >>>> >>>> I am attempting to create my own docker image on Centos 7. During the >>>> build, I get the following error message during the "make all" phase. >>>> ---------------------------------------------------------------------- >>>> CC (gcc) [kamailio] core/cfg.tab.o >>>> LD (gcc) [kamailio] kamailio >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> make[1]: which: Command not found >>>> CC (gcc) [M db_mysql.so] km_dbase.o >>>> km_dbase.c:36:19: fatal error: mysql.h: No such file or directory >>>> #include <mysql.h> >>>> ^ >>>> compilation terminated. >>>> make[1]: *** [km_dbase.o] Error 1 >>>> make: *** [modules] Error 1 >>>> ------------------------------------------------------------------------ >>>> >>>> I checked and the mysql.h exists in the /usr/include/mysql directory. >>>> >>>> Any thoughts? >>>> >>>> FYI, here is my Dockerfile >>>> >>>> FROM centos:7 >>>> MAINTAINER “Sam D Ware” sware(a)o1.com >>>> ENV container docker >>>> RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i >>>> == systemd-tmpfiles-setup.service ] || rm -f $i; done); >>>> RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ >>>> && rm -rf /etc/systemd/system/.wants/ \ >>>> && rm -rf /lib/systemd/system/local-fs.target.wants/ \ >>>> && rm -f /lib/systemd/system/sockets.target.wants/udev \ >>>> && rm -f /lib/systemd/system/sockets.target.wants/initctl \ >>>> && rm -rf /lib/systemd/system/basic.target.wants/ \ >>>> && rm -f /lib/systemd/system/anaconda.target.wants/* >>>> VOLUME [ “/sys/fs/cgroup”] >>>> RUN yum -y install epel-release >>>> RUN yum -y update >>>> RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel >>>> RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make >>>> WORKDIR /usr/local/src/ >>>> RUN git clone --depth 1 --no-single-branch >>>> https://github.com/kamailio/kamailio kamailio >>>> WORKDIR /usr/local/src/kamailio/ >>>> RUN git checkout -b 5.3 origin/5.3 >>>> RUN make cfg >>>> COPY modules.lst /usr/local/src/kamailio/src/ >>>> RUN make all >>>> RUN make install >>>> RUN groupadd kamailio >>>> RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" >>>> --home /var/run/kamailio kamailio >>>> RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig >>>> /etc/sysconfig/kamailio >>>> COPY kamailio.service /etc/systemd/system/kamailio.service >>>> RUN yum clean all; systemctl enable kamailio >>>> CMD ["/usr/sbin/init"] >>>> >>>> -- >>>> Sam D Ware >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> -- >>>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > > > ---------- Forwarded message ---------- > From: Sam Ware <sam.ware(a)gmail.com> > To: miconda(a)gmail.com > Cc: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Bcc: > Date: Fri, 10 Jan 2020 07:23:46 -0600 > Subject: Re: [SR-Users] Kamailio in Docker Compile Error > Thanks Daniel, > > That got me past that hurdle. It was the "which". Looks like the > pkg-config tools was already install; probably a preq for one of the other > pkgs. > > Sam > > On Fri, Jan 10, 2020 at 1:58 AM Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> try to install which and pkg-config tools. >> >> Cheers, >> Daniel >> On 10.01.20 02:21, Sam Ware wrote: >> >> I am attempting to create my own docker image on Centos 7. During the >> build, I get the following error message during the "make all" phase. >> ---------------------------------------------------------------------- >> CC (gcc) [kamailio] core/cfg.tab.o >> LD (gcc) [kamailio] kamailio >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> make[1]: which: Command not found >> CC (gcc) [M db_mysql.so] km_dbase.o >> km_dbase.c:36:19: fatal error: mysql.h: No such file or directory >> #include <mysql.h> >> ^ >> compilation terminated. >> make[1]: *** [km_dbase.o] Error 1 >> make: *** [modules] Error 1 >> ------------------------------------------------------------------------ >> >> I checked and the mysql.h exists in the /usr/include/mysql directory. >> >> Any thoughts? >> >> FYI, here is my Dockerfile >> >> FROM centos:7 >> MAINTAINER “Sam D Ware” sware(a)o1.com >> ENV container docker >> RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i == >> systemd-tmpfiles-setup.service ] || rm -f $i; done); >> RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ >> && rm -rf /etc/systemd/system/.wants/ \ >> && rm -rf /lib/systemd/system/local-fs.target.wants/ \ >> && rm -f /lib/systemd/system/sockets.target.wants/udev \ >> && rm -f /lib/systemd/system/sockets.target.wants/initctl \ >> && rm -rf /lib/systemd/system/basic.target.wants/ \ >> && rm -f /lib/systemd/system/anaconda.target.wants/* >> VOLUME [ “/sys/fs/cgroup”] >> RUN yum -y install epel-release >> RUN yum -y update >> RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel >> RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make >> WORKDIR /usr/local/src/ >> RUN git clone --depth 1 --no-single-branch >> https://github.com/kamailio/kamailio kamailio >> WORKDIR /usr/local/src/kamailio/ >> RUN git checkout -b 5.3 origin/5.3 >> RUN make cfg >> COPY modules.lst /usr/local/src/kamailio/src/ >> RUN make all >> RUN make install >> RUN groupadd kamailio >> RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" --home >> /var/run/kamailio kamailio >> RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig >> /etc/sysconfig/kamailio >> COPY kamailio.service /etc/systemd/system/kamailio.service >> RUN yum clean all; systemctl enable kamailio >> CMD ["/usr/sbin/init"] >> >> -- >> Sam D Ware >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >> >> > > -- > Sam D Ware > > > > > ---------- Forwarded message ---------- > From: "Woscek, Martin W." <mwoscek(a)mitre.org> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 16:57:29 +0000 > Subject: [SR-Users] Kamailio 5.2.2 IMS -enum_query question > > Hi, > > > > I am trying to invoke unum lookups from calls to enum_query() > ,is_from_user_enum() and enum_pv_query(). > > However we are not seeing any enum lookups over the wire using wireshark. > > > > In the Kamailio.cfg the following are set: > > > > loadmodule "enum" > > modparam("enum", "domain_suffix", "mydnsserver.org.") > > modparam("enum", "i_enum_suffix", " mydnsserver.org.") > > > > sample use in our Kamailio.cfg file: > > > > if(!enum_query("mydnsserver.org.")) { > > xlog(“….); > > #do something > > } else { > > Xlog(“….”) > > #do something else > > } > > > > I do see NAPTR lookups being done by the other Kamailio IMS processes > initially when it starts up so I know it has connectivity to the DNS > server, but none for the invoked enum queries. > > > > We are running Kamailio 5.2.2. > > > > Thanks, > > Martin > > > > ---------- Forwarded message ---------- > From: Sam Ware <sam.ware(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 17:22:17 -0600 > Subject: [SR-Users] String Transformation Question > We are receiving traffic with a tech prefix. We want to be able to use a > # (hash tag) for the separator for the tech prefix and the remote user > ($rU). I was using the s.select transformation before when using * for > separator, but I get an error because it appears that it being interpreted > as a comment. Any suggestions? > > Example: > ## How I was doing it with the * > $avp(techprefix) = $(rU{s.select,0,*}); > > ## How I tried and got an error for the hash tag > $avp(techprefix) = $(rU{s.select,0,#}); > -- > Sam D Ware > > > > > ---------- Forwarded message ---------- > From: Alex Balashov <abalashov(a)evaristesys.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 18:24:44 -0500 > Subject: Re: [SR-Users] String Transformation Question > What’s the error? > > — > Sent from mobile, with due apologies for brevity and errors. > > On Jan 10, 2020, at 6:23 PM, Sam Ware <sam.ware(a)gmail.com> wrote: > > > We are receiving traffic with a tech prefix. We want to be able to use a > # (hash tag) for the separator for the tech prefix and the remote user > ($rU). I was using the s.select transformation before when using * for > separator, but I get an error because it appears that it being interpreted > as a comment. Any suggestions? > > Example: > ## How I was doing it with the * > $avp(techprefix) = $(rU{s.select,0,*}); > > ## How I tried and got an error for the hash tag > $avp(techprefix) = $(rU{s.select,0,#}); > -- > Sam D Ware > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > > ---------- Forwarded message ---------- > From: David Villasmil <david.villasmil.work(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> > Cc: > Bcc: > Date: Fri, 10 Jan 2020 23:24:51 +0000 > Subject: Re: [SR-Users] String Transformation Question > Have you tried scaping it? Never done it, tho > > On Fri, 10 Jan 2020 at 23:23, Sam Ware <sam.ware(a)gmail.com> wrote: > >> We are receiving traffic with a tech prefix. We want to be able to use a >> # (hash tag) for the separator for the tech prefix and the remote user >> ($rU). I was using the s.select transformation before when using * for >> separator, but I get an error because it appears that it being interpreted >> as a comment. Any suggestions? >> >> Example: >> ## How I was doing it with the * >> $avp(techprefix) = $(rU{s.select,0,*}); >> >> ## How I tried and got an error for the hash tag >> $avp(techprefix) = $(rU{s.select,0,#}); >> -- >> Sam D Ware >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > -- > Regards, > > David Villasmil > email: david.villasmil.work(a)gmail.com > phone: +34669448337 > _______________________________________________ > sr-users mailing list > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- Sam D Ware

3 3

Kamailio service error
by Ratanraj Singh 14 Jan '20

14 Jan '20

Hi All, I am troubling with kamailio service error. Can someone help on this issue? [image: image.png]

2 1

Trigger a custom SIP Notify Message on demand
by Rupesh Kumar 14 Jan '20

14 Jan '20

Hello kamailio gurus, I was wondering if there is a way to send a custom SIP NOTIFY message from kamailio to the registered SIP clients on-demand. For example, I want to take a kamailio node for maintainance and want all polycom phones registered on a kamailio node to re-register and move to a different kamailio node. I know the uac module can send custom requests via uac_req_send, but can I invoke this via a kamcmd or kamctl command ?Any pointers/clues would be a great help. Thanks,RK

2 3

topos and db_redis
by Ben Merrills 13 Jan '20

13 Jan '20

Hi All, Quick question, I've tried to find this myself, but there doesn't seem to be any documentation that I can find on it. Does the topos module work with db_redis module? I can see two schema files for topos (topos_t, topos_d), but I can't work out what the keys should be. modparam("db_redis", "keys", "topos_t=entry:hash_entry,hash_id,...?") modparam("db_redis", "keys", "topos_d=entry:hash_entry,hash_id,...?") modparam("topos", "db_url", "redis://127.0.0.1:6379/kamailio") modparam("topos", "storage", "db") Any pointers would be greatly appreciated. Kind Regards, Ben Merrills

2 1

Diameter EVENT_REQUEST
by Steve Bucklin 13 Jan '20

13 Jan '20

Dear All, I was wondering if someone could give me a pointer on handling diameter "EVENT REQUEST" (416) I am a complete newb to the diameter, and have used a RESTCOMM SMS-C to generate charging requests. 1. I have got the CDP and other modules going and it connects with application (Diameter Credit Control Application (4) 2. I tried to use a simple ims_ocs module, but get a "getMethod()" error. Looking at the code, there are only three supported, START, INTERIM, & STOP. 3. I have captured the message in pcap - and all looks OK. 4. To examine the whole message I tried using the ims_diameter_server module to produce a JSON of all the values sent in the $diameter_request within the event_route, HOWEVER once the request arrives I get an error: ERROR: cdp [diameter_avp.c:645]: AAAUngroupAVPS(): hss3g_ungroup_avps: source buffer to short!! Cannot read the whole AVP header! I assume that the cdp function AAAUngroupAVPS() cannot sort the data into JSON due to a buffer issue. Any pointers? Steve

1 0

Dúvidas
by Genildo Nascimento 13 Jan '20

13 Jan '20

Boa Noite. Instalei o kamailio mas não consigo usar Segue a configuração. baixei e adicionei chave GPG à sua lista de chaves apt wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key a Em seguida, adicionei as linhas do repositório ao seu arquivo /etc/apt/sources.list , Adicionei um repositório para a versão mais recente. sudo nano /etc/apt/source.list.d/kamailio.list versão ubuntu 16.04 deb http://deb.kamailio.org/kamailio53 xenial main deb-src http://deb.kamailio.org/kamailio53 xenial main Atualização do sistema e instalação do Kamailio sudo apt update sudo apt install kamailio kamailio-mysql-modules Editei o arquivo /etc/kamailio/kamctlrc e verifiquei se a DBENGINE variável está definida como MySQL. Removi o #símbolo para descomentá-lo. $ sudo nano / etc / kamailio / kamctlrc DBENGINE = MYSQL DBHOST = host local Instalei banco de dados apt install mysql-server modulos também apt -y install kamailio kamailio-mysql-modules kamailio-utils-modules rtpproxy Em seguida criei banco de dados para Mysql. sudo kamdbctl create logo após foi inserida a senha (xxxx) editei o arquivo /etc/kamailio/kamailio.cfg #! define WITH_MYSQL #! define WITH_AUTH *#! define WITH_USRLOCDB -> ao adicionar este iten o kamailio reiinicializa com erro (segue anexo arquivo jpeg)* #! define WITH_ACCDB Editei a linha da variável RUN_KAMAILIO no arquivo /etc/default/kamailio para que o kamailio possa rodar. RUN_KAMAILIO=yes configuração do rtpproxy no arquivo /ec/default/rtpproxy retirada do #simbolo control_sock=udp:localhost:7722 EXTRA_OPTS=" -l < IP local >" reinicio RTPproxy Obs: O sistema tambem não existe o arquivo /var/log/kamailio.log

1 0

String Transformation Question
by Sam Ware 10 Jan '20

10 Jan '20

We are receiving traffic with a tech prefix. We want to be able to use a # (hash tag) for the separator for the tech prefix and the remote user ($rU). I was using the s.select transformation before when using * for separator, but I get an error because it appears that it being interpreted as a comment. Any suggestions? Example: ## How I was doing it with the * $avp(techprefix) = $(rU{s.select,0,*}); ## How I tried and got an error for the hash tag $avp(techprefix) = $(rU{s.select,0,#}); -- Sam D Ware

3 2

Kamailio in Docker Compile Error
by Sam Ware 10 Jan '20

10 Jan '20

I am attempting to create my own docker image on Centos 7. During the build, I get the following error message during the "make all" phase. ---------------------------------------------------------------------- CC (gcc) [kamailio] core/cfg.tab.o LD (gcc) [kamailio] kamailio make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found make[1]: which: Command not found CC (gcc) [M db_mysql.so] km_dbase.o km_dbase.c:36:19: fatal error: mysql.h: No such file or directory #include <mysql.h> ^ compilation terminated. make[1]: *** [km_dbase.o] Error 1 make: *** [modules] Error 1 ------------------------------------------------------------------------ I checked and the mysql.h exists in the /usr/include/mysql directory. Any thoughts? FYI, here is my Dockerfile FROM centos:7 MAINTAINER “Sam D Ware” sware(a)o1.com ENV container docker RUN (cd /lib/systemd/system/sysinit.target.wants/; for i in ; do [ $i == systemd-tmpfiles-setup.service ] || rm -f $i; done); RUN rm -rf /lib/systemd/system/multi-user.target.wants/ \ && rm -rf /etc/systemd/system/.wants/ \ && rm -rf /lib/systemd/system/local-fs.target.wants/ \ && rm -f /lib/systemd/system/sockets.target.wants/udev \ && rm -f /lib/systemd/system/sockets.target.wants/initctl \ && rm -rf /lib/systemd/system/basic.target.wants/ \ && rm -f /lib/systemd/system/anaconda.target.wants/* VOLUME [ “/sys/fs/cgroup”] RUN yum -y install epel-release RUN yum -y update RUN yum -y install git-core gcc gcc-c++ flex bison mysql-devel RUN yum -y install mariadb-devel hiredis hiredis-devel curl nss make WORKDIR /usr/local/src/ RUN git clone --depth 1 --no-single-branch https://github.com/kamailio/kamailio kamailio WORKDIR /usr/local/src/kamailio/ RUN git checkout -b 5.3 origin/5.3 RUN make cfg COPY modules.lst /usr/local/src/kamailio/src/ RUN make all RUN make install RUN groupadd kamailio RUN adduser --system -g kamailio --shell /bin/false -c "Kamailio" --home /var/run/kamailio kamailio RUN cp /usr/local/src/kamailio/pkg/kamailio/obs/kamailio.sysconfig /etc/sysconfig/kamailio COPY kamailio.service /etc/systemd/system/kamailio.service RUN yum clean all; systemctl enable kamailio CMD ["/usr/sbin/init"] -- Sam D Ware

5 6

caching 404 responses
by Krzysztof Drewicz 10 Jan '20

10 Jan '20

Hello would like to see if I'm not reinventing the wheel: scenario: some old powerdialler/predictive calls kamailio as call dispatcher, kamailio then dials carrier. But this dialler does many calls to the same B number with 404 response (from carrier) in short (retry is < 600s) time, so no need to bother PSTN/carrier with retry. Should I simply use htable, cache 404 hit, use some reasonable value modparam("htable", "db_expires", 1) put there like 600 s TTL or something like that? Any hints/thoughts? Maybe this scenario is common and being used with success somewhere (and yes, this powerdialer is not very clever, and I do know it should not be needed for this) Thanks, -- *Krzysztof Drewicz* Senior Infrastructure Administrator CLUDO | ul. Grochowska 306/308, 03-840 Warszawa t+48221223977 kdrewicz(a)cludo.pl | www.cludo.pl <http://www.cludo.pl/>

2 1

TCP packet bundling
by Alex Balashov 09 Jan '20

09 Jan '20

Hi, I wanted to get clarity on a rather basic question: When using Kamailio with TCP, is it possible to end up with fragments of multiple SIP messages in a single "on-the-wire" packet? >From what I understand, the purpose of Kamailio's setting TCP_NODELAY is to prevent such bundling--as much as it is possible to control the fragmentation/packetisation behaviour of the transport protocol from a user-space application. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 7

Huge DB load while using location and registrar module (save()).
by Voip support 09 Jan '20

09 Jan '20

Dear Community, I am facing an issue with using the usrloc and registrar module to save registration informations in database. It works for me but when tested yesterday with around 1500-1600 users i got a lot of queries on database mostly SELECT. I figured out that changing db_mode to 0 resolves the issue because it works totally in memory (almost no select query on DB). So i think its location module. I tried db_mode=3 and db_mode=2 but both create a lot of load on database. The only function i need is to store some information of REGISTER in DB to know the count of registered users and the source IP / user-agent / username data. It needs to be saved in table on successfull REGISTRATION and removed when registration is expired or somebody UN-REGISTER. When used location module for that there was lots of queries like this (20000-50000 per second): select `contact`,`expires`,`q`,`callid`,`cseq`,`flags`,`cflags`,`user_agent`,`received`,`path`,`socket`,`methods`,`last_modified`,`ruid`,`instance`,`reg_id`,`server_id`,`connection_id`,`keepalive` from `location` where `username`='xxxx' order by q update `location` set `expires`='2020-01-08 18:51:39',`q`=-1.000000 ,`cseq`=2042,`flags`=0,`cflags`=64,`user_agent`='n/a',`received`='sip:xxxx:5060',`path`=NULL,`socket`='udp:xxxx:20051',`methods`=NULL,`last_modified`='2020-01-08 18:50:39',`callid`='1205705227@xxxx ',`instance`=NULL,`reg_id`=0,`server_id`=0,`connection_id`=-1,`keepalive`=1,`contact`='sip:1019@xxx:5060' where `ruid`='uloc-5e1642a3-2b90-0161' Looking up for ideas how to resolve this issue. I was thinking of implementing INSERT UPDATE and DELETE based on SQLops. But how to recognize that REGISTER was expiered? I would like location module to work mostly like in-memory. For each new registration INSERT entry to DB, each REGISTER re-new an UPDATE or DELETE should be triggered. So i would expect for 1000 users around 2000 operations per minute. Best regards, Tom

5 6

Kamailio v5.3.2 Released
by Daniel-Constantin Mierla 09 Jan '20

09 Jan '20

Hello, Kamailio SIP Server v5.3.2 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. 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 v5.3.2. For more details about version 5.3.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-3-2-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 Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Ask for Siremis document
by Thanh Tran 09 Jan '20

09 Jan '20

Hi, I looking for Administration Document for Siremis but can't find it. Can you share document ? Thanks,

2 1

Releasing Kamailio v5.3.2
by Daniel-Constantin Mierla 09 Jan '20

09 Jan '20

Hello, not so many fixes since the release of 5.3.1, but it looks enough to release a new minor version from branch 5.3. I am planning to do in on Thursday, Jan 9, 2020, if there are no new reports of relevant issues that make us to reconsider and postpone. Soon after we should probably do a release from branch 5.2 as well as the last one from branch 5.1. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 1

Parsing exception when "user=phone" is present
by Duarte Rocha 08 Jan '20

08 Jan '20

Greetings, I'm testing some portability scenarios with carriers and i've ran into some issues. Take this example URI from ETSI TR 184 003 ( https://www.etsi.org/deliver/etsi_tr/184000_184099/184003/03.01.01_60/tr_18… ) - Sip:+49234598765;npdi;rn=+49-D123@provider5.net;user=phone In this case i need to parse user, user's parameters, domain and uri parameters. In order to that i'm using uri's transformations like {uri.user}, {uri.domain}, etc. However, when "user=phone" is present, Kamailio's parser works differently. Without "user=phone" -> {uri.user} is equal to "+49234598765;npdi;rn=+49-D123" With "user=phone" -> {uri.user} is equal to "+49234598765" . I really need to access "rn" and "npdi" parameters in those cases. How can i do it when "user=phone" is present? Is there a way to disable the "user=phone" exception for parsing? Best Regards, Duarte Rocha

2 2

possible TCP deadlock (tls again?) // pike module not releasing IPs
by Aymeric Moizard 08 Jan '20

08 Jan '20

Hi List, History: * In the past, I had deadlock which was, most probably, related to ssl1.1. We have discussed this issue, and a fix is supposed to workaround the issue that was detected. * With latest 5.2.X, I have experienced ONCE a similar behavior with TCP and TLS being mostly stuck. I have not been using this version much, but the fix was supposed to be in the core of kamailio. The status of the server this night: * I'm today running version: kamailio 5.3.1 (x86_64/linux), * Installed on stretch using http://deb.kamailio.org/kamailio53 repository. * This versions use libssl1.1 * A user reported that he can't connect with TCP * An average of 5000 IPs per 10 minutes are being banned by the pike module (could be twice the same) Yesterday/Today: * at the end of the outage, I had 2479 IP in my ipban htable. (which is equivalent to my statistics showing 2 bans/IP every 10 minutes = 5000) * looking at my logs, it appears that most (ALL?) ip being banned... are my regular users. * looking at my logs, I can't understand why pike would block them. This is a graph for statistics on my service for the last 24 hours: https://www.antisip.com/sip-antisip-com-register/status2.html Yesterday, at 22:18:39, kamailio started to BAN some IPs. 52 IPs were banned in a period of 10 minutes. I can confirm this from my logs. My pike configuration is this one: modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 64) modparam("pike", "remove_latency", 4) When detecting the issue, this morning, I typed: $> sudo kamctl stats $> sudo kamcmd htable.dump ipban //FAILURE (answer too large...) $> sudo kamctl trap Then, I started an agent with TCP and it worked...??? Then, a few seconds, may be a minute after: $> sudo kamcmd htable.dump ipban //SUCCESS and shows 2479 banned ip. and... everything is back to normal in a few minutes. I haven't restarted kamailio, and all statistics are as expected, as usual. Thus, it looks that " sudo kamctl trap" has triggered something. I already experienced a similar behavior -when testing my ssl1.1 deadlock last year-. 2 questions: 1/ I beleive my "pike" configuration should not ban users. Is my pike configuration wrong? As an example, pike has banned an IP sending one message/second. I believe my configuration should accept that? 2/ Could there still be a TLS issue with libssl1.1? This is the result of the "kamctl trap": https://sip.antisip.com/kamailio-pike-or-tls-issue-13-12-2019.kamctl-trap Sorry for the long story & hoping to find a long term solution or at least a workaround! Regards Aymeric -- Antisip - http://www.antisip.com

3 11

Re: [SR-Users] Add user issue - Multiicon
by technical＠multiicon.in 08 Jan '20

08 Jan '20

Dear Kamailio team, Greetings from Multiicon. We setuping kamailio sip server and all thing are done but when we going to add user we are getting Command not found. Using below command. I also check the server status by using systemctl status kamailio is showing active Please advise on this. Thanks & Regards, VIVEK AGHERA #technical department IND : +91 63548 23958 <mailto:vivek@multiicon.in> vivek(a)multiicon.in MULTIICON IDEOTECHNOLOGY PVT.LTD. 3rd Floor, Rumi Plaza, Airport Road, Rajkot - 360 001, Gujarat (INDIA) Please consider the environment before printing this e-mail.

2 1

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

3 4

Kamailio when acting as client doesnt send SNI in client hello handshake message
by mahesh b 07 Jan '20

07 Jan '20

Hi, Am using Kamailio 5.1.9 version My Setup : client1 -> kamailio server 1 ( IP : 10.211.160.172) ----> kamailio server 2( IP : 10.211.160.176) -> client2 I have a scenario where kamailio server 1 has to initiate an outgoing tls connection to kamailio server 2, i have set the server_name and server_id in the client profile in tls.cfg like below on kamailio server 1 [client:default] verify_certificate = no require_certificate = no server_name = mahesh.client.com [client:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.176.com server_id = btip.176.com And in sar.cfg $xavp(tls=>server_name)="btip.176.com"; $xavp(tls=>server_id)="btip.176.com"; $du = "sip:10.211.160.176:5061;transport=tls"; .... t_relay(); What i observe is that , when client hello is sent by 10.211.160.172 to 10.211.160.176, i dont see Extension server_name being sent. Am i missing anything. Please help !

4 7

app_lua with kamailio 5.3.X
by Aymeric Moizard 07 Jan '20

07 Jan '20

Hi All! I have moved from kamailio 5.2.X to 5.3.X and I have now an error with calling "sr.err" from my lua script: sr.err("ERROR: param $tU [" .. tu .. "]\n") app_lua [app_lua_api.c:626]: app_lua_runstring(): error from Lua: /etc/kamailio/lua/my-script.lua:69: attempt to index global 'sr' (a nil value) I do understand that I should add this to continue with old way: loadmodule "app_lua_sr.so" modparam("app_lua_sr", "register", "sl") However, I would prefer to fix my script to use the NEW way but can't find it. Thanks a lot. Aymeric -- Antisip - http://www.antisip.com

2 1

RTPEngine performance - How far are you pushing your systems
by davy van de moere 07 Jan '20

07 Jan '20

Dear all, Not a direct question on kamailio, but on rtpengine, so feel free to ignore :) I have a farm of rtpengine servers , supporting our set of kamailio's. On recent decent hardware (hp dl360 gen9/10 servers), I typically push my systems upto 2500 channels per machine. Then I tend to see the load of the machine go up in a logarithmic manner. E.g. a load of 3 supports 2500 channels, but when its at 3000 channels, load can be at 8. etc. I have the kernel module running, and I have a good impression it's being used too. No transcoding is going on, no recording, ... I've tried on different OS'es, different network cards, and I always end up at that 2500 channels +-. Is that what I can expect from rtpengine? Or am I really overlooking something? What numbers do you push out of your systems? kind regards and happy 2020!

5 10

Siremis & freeswitch
by John Tuxies 07 Jan '20

07 Jan '20

I have a Debian server running Kamailio and i have all my users regiatering and talking to each other. I do create/remove users from Siremis. I have followed the guide from here: https://www.powerpbx.org/content/kamailio-v5-siremis-v5-debian-v9-mariadb-a… I would like to add freeswitch to the same server and add conference, voicemail, supplementary services for my users. I have given it a try to install freeswitch and tried to configure it accordingly to https://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc But.now the Siremis gui shows a lot of errors and i cannot login any more. Since it is a Virtual machine i reverted it back and it works without freeswitch. How could i integrate freeswitch to my system please?

4 11

opensource VoLTE/IMS Clients like IMSDroid
by Hamid Hashmi 06 Jan '20

06 Jan '20

Dear Users Can you please name some of VoLTE/IMS Clients (Like IMSDroid for Android) for Windows/Linux/MAC to test IMS Network. IMSdroid available clients on the internet are not working with android v8,9 and 10. or if someone can guide me to create a working APK from IMSDroid available code on Github<https://github.com/DoubangoTelecom/imsdroid>. Regards Hamid

1 0

SNI Query
by mahesh b 06 Jan '20

06 Jan '20

Hi All, Am using Kamailio 5.1.9 version. *Below is my tls.cfg* [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = server.key certificate = server.crt ca_list = bundle.crt cipher_list = RSA verify_depth = 9 [client:default] verify_certificate = no require_certificate = no [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile2/btip_172_server_private.key certificate = /root/mahesh_openssl/profile2/btip_172_server_public.crt ca_list = /root/mahesh_openssl/profile2/btip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = btip.172.com [server:10.211.160.172:5061] method = TLSv1+ verify_certificate = yes require_certificate = yes private_key = /root/mahesh_openssl/profile1/ctip_172_server_private.key certificate = /root/mahesh_openssl/profile1/ctip_172_server_public.crt ca_list = /root/mahesh_openssl/profile1/ctip_ca_public.crt cipher_list = RSA verify_depth = 9 server_name = ctip.172.com My Kamailio server ip is 10.211.160.172 i)When i initiate a tls connection from remote server(which is also a kamailio server) say 10.211.160.176 to 10.211.160.172 In the client hello am setting sni name as btip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name btip.172.com for the tls handshake.*// Working as expected* ii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.163 to 10.211.160.172 In the client hello am setting sni name as ctip.172.com => so on 10.211.160.172 side it is picking up the server profile with serve_name ctip.172.com for the tls handshake.*// Working as expected* iii)When i initiate a tls connection from another remote server(Which is also a kamailio server) say 10.211.160.175 to 10.211.160.172 In the client hello am NOT setting sni name => so on 10.211.160.172 side should it pick up the server default profile or the first profile to which IP and port matches ? what i observe from logs is that it is picking up the server profile with server_name ctip.172.com for the tls handshake. I had a look at the code in function tls_lookup_cfg, from the debug prints i understand it is trying to match profile for IP and port if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip))* // IP and port matched* { if(sname && sname->len>0) *//Incoming Client hello dint have sname, so it will hit the else part* { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { return p; *// so it is returning the first profile to which IP and port matched.* } } Am i missing anything or is this a bug ? if in the clienthello there is no sni , what needs to be done to make use of the default profile for the tls handshake ? Or is this something fixed in latest. I just Tried and Modified the code as below, after which it is giving the server default profile when no sni in Incoming Client Hello. if ((p->port==0 || p->port == port) && ip_addr_cmp(&p->ip, ip)) { if(sname && sname->len>0) { if(p->server_name.s && p->server_name.len==sname->len && strncasecmp(p->server_name.s, sname->s, sname->len)==0) { LM_DBG("socket+server_name based TLS server domain found\n"); return p; } } else { if( (type & TLS_DOMAIN_SRV) && (p->server_name.s) ) { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return cfg->srv_default; } else { LM_DBG("Inside %s at %d\n",__FUNCTION__,__LINE__); return p; } } } Regards, Mahesh.B

2 4

Installation issue
by Hassen Bencheikh 06 Jan '20

06 Jan '20

Good morning, I am a telecommunication engineer from the National Establishment of Aerial Navigation (Algeria). First of all i would like to thank you for making the SIP sever Kamailio available. But actually i have got issues during the installation process. I was using the steps described on the link below : https://www.kamailio.org/wiki/install/devel/git The issue was encountered at the step #make all, an error was found, and here is what was displayed on the terminal: *linux-vd3w:/usr/local/src/kamailio-devel/kamailio # make allmake -C src/ all make[1]: Entering directory '/usr/local/src/kamailio-devel/kamailio/src'generating core/autover.h ...CC (gcc) [kamailio] core/ver.oLD (gcc) [kamailio] kamailiowhich: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mysql_config5 in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)which: no mariadb_config in (/usr/local/sbin:/usr/local/bin:/sbin:/bin:/usr/sbin:/usr/bin)CC (gcc) [M db_mysql.so] my_res.oIn file included from my_res.c:24:0:my_cmd.h:27:10: fatal error: mysql.h: No such file or directory #include <mysql.h> ^~~~~~~~~compilation terminated.make[2]: *** [../../Makefile.rules:100: my_res.o] Error 1make[1]: *** [Makefile:511: modules] Error 1make[1]: Leaving directory '/usr/local/src/kamailio-devel/kamailio/src'make: *** [Makefile:30: all] Error 2linux-vd3w:/usr/local/src/kamailio-devel/kamailio #* Actually my knowledge on linux (opensuse leap) that i am using is very limited. Hopefully i will be able to use the server once installed and it will be used for ip recording tests purpose. Thank you for reading this email. Good bye.

5 4

MSRP proxy
by Sergey Safarov 06 Jan '20

06 Jan '20

I want proxy MSRP data on kamailio for SIP call (not websocket). To do this need update connection information for "message" media in SDP. How i can properly do it only for one media (other medias will be proxed using rtpproxy or rtpengine ). Looks as fix_nated_sdp fix for all medias in SDP. Sergey

2 1

pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)]
by Denys Pozniak 06 Jan '20

06 Jan '20

Hello! Looks like $T(reply_reason) does not work well (although $T_reply_reason is ok). Kamailio script: failure_route[TRUNK_FAILURE] { ... xlog("L_WARN","rm=$rm / $T(reply_reason) \n"); ... } 0(8894) ERROR: tmx [t_var.c:696]: pv_parse_t_name(): unknown PV name reply_reason 0(8894) ERROR: <core> [core/pvapi.c:952]: pv_parse_spec2(): pvar "T" has an invalid name param [reply_reason] 0(8894) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)] 0(8894) ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[rm=$rm / $T(reply_reason) https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables $T(name) - pseudo-variable class to access TM attributes The *name* can be: - reply_reason - reply reason [kamailio]$ kamailio -v version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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 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 on 18:04:26 Apr 3 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 1

Tweaking TCP settings
by Dovid Bender 06 Jan '20

06 Jan '20

Hi, First off please forgive my lack of knowledge on how TCP works. We are using Kamailio 5.0.7 and we have an issue where clients are connecting via TCP and their NAT devices are closing up. Because of this we want to send TCP keep alives every so often. 1) When restarting Kamailio it's sends a RST. Is this Kamailio sending it out or is it linux sending it when the application is killed? The issue we have is if say we need to do a restart 2-3 times (yes we should normally do that) then we end up with 3x registrations in the db (since when using db mode if the connection goes away it won't remove the reg from the db (as per - https://kamailio.org/docs/modules/5.0.x/modules/usrloc.html#usrloc.p.handle… ) 2) I haven testing with the following settings. a. tcp_crlf_ping=yes b. tcp_keepcnt = 3 c. tcp_keepidle = 5 With the above I see the TCP keep alives coming in every 75 seconds. If I tcp_keepintvl = 10 then I see TCP keep alives going out from Kamailio to the phone. As per https://www.kamailio.org/wiki/cookbooks/5.0.x/core#tcp_keepintvl it says "Time interval between keepalive probes, when the previous probe failed". Looking at my captures Kamilio sends out the TCP keep alive and gets it back. Is Kamailio not seeing it? With the above it seems to be working the way I want it but I want to make sure that I am doing it right. 3) For the devs on here how hard would it be to implement handle_lost_tcp for DB-Only? TIA and a happy new year to all. Regards, Dovid

2 1

KSR.sdpops.remove_media is not working as expected
by Miteshkumar Thakkar 06 Jan '20

06 Jan '20

Hi, I wants to remove media type "image" from SDP. For that I have tried to use KSR.sdpops.remove_media but it is not working. Not working in the sense that there is no effect of that function on SDP. I am using lua. Kindly find more details below. *Kamailio version*: kamailio 5.2.5 (x86_64/linux) 62d35f *SDP to process:* INVITE sip:xxxxxxxxxx@xxxxxxxxx.com:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK-60823-1-0 From: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060>;tag=xxxxx To: xxxxxxxxxx <sip: xxxxxxxxxx@ xxxxxxxxxx.com:5060> Call-ID: 1-60823(a)172.16.19.64 CSeq: 1 INVITE Contact: sip:xx@xxxxxxx:5060 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 337 v=0 o=zt 53655765 2353687637 IN IP4 xxx.xxx.xxx.xxx s=- c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=image xxxx RTP/AVP udptl t38 a=sendrecv a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxBuffer:262 a=T38FaxMaxDatagram:176 a=T38FaxUdpEC:t38UDPRedundancy m=audio xxxx RTP/AVP 0 a=rtpmap:0 PCMU/8000 *Code snippet:* KSR.sdpops.remove_media("image") logger.log("info", "SDP:" .. headers.get("$sdp(body)")) *Log snippet:* https://pastebin.com/5JZmQUfq One more thing I would like to mention that if I do not use dialog module then the function KSR.sdpops.remove_media works. But I can not avoid using dialog module. Thanks in advance. Mitesh

3 8

dlg_set_timeout error
by Pepelux 03 Jan '20

03 Jan '20

Hi all I'm trying to set a dialog timeout when a call is established: event_route[dialog:start] { if(dlg_set_timeout("10")) { xinfo("10 secs"); } } But when the time expired I obtain this error: Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: CRITICAL: dialog [dlg_timer.c:199]: update_dlg_timer(): Trying to update a bogus dlg tl=0x93947b30 tl->next=(nil) tl->prev=(nil) Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: ERROR: dialog [dlg_hash.c:1267]: update_dlg_timeout(): failed to update dialog lifetime # kamailio -v version: kamailio 5.3.1 (i386/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 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 6.3.0 Any idea? Thanks and Happy New Year

3 6

rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

5 10

rtjson and uac_auth not updating to/from headers
by Ben Merrills 02 Jan '20

02 Jan '20

Hi, I am using rtjson module to do some call routing, as part of that I've added an additional section to the document to include authentication information. I am using the uac_auth function from within failure_route to send authentication on to the uri. This works fine... however. When the initial invite is sent, the To/From headers are updated as per the rtjson document. This isn't the case however when the uac_auth function re-sends the invite with the digest header. The original to/from are being sent. I am using the dialog module restore_mode = auto restore_dlg = 1 I assume there is something I need to call, but none of the RTJSON functions seem to be available within failure_route. Any pointers as to where I am going wrong would be greatly appreciated. Thanks in advance. Ben failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { $avp(auser) = $avp(s:authUser); $avp(apass) = $avp(s:authPass); uac_auth(); t_relay(); exit; } }

2 2

Changing name of the 'versions' table
by Ben Kaufman 31 Dec '19

31 Dec '19

Is it possible to use a table other than 'version' a the version table? i.e. if I'm in a shared DB, and there's an existing table named 'version' with completely different schema, is there a way I can create a table like 'kamailio_version' with the correct schema and data and use it instead? -- Ben Kaufman STATEMENT OF CONFIDENTIALITY: The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain confidential or privileged information. If you are not the intended recipient, or the person responsible for delivering the e-mail to the intended recipient, be advised you have received this message in error and that any use, dissemination, forwarding, printing, or copying is strictly prohibited. Please notify AltiGen Communications immediately at either (888)258-4436 or via email to administrator(a)altigen.com, and destroy all copies of this message and any attachments.

3 2

Calculate call duration
by Duarte Rocha 31 Dec '19

31 Dec '19

Greetings, How sould a call duration be calculated? Let's say the call creation on Kamailio as a proxy has those steps : 1 - Invite is received 2 - Provisional responses 3 - 200 OK is received 4 - ACK to 200 OK is received 5 - BYE is received 6 - 200 OK to BYE is received Should the duration begin to count on step 3 (200-Ok Received) or 4 (ACK received) ? Should it end on the step 5 or 6? ACC module tells me that start time for CDR can either be on dialog creation or confirmation - https://www.kamailio.org/docs/modules/5.2.x/modules/acc.html#acc.p.cdr_star… - What is the dialog creation ? Is it when the INVITE is received ? - According to DIALOG module, a confirmed dialog can be "Waiting for ACK" or "Active Call". Which one is used by acc module? Best Regards,

3 2

Merry Christmas and Happy Holidays!
by Daniel-Constantin Mierla 25 Dec '19

25 Dec '19

Another year getting to its end, the season holidays are ahead, therefore it is no better time than now to express my thanks and greetings to all the friends, developers, supporting companies and community members that made 2019 an outstanding year for Kamailio project. Enjoy the holidays! Merry Christmas! Daniel * Santa is flying Kamailio! * -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 3

topos: how to hide one side only
by Dmitri Savolainen 24 Dec '19

24 Dec '19

Hi all. I have Kamailio with 2 addr: kamailio_external_ip - for client interconnect kamailio_internal_ip - for local media server interconnect Like this: client - kamailio_external_ip - kamailio_internal_ip - media_server "topos" module hide sip headers for both "client" and "media_server" side. Is it true: with topos i can't hide headers only for the "client" side and show them for the "media_server"? I want full header set visible for the "media_server", but only kamailio_external_ip visible for the "client". -- Savolainen Dmitri

2 6

Getting error in auth modules- Installing Kamalio 5.2
by Mahesh Kumar 24 Dec '19

24 Dec '19

Hi Team, Getting error in auth modules during Kamalio 5.2 installation and doing SIP routing, anyone know this why i am getting below error..?? Please support..!! Error Snapshot ======================================== (2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command www_authorize (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1081, column 48: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command www_challenge (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1082, column 29: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command consume_credentials (params 0) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1086, column 22: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command proxy_authorize (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1097, column 41: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command proxy_challenge (params 2) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1098, column 31: unknown command, missing loadmodule? 0(2807) ERROR: <core> [core/cfg.y:3370]: yyparse(): cfg. parser: failed to find command consume_credentials (params 0) 0(2807) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 1102, column 22: unknown command, missing loadmodule? ======================================== Commands in Kamalio.cfg =================================================== route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!www_authorize("192.168.1.67","subscriber")) { www_challenge("$td","0"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("$fd","subscriber")) { proxy_challenge("$fd","0"); break; }; consume_credentials(); # Forward to SDIN S3 rewritehostport("uk.sdin.bt.net:5060"); forward(uk.sdin.bt.net,5060); break; } route[4] { # ----------------------------------------------------------------- # CANCEL or ACK Message Handler # ----------------------------------------------------------------- # Forward to SDIN S3 rewritehostport("uk.sdin.bt.net:5060"); forward(uk.sdin.bt.net,5060); break; } ========================================================

2 1

Kamailio at 36c3, Happy holidays
by Henning Westerholt 24 Dec '19

24 Dec '19

Hello, I will be at the 36c3 [1] congress in Leipzig. So, if you are also there and want to chat about Kamailio or other RTC topics, please contact me. I wish you all Merry Christmas and a Happy New Year! Henning [1] https://events.ccc.de/congress/2019/wiki/index.php/Main_Page -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://gilawa.com<https://gilawa.com/>

1 0

Not working : Using $cfg(line) to print line number
by mahesh b 24 Dec '19

24 Dec '19

Hi All, Below is the syntax am using to print the line number in kamailio configuration file. xlog("L_WARN","Hi Mahesh at line $cfg(line)\n"); Unfortunately , when i do this looks like kamailio is not starting, Any help please regarding what am i missing here. I tried Xlogl as well, which doesnt print the line number by default in my setup. This is the details i had read to understand config file attributes : $cfg(key) - Config File Attributes Attributes related to configuration file. The key can be: - line - return the current line in config - name - return the name of current config file - file - return the name of current config file - route - return the name of routing block Example: send_reply("404", "Not found at line $cfg(line)")

2 2

rtpproxy_manage() doesn't recognize SDP when Content-Length is not present
by Leonid Fainshtein 22 Dec '19

22 Dec '19

Hi, According to RFC-3261 p.20.14, the Content-Length header field must be present only if a stream-base transport protocol (such as TCP) is used. Therefore, in the case of the UDP transport protocol, this field is optional. Unfortunately, the rtpproxy_manage() function in Kamailio v.5.2.5 doesn't recognize the SDP body when the Content-Length header field is not present: Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) WARNING: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} sanity [sanity.c:585]: check_cl(): content length header missing in request Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) INFO: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} <script>: rtproxy ei. $source:81.xxx.xxx.xxx, INVITE sip:098232787@xxx.xxx.xxx.xxx:5060;user=phone $ Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) ERROR: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} rtpproxy [rtpproxy_funcs.c:182]: extract_body(): failed to get the content length in message Dec 22 08:58:34 pbx kamailio[13456]: 4(13488) ERROR: {1 1 INVITE wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64} rtpproxy [rtpproxy.c:2271]: force_rtp_proxy(): can't extract body from the message Dec 22 08:58:34 pbx rtpproxy[1386]: DBUG:GLOBAL:get_command: received command "13498_4 LSEIc0,101 wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64 10.70.227.183 13592 d5k0w3ft-CC-1116-OFC-253;1 as0bf73ed7;1" Dec 22 08:58:34 pbx rtpproxy[1386]: INFO:GLOBAL:handle_command: lookup request failed: session wrjxwyuke5ukvvwwxd00zwtu0ktfuzuf(a)10.18.5.64, tags d5k0w3ft-CC-1116-OFC-253;1/as0bf73ed7;1 not found Dec 22 08:58:34 pbx rtpproxy[1386]: DBUG:GLOBAL:rtpc_doreply: sending reply "0\n" Also, the textops' has_body() function always checks the “Content-Length” presence that is not 100% correct behavior. Just for your information, the problem was discovered in a system where Huawei SoftX3000 is used. It looks like the softswitch removes the “Content-Length” field in an attempt to decrease size of SIP message when it is too long. If the user disables some codecs in their SIP phone then SoftX3000 keeps the “Content-Length” untouched. Should I open a bug regarding this matter? Best regards, Leonid Fainshtein Xorcom Ltd

2 1

Disabling weak SSL Cypher suites
by Arik Halperin 22 Dec '19

22 Dec '19

Hello, How can I disable: TLS_RSA_WITH_RC4_128_SHA (0x5) INSECURE128 TLS_RSA_WITH_RC4_128_MD5 (0x4) INSECURE128 What should I put in cypher_list in order to disable the above? I would also like support TLS 1.2 and TLS 1.3, but remove support for 1.0 and 1.1 Thanks, Arik Halperin

3 5

Need support
by Srikanth Magloori 21 Dec '19

21 Dec '19

Hi Team, We are using your product Kamalio in Call Recording services. We are planning for Performance testing of our recording system. Currently we have bypassed Twilio and Kamailio. Generating Calls using SIPP directly to VOIP. So, we want calls to route through KAMAILIO -> VOIP. So performance can be bit realistic. Any help is greatly appreciated! Thanks Srikanth

4 3

Questions about Presence
by Володимир Іванець 20 Dec '19

20 Dec '19

Hello! I'm testing Kamailio as a presence server with configuration suggested by https://kb.asipto.com/kamailio:presence:k43-blf tutorial. When I set presence module subs_db_mode setting to 0 or 2 (did not test with 1 and 3 seemed to have no problems) I was getting "DEBUG: presence [notify.c:1234]: publ_notify(): Could not find subs_dialog" message after Kamailio received some PUBLISH requests. And NOTIFY is not being generated. It appears to be a mistake. There should be an active watcher according to Expires header values of the previous SUBSCRIBE request and the response to it. I could not find how to get the list of active watchers from Kamailio memory. *Is it possible or it can be done only by adding extra debug line in get_subs_dialog function?* Also I found that after setting pua module db_mode setting to 0, Kamailio is still writing records to presentity table. And it is constantly queried "select `username`,`domain`,`etag`,`event` from `presentity` where `expires`<1576584910 AND `expires`>0 order by username" during the run time. *Is it correct behavior?* Thank a lot!

2 2

Kamailio Product Support
by Simal Mathai 20 Dec '19

20 Dec '19

Hi Team, We are deploying Kamailio 5.3.1 on CentOS 7 for testing with a known carrier in Europe. We would like to get support for the product installation. Regards, Simal Mathai O: +1 214 242 5975 [cid:image001.png@01D13657.67E2B360]<http://www.tekvizion.com/> Please follow us on LinkedIn and Twitter [cid:image002.png@01D13657.67E2B360]<https://www.linkedin.com/company/tekvizion?trk=top_nav_home>[cid:image003.png@01D13657.67E2B360]<https://twitter.com/tekvizion>

2 1

how to detect "failure to add branches"?
by Juha Heinanen 20 Dec '19

20 Dec '19

In route block I call t_on_failure, t_on_branch_failure, and t_on_branch. If branch route fails like this: Dec 15 08:52:19 box /usr/bin/sip-proxy[84346]: ERROR: tm [ut.h:284]: uri2dst2(): failed to resolve "foo.tutpro.com" :unresolvable A or AAAA request (-7) Dec 15 08:52:19 box /usr/bin/sip-proxy[84346]: ERROR: tm [t_fwd.c:1738]: t_forward_nonack(): failure to add branches failure_route or tm:branch-failure event_route is not executed. Is there a way to catch such a failure? -- Juha

3 5

TLS testing (with sipp)
by Sebastian Damm 19 Dec '19

19 Dec '19

Hi, I'm trying to construct an end-to-end encrypted signalling test through our setup. I thought I could use sipp for that, as it supports TLS according to the man page. However, when I try to run it, I get this error: FI_init_ssl_context: SSL_CTX_use_certificate_file failed. I searched the web; however, all similar questions end up without answers. I tried specifying a local key and cert without success. I'd think I should not need a cert for my client, though. Has anyone ever successfully conducted an automated TLS test? I'm open to using a different tool if necessary. Thanks for all hints or examples. Regards, Sebastian -- Sebastian Damm Voice Engineer __________________________________________ sipgate GmbH Gladbacher Straße 74 | 40219 Düsseldorf

4 6

Kamailio SIP TLS Troubleshooting
by Sudhakar Parasher 19 Dec '19

19 Dec '19

Hi Experts. Need your opinion on below part related to kamailio I am using SIP Kamailio module Version for the SIP Registrations from the Client mobile phones using Android and IOS which is Over TLSv1.2 , So in nutshell the SIP Signaling is TLS version 1.2 encrypted and Customer is requesting for the forward of the unencrypted SIP signaling messages especially the SIP Registration messages to their probe solutions without storing the SIP messages locally in the Kamailio Database Question :Would it not be possible for Kamailio imply “mirror” all the SIP messages related to all SIP REGISTER dialogs (after removing TLS for inbound messages and prior to adding TLS for outbound message) to a designated IP Address and Port? No need for local database storage. No need for encapsulation. No mirroring of other messages/dialogs. Cheers Sudhakar

4 10

[dispatcher] freeing already freed pointer, called from core: core/xavp.c: xavp_free
by Denys Pozniak 19 Dec '19

19 Dec '19

Hello! The configuration below leads to a CRITICAL message in the console. Please let me know how to avoid this issue. event_route[htable:mod-init] { route(AUTORUN); ... } route[AUTORUN] { if ( ds_select("9", "4") ) { xlog("L_WARN", "[$cfg(route)] *ds_select* $xavp(_dsdst_[0]=>uri) $xavp(_dsdst_[1]=>uri) $xavp(_dsdst_[2]=>uri)\n"); } ... } dispatcher.list # group 9 9 sip:192.168.20.125:5070 9 sip:192.168.20.160:5070 9 sip:192.168.20.7:5070 9 sip:192.168.20.107:5070 9 sip:192.168.20.190:5070 0(6088) DEBUG: dispatcher [dispatch.c:2054]: ds_manage_routes(): set [9] 0(6088) DEBUG: dispatcher [dispatch.c:2158]: ds_manage_routes(): using alg [4] hash [0] 0(6088) DEBUG: dispatcher [dispatch.c:2199]: ds_manage_routes(): selected [4-9-2/0] <sip:192.168.20.190:5070> 0(6088) DEBUG: dispatcher [dispatch.c:2218]: ds_manage_routes(): using first entry [9/0] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/1] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/2] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/3] 0(6088) DEBUG: dispatcher [dispatch.c:2272]: ds_manage_routes(): using default entry [9/4] 0(6088) DEBUG: dispatcher [dispatch.c:2016]: ds_select_dst_limit(): selected target destinations: 5 0(6088) WARNING: <script>: [AUTORUN] *ds_select* sip:192.168.20.190:5070 sip:192.168.20.107:5070 sip:192.168.20.7:5070 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list 0x7fe5f7d4f420 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list 0x7fe5f7d4faf0 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4faf0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f168), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f378), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f580), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f790), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f9a0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/receive.c:457]: receive_msg(): cleaning up ... version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, 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 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 on 19:29:40 Jun 20 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 2

kamailio 5.2.5 wrong send socket on a mhomed proxy.
by Vitalii Aleksandrov 19 Dec '19

19 Dec '19

Hi, kamailio 5.2 branch has a commit 09ac3e47ef79dbd599b7dec5b84ae3b792f025ed cherry picked from master branch. It makes sense for master because get_send_socket2() behavior has changed and function became started and also there were some patches to propagate connection it to uac structure. 5.2 kamailio takes the first interface for TCP/TLS/WS connections and then finds existing tcp connection matched by ip:port, but when the patch is applied connection lookup fails and kamailio can't forward in-dialog messages.

2 1

How to reduce re-INVITE time in Kamailio
by Anuran Barman 18 Dec '19

18 Dec '19

Hi, I am using Kamailio for my SIP Phone app. My main SIP client is the iOS app. Till now everything is working fine regarding the calling. But the problem is when the app is killed I am not able to pick up incoming call. The thing is, when app is killed I am sending PushKit VOIP message to my ios app to wake it up. App is waking up. But by the time app is waking up, the original INVITE message got missed up. So according to SIP protocol it will re-INVITE after some time. Currently if I keep my app open for 10-15 secs after it wakes up, I am able to get the re-INVITE and everything works fine just as it was supposed to. But waiting 10-15 sec on the app is not practical in any sense. So my question is there any way I can reduce the re-INVITE time to like 5/6 seconds so that as soon as the app open it will wait for 5/6 seconds (practical upto some extent) and it will get the call ? Or what kamailio is using as the interval time is a SIP standard and can not be changed? If possible to do so, how can I do that? Please help.

6 11

kamcmd stuck
by Andrew Chen 17 Dec '19

17 Dec '19

Hi everyone, Have anyone experienced kamcmd getting stuck and the only way to fix it was to restart the process? Running 5.2.5 Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

OPTION pings with dispatcher to asterisk box
by Karsten Horsmann 16 Dec '19

16 Dec '19

Hello all! i am using kamailio 5.3.1-4 on CentOS and i am using dispatcher to ping his targets. Now i have someone with asterisk that answers with 401 or 404 (depends on asterisk version). So i tried code=401;code=404 as aditional attrs. Seems not to work. Also i tired to change the modparam from class=2 to this: modparam("dispatcher", "ds_ping_reply_codes", "class=2,code=401,code=404") Is that a "bug" or a hidden feature, or i missed something? Btw - Have a nice weekend! SELECT * FROM dispatcher WHERE id=1 \G *************************** 1. row *************************** id: 1 setid: 1100 destination: sip:SOMEWHERE;transport=tls flags: 0 priority: 0 attrs: socket=tls:MY.IP.AD.RR:5061;ping_from=sip:MY.SIP.DOMAIN;code=401;code=404 description: UNTRUSTED "SET": { "ID": 1100, "TARGETS": [{ "DEST": { "URI": "sip:SOMEWHERE;transport=tls", "FLAGS": "IP", "PRIORITY": 0, "ATTRS": { "BODY": "socket=tls:MY.IP.AD.RR:5061;ping_from=sip:MY.SIP.DOMAIN;code=401;code=404", "DUID": "", "MAXLOAD": 0, "WEIGHT": 0, "RWEIGHT": 0, "SOCKET": "tls:MY.SIP.DOMAIN:5061" }, "LATENCY": { "AVG": 48768.816406, "STD": 139490.953125, "EST": 43359.566406, "MAX": 568876, "TIMEOUT": 8 } } }] } }] #!ifdef WITH_MYSQL modparam("dispatcher", "db_url", "DBURL") modparam("dispatcher", "table_name", "dispatcher") #!endif #!ifndef WITH_MYSQL modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") #!endif modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "<sip:ping@FANCY-COMPANY-WEBNAME>") modparam("dispatcher", "ds_ping_interval", 30) modparam("dispatcher", "ds_probing_mode", 1) modparam("dispatcher", "ds_probing_threshold", 2) modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") modparam("dispatcher", "ds_ping_reply_codes", "class=2,code=401,code=404") modparam("dispatcher", "ds_ping_latency_stats", 1) https://community.asterisk.org/t/asterisk-responds-with-404-to-options-even… Dec 13 17:25:46 siptrunk3 /usr/sbin/kamailio[25115]: INFO: <script>: Sending message 'OPTIONS sip:SOMEWHERE;transport=tls SIP/2.0 Via: SIP/2.0/TLS MY.IP.AD.RR:5061;branch=z9hG4bK2527.825671c0000000000000000000000000.0 To: <sip:SOMEWHERE;transport=tls> From: <sip:MY.SIP.DOMAIN>;tag=f9bf2579cb456ea8d260d4aabc8248ef-9ca844ad CSeq: 10 OPTIONS Call-ID: 01cb6e272da7b0dd-25115(a)MY.IP.AD.RR Max-Forwards: 70 Content-Length: 0 User-Agent: SBC-OS ' Dec 13 17:25:46 siptrunk3 /usr/sbin/kamailio[25127]: INFO: <script>: Received message 'SIP/2.0 404 Not Found Via: SIP/2.0/TLS MY.IP.AD.RR:5061;branch=z9hG4bK2527.825671c0000000000000000000000000.0;received=MY.IP.AD.RR From: <sip:MY.SIP.DOMAIN>;tag=f9bf2579cb456ea8d260d4aabc8248ef-9ca844ad To: <sip:SOMEWHERE;transport=tls>;tag=as54adca41 Call-ID: 01cb6e272da7b0dd-25115(a)MY.IP.AD.RR CSeq: 10 OPTIONS Server: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ' -- Cheers *Karsten Horsmann*

3 6

Reject tls invites without a=crypto sdp
by Karsten Horsmann 16 Dec '19

16 Dec '19

Hi, I use Kamailio 5.3.1 with rtpengine to offer an siptrunk endpoint for my customers. I observe that someone of them use tls to encrypt signaling but forgotten to encrypt rtp. I want to reject this invites. Are there any hints how to do this? Thought about reading the sdp and search for a=crypto line and if not send reply with (what code ever will be good for that). Cheers Karsten

2 4

Removing extra Connection information in SDP
by Kjeld Flarup 16 Dec '19

16 Dec '19

Hello I have a problem with some phone numbers at Telenor. Our door telephone does not reply with an ACK on the 200 OK. All I found was a weird SDP response with two "connection information" lines Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): BroadWorks 307119541 1 IN IP4 194.247.61.31 Session Name (s): - Connection Information (c): IN IP4 194.247.61.31 Time Description, active time (t): 0 0 Session Attribute (a): sendrecv Media Description, name and address (m): audio 23116 RTP/AVP 8 101 Connection Information (c): IN IP4 194.247.61.31 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-15 Media Attribute (a): maxptime:40 Media Attribute (a): bsoft:1 image udptl t38 Media Attribute (a): sendrecv Media Attribute (a): silenceSupp:off - - - - Media Attribute (a): setup:actpass To debug, I would like to remove one of the "Connection Information (c): IN IP4 194.247.61.31", to see if this is what confuses the door telephone. But how do I do that? I tried with this but nothing happeded onreply_route[DOOR_MANAGE_REPLY] { if (is_method("INVITE|SUBSCRIBE")) { sdp_remove_line_by_prefix("c=IN IP4 194"); } return 1; } -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen -www.liberalismen.dk <http://www.liberalismen.dk>

2 1

using tls targets in dispatcher
by Narasimhaiah, Vinod Maralenahalli 15 Dec '19

15 Dec '19

Hi I am using kamailio server to forward the request to a new kamailio-server which listens only on TLS. *problem: Not able to specify the dispatcher with tls protocol* I am using the below configuration in dispatcher.list kamailio version:4.4.2 *# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)1 sip:mynetwork.com <http://mynetwork.com>;transport=tls 8 1 socket=tls:mynetwork.com:5328 <http://mynetwork.com:5328>* and if I specify the destination as just sip:mynetwork.com:5328 with out attrs and transport=tls if is falling back to UDP. Thank you vinod.M.N

2 1

Kamailio dispatcher to use tls targets.
by vinod mn 15 Dec '19

15 Dec '19

Hi I am using kamailio server to forward the request to a new kamailio-server which listens only on TLS. *problem: Not able to specify the dispatcher with tls protocol* I am using the below configuration in dispatcher.list kamailio version:4.4.2 *# setid(integer) destination(sip uri) flags (integer, optional), priority(int,opt), attrs (str,optional)1 sip:mynetwork.com <http://mynetwork.com/>;transport=tls 8 1 socket=tls:mynetwork.com:5328 <http://mynetwork.com:5328/>* and if I specify the destination as just sip:mynetwork.com:5328 with out attrs and transport=tls if is falling back to UDP. Thank you vinod.M.N

1 0

using a pseudo-variable as a parameter to a module function
by Michael Iedema 14 Dec '19

14 Dec '19

Hi all, I’m trying to write a basic Kamailio module from scratch and have run into an area where I’m in need of a couple pointers (pun not intended). :-) I have the module loading correctly and can call a function in my routing script like this: mymod_getinfo(“5551234”); This, for now, just triggers a print of the submitted parameter. What I’d like to do next is have it accept and print the From URI in configuration script like so: mymod_getinfo(“$fU”); For now, this prints the string literal of “$fU” and I’m trying to extract the value of the referenced variable and print that. The example module “print” does not involve this concept of pseudo-variables. The skeleton module example in the wiki also does not mention variables. I’ve looked into other modules to see how it works but they all seem to have a slightly different handling of this task. Can someone point me to an example in the modules which accomplishes this? What I’ve been searching for is a replace_all function which accepts a string containing pseudo-variables and returns a new string with them all replaced with their corresponding values. Thanks in advance and apologies if I’ve overlooked a relevant piece of developer documentation or if I’m looking at this incorrectly. Regards, -Michael

3 3

lookup(aliases) issues with 5.2
by igor.potjevlesch＠gmail.com 13 Dec '19

13 Dec '19

Hello! I have notice some issues with lookup(aliases) since I upgraded to 5.2 from 5.1: First with kamctl: If this default command is executed: ctl_cmd_run ul.add "$USRLOC_TABLE" "$OSERUSER@$OSERDOMAIN" "$2" \ "$UL_EXPIRES" "$DEFAULT_Q" "$UL_PATH" "$UL_FLAGS" "$BR_FLAGS" "$ALL_METHODS" as $ALL_METHODS is not defined, there is a huge value in methods and then lookup(aliases) fails with -2. I have to update kamctl script and put -1 ("-1") instead of the variable. So, now, methods is a NULL value in DB and my alias is not ignored. But I use to make my alias like this: sip:number@blabla_1.local Which is no longer works: ERROR: registrar [common.c:62]: extract_aor(): failed to parse AoR [sip:number@blabla_1.local] Any idea of the difference since 5.2 with this behaviour? Without the underscore, the AoR seems to be valid but I need to support underscore as before. Regards, Igor.

4 16

t_newtrans() README issue
by Juha Heinanen 12 Dec '19

12 Dec '19

I noticed this in t_newtrans() README: Note that any flag operations (e.g. for accounting) after this function has been called will be ignored. You can use the the tmx module function t_flush_flags() to flush the altered flags to the already created transaction. The text is wrong. I have for ages set bflags after calliing t_newtrans() and the sets are NOT ignored. Here is an example: In route blocks: Dec 12 08:41:53 /usr/bin/sip-proxy[231131]: INFO: ********* called t_newtrans() Dec 12 08:41:53 /usr/bin/sip-proxy[231131]: INFO: ************ set bflag 9 In failure route block: Dec 12 08:41:58 /usr/bin/sip-proxy[231195]: INFO: ************ bflag 9 is set -- Juha

3 2

S-CSCF tear down call while deleting contact
by Liu, Charles 12 Dec '19

12 Dec '19

Hi All, I need some expert help to trouble shoot a problem I have seen, S-CSCF send BYE to tear down a call. I have make some progress to trace the problem to impurecord.c in ims_usrloc_scscf module, _c->first_dialog_data !=0 while try to delete an expired contact. I need some help to find out why _c->first_dialog_data is not 0 for an expired contact. To reproduce the problem, I began with the following for user1. Contact id 4098, 4114, and 4115 would expire and a call was using id 4116. When the time to delete contact, somehow scscf mistakenly think id 4098 still has a dialog, but the dialog was with id 4116 which was NOT expired. Therefore, S-CSCF send BYE to tear down the call, so end user see call drop. Please note, if id 4098 never make calls before, first call was working fine. But once a call was make, you will see this problem for next call. it seem S-CSCF did not clean up the dialog_data from last call properly and some data still hanging with the contact caused this problem. any insight how to fix or avoid this would be appreciated. id contact params path received user_agent expires callid 2192 sip:user2@192.168.164.136:62837;rinstance=bca96a7c77f53916;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:10 UPIM_gl22a13wlo47aeHRQ.. 4098 sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:49:21 ocl7UYxJLo8TnMw2p2-6NQ.. 4114 sip:user1@192.168.164.9:61946;rinstance=65afabdb886de0a7;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:11 GZrZHDyL3ojhDlLDPpJnvQ.. 4115 sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:13 DCKsqvUjZ1dAfJLfJ3nOhA.. 4116 sip:user1@192.168.164.9:61946;rinstance=38cc0c6188509aea;transport=UDP NULL <sip:term@pcscf.ims.mnc001.mcc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-12 15:50:15 gF3fphqyrSbQwaN0eW4JLw.. 13(18) DEBUG: ims_usrloc_scscf [udomain.c:384]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - FINISHED *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:user1@ims.mnc001.mcc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:user1@192.168.164.9:61946;rinstance=074e2e926212237b;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:312]: mem_delete_ucontact(): first dialog is 0! 13(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:user1@ims.mnc001.mcc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:user1@192.168.164.9:61946;rinstance=7ab86c5e85643767;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:314]: mem_delete_ucontact(): first dialog is not 0 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:317]: mem_delete_ucontact(): Going to tear down dialog with info h_entry [1081] h_id [2579] 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f803f803108 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2579 found on entry 1081 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7f803f803108 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog Thanks, --Charles CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 0

Kamailio delayed reponse in case of database load
by Ali Taher 12 Dec '19

12 Dec '19

Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I'm using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16 Following is the routing logic I'm using: route{ if (is_method("INVITE")) { $var(inc) = $(ct{param.value,tgrp}); xlog("L_INFO","CONTACT : $var(inc)" ); xlog("L_INFO","from number : $rU" ); if($var(inc)!=0) route(customer); route(block); } } route[customer] { sql_query("cb", "select caid from customeridentification where trunk='$var(inc)'", "ra"); $var(cid)=$dbr(ra=>[0,0]); sql_result_free("ra"); xlog("L_INFO","cid: $var(cid) \n"); sql_query("cb", "select routetablename from carrieraccount where caid=$var(cid)", "ra"); $var(tid)=$dbr(ra=>[0,0]); sql_result_free("ra"); if($var(cid)!=0) route(query); else route(block); } route[query] { #sql_query("cb", "select supplierid from route_$var(cid) where prefix @>'$rU' order by prefix limit 1", "ra"); sql_query("cb", "select get_routing('$var(tid)','$rU')", "ra"); $var(sup)=$dbr(ra=>[0,0]); sql_result_free("ra"); xlog("L_INFO","supplieris: $var(sup) \n"); #$var(x)=$(var(sup){s.replace,|,}{s.len})/4; $var(a)=$(var(sup){s.len}); $var(z)=$(var(sup){s.replace,|,}); $var(b)=$(var(z){s.len}); $var(suplen)=$var(a) - $var(b); if($var(sup)!=0) route(send); else route(block); } route[send] { xlog("L_INFO","Inside Send \n"); $var(i)=0; append_to_reply("Contact:"); while($var(i)<$var(suplen) + 1) { $var(s)=$(var(sup){s.select,$var(i),|}); $var(s)=$(var(s){s.select,0,;}); append_to_reply("<sip:$rU;tgrp=$(var(s){s.ltrim})@ xxx.xxx.xxx.xxx:5060;user=phone>"); if($var(i)<$var(suplen)) append_to_reply(","); $var(i)=$var(i)+1; } append_to_reply("\r\n"); sl_send_reply("300","Multiple Choices"); exit; } route[block] { xlog("L_INFO","INSIDE BLOCK" ); append_to_reply("Contact:<sip:$rU;tgrp=999999@xxx.xxx.xxx.xxx:5060;user=phone>\r\n"); sl_send_reply("300","Multiple Choices"); exit; } How can we enhance the Kamailio response in case of load on the postgresql server. Regards, Ali Taher

2 4

xhttp_prom Metric Naming Issue
by ENGIN ESIN 12 Dec '19

12 Dec '19

Hi everyone, I am trying to use xhttp_prom and there are some issues that i want to share with you. Metric Naming: I get warning at prometheus side like as follows, WARN[0659] append failed err="no token found" source="scrape.go:648" target="{__address__="192.168.42.1:9080", __metrics_path__="/metrics", __scheme__="http", instance="192.168.42.1:9080", job="vmm"}" With respect to prometheus documentation, metric names need to fullfill following naming requirement -> "It must match the regex [a-zA-Z_:][a-zA-Z0-9_:]*" (source: https://prometheus.io/docs/concepts/data_model/#metric-names-and-labels) However, kamailio gives unmatch outputs as follows because they have '-' char, (related discussion: https://github.com/prometheus/prometheus/issues/3154 ) * kamailio_usrloc_location-contacts 0 1576046310669 * kamailio_usrloc_location-expires 0 1576046310669 * kamailio_usrloc_location-users 0 1576046310669 whit that naming, prometheus can not parse values of any metrics. Documentation: Example 1.1. Set xhttp_prom_buf_size parameter ... modparam("xhttp", "xhttp_prom_buf_size", 1024) ... For that example, it should be as follows, ... modparam("xhttp_prom", "xhttp_prom_buf_size", 1024) ... [http://www.turkcell.com.tr/downloads/bireysel/img/Tcelldis.gif] <http://turkcell.li/iyaani> Bu elektronik posta ve onunla iletilen butun dosyalar sadece gondericisi tarafindan almasi amaclanan yetkili gercek ya da tuzel kisinin kullanimi icindir. Eger soz konusu yetkili alici degilseniz bu elektronik postanin icerigini aciklamaniz, kopyalamaniz, yonlendirmeniz ve kullanmaniz kesinlikle yasaktir ve bu elektronik postayi derhal silmeniz gerekmektedir. TURKCELL bu mesajin icerdigi bilgilerin do?rulu?u veya eksiksiz oldugu konusunda herhangi bir garanti vermemektedir. Bu nedenle bu bilgilerin ne sekilde olursa olsun iceriginden, iletilmesinden, alinmasindan ve saklanmasindan sorumlu degildir. Bu mesajdaki gorusler yalnizca gonderen kisiye aittir ve TURKCELLin goruslerini yansitmayabilir Bu e-posta bilinen butun bilgisayar viruslerine karsi taranmistir. ________________________________ This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient you are hereby notified that any dissemination, forwarding, copying or use of any of the information is strictly prohibited, and the e-mail should immediately be deleted. TURKCELL makes no warranty as to the accuracy or completeness of any information contained in this message and hereby excludes any liability of any kind for the information contained therein or for the information transmission, reception, storage or use of such in any way whatsoever. The opinions expressed in this message belong to sender alone and may not necessarily reflect the opinions of TURKCELL. This e-mail has been scanned for all known computer viruses.

2 1

Please Help
by Sharath 12 Dec '19

12 Dec '19

Dear Team, If we want add a free-pbx as a trunk then what are configuration we need to do in admin panel of siremis. * SHARATH. O.M.** *

2 1

Kamailio delayed reponse in case of database load
by Ali Taher 10 Dec '19

10 Dec '19

Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I'm using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options ----------------------- 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16

4 6

authcheck and fd variable
by Robert Lieback 10 Dec '19

10 Dec '19

Hello, i'm looking into kamalio (btw: thanks to asipto for the great book) and i hope someone can push me in the right direction. I have a question/problem with some lines in the default config (kamailio.cfg) around line 754: ... if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } ... auth_check() and auth_challenge doesn't work, but i've provided the correct credentials. When i replace "$fd" with domain "voip.XXXXXX.de" it works. This domain is also defined as alias in the config and all sip messages i send to the server have a correct from header with this domain. First step for me would be to look whats stored in $fd, but how can i output the content of $fd? Cause log(2, "$fd") or xlog("$fd") doesn't work and throws errors about unknown variables. Thanks, Robert Lieback

2 2

Kamailio management group updates
by Daniel-Constantin Mierla 09 Dec '19

09 Dec '19

Hello, Jesus Rodrigues and Inaki Baz Castillo changed their work roles a while ago, so they no longer deal with Kamailio and they do not expect to do it in the near future. To keep the Management group with members connected as much as possible with the project, Fred Posner and Victor Seva were proposed to replace them. For the new members in the public lists: the management group of Kamailio is not involved in the daily development or administration of the project -- all those are done on public mailing lists or kadmin group (internal infrastructure administration) -- management team is supposed to take the decisions on sensitive situations for the evolution of the project. Should anyone have opinions about these changes, let's discuss on sr-users mailing list. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

duplicated registrations
by Liu, Charles 08 Dec '19

08 Dec '19

Hi All, Does this look correct? I am using Zoiper client and did multiple registration/re-registration, but I saw duplicated contacts in the database for the same user/device(alice). I was expecting to see only one. Is this a bug? How do I prevent this happening? "select * from scscf.contact" id contact params path received user_agent expires callid 49 sip:bob@192.168.10.201:60944;rinstance=44b4024df17bd284;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 r7jCYp7JuVbA8QjocnouLw.. 64 sip:alice@192.168.10.241:60383;rinstance=e850d3ed0605bcd3;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:11 EHLOADkLUEe8d0CMpdLL9A.. 142 sip:alice@192.168.10.241:60383;rinstance=bc755d87cc438890;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:31:51 P1rBkHGUVdv7oqW3_UWQPg.. 143 sip:alice@192.168.10.241:60383;rinstance=146303cf7501a125;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:15 etUEoAGSM4q4e_39wFKa7w.. 144 sip:alice@192.168.10.241:60383;rinstance=e0a260b7b78ecf6b;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:17 WUtIYeGlBZmHD3TGklmKCQ.. 145 sip:alice@192.168.10.241:60383;rinstance=2681245431c4e4e4;transport=UDP NULL <sip:term@pcscf.mnc001.mnc001.3gppnetwork.org;lr> Zoiper rv2.10.3.2 2019-12-08 10:32:19 HNZPyeO173jfscKYNOT0bw.. Thanks, --Charles CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

ul.dump empty
by Liu, Charles 07 Dec '19

07 Dec '19

Hi All, I am running Kamailio v5.2.4 with P/S/I-cscf instances on 3 different machines with default IMS example configurations. Kamcmd ul.dump show empty, but I can make calls... any idea what is the problem? root@pcscf:/# kamcmd ulpcscf.status { Domain: location Size: 512 AoRs: { Stats: { Records: 7 Max-Slots: 6 } } } root@pcscf:/# kamcmd ul.dump { Domains: { } } CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 1

Pike module never release IP
by Andrew Chen 06 Dec '19

06 Dec '19

Hi all, Have anyone ever experienced pike module not releasing the IP after "remove_latency" period? We ran a load test and this is what happened to one of the two paired nodes and I had to restart kamailio process to release it. I'm running Kam 5.2.5 Thanks. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 6

How can I make calls between the two domains?
by Yanko 06 Dec '19

06 Dec '19

Hello: I have two virtual machines with two kamailo domains VM1: Domain: 172.16.16.1 Users:2000, 2001, 2002... VM2: Domain: 172.16.16.2 Users:3000, 3001, 3002... In each domain I can make calls. For example, user 2000 can call 2001. User 3000 can call 3001. How can I make calls between the two domains? What modules do I need to load? How to configure the kamailio.cfg file? Thanks, YANKO.

2 1

Scripts still valid?
by David Villasmil 05 Dec '19

05 Dec '19

Hello guys, Is this still valid? https://github.com/kamailio/kamailio/blob/master/misc/examples/scripts/ctd.… Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 4

call disconnected between 15 to 45 minutes
by Liu, Charles 05 Dec '19

05 Dec '19

Hi all, I am still looking for some help to debug why S-CSCF send BYE to caller and callee after a call was connected. from wireshark, I just saw S-CSCF send BYE to both caller and callee. That matched the S-CSCF debug output. But the question is why S-CSCF delete contact and send BYE to a active call? There was no timer expire or something like that in the log... Anyone can help? Thanks, --Charles 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e6d0 alloc'ed from cdp: diameter_msg.c: AAATranslateMessage(484) 19(24) DEBUG: tm [t_lookup.c:1618]: t_lookup_ident(): transaction found 19(24) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) called from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb093db6000, 6872) returns address 0x7fb09414d450 frag. 0x7fb09414d418 (size=6872) on 1 -th hit 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=767 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=769 a=26 n=xlog 19(24) DEBUG: <script>: saa_return code (for scscf_save on register) is 1 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=783 a=21 n=switch 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=772 a=26 n=xlog 19(24) DEBUG: <script>: SAR success - 200 response sent from module 19(24) ERROR: *** cfgtrace:failure_route=[REG_SAR_REPLY] c=[/etc/kamailio/kamailio.cfg] l=773 a=2 n=exit 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09414d450), called from tm: t_reply.c: free_faked_req(948) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09414d418 alloc'ed from core: core/sip_msg_clone.c: sip_msg_shm_clone(496) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e978), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(94) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e940 alloc'ed from ims_registrar_scscf: save.c: save(1414) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09413e0b0), called from ims_registrar_scscf: reply.c: free_contact_buf(830) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09413e078 alloc'ed from ims_registrar_scscf: reply.c: build_contact(443) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094145218), called from ims_registrar_scscf: reply.c: free_contact_buf(835) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb0941451e0 alloc'ed from ims_registrar_scscf: reply.c: build_contact(435) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb094147dc0), called from ims_registrar_scscf: cxdx_sar.c: free_saved_transaction_data(98) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094147d88 alloc'ed from ims_registrar_scscf: save.c: save(1396) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb09415b4c8), called from cdp: transaction.c: cdp_free_trans(182) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb09415b490 alloc'ed from cdp: transaction.c: cdp_add_trans(110) 19(24) DEBUG: <core> [core/mem/q_malloc.c:475]: qm_free(): qm_free(0x7fb093db6000, 0x7fb0941442a0), called from cdp: transaction.c: cdp_free_trans(183) 19(24) DEBUG: <core> [core/mem/q_malloc.c:519]: qm_free(): freeing frag. 0x7fb094144268 alloc'ed from cdp: transaction.c: cdp_add_trans(105) 13(18) DEBUG: ims_usrloc_scscf [ims_usrloc_scscf_mod.c:481]: timer(): Syncing cache 13(18) DEBUG: ims_usrloc_scscf [udomain.c:291]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - START *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 95 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP) which expires in -60.000000 seconds and has a ref count of 0 (state: Contact with delayed delete) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:309]: mem_timer_udomain(): contact in state CONTACT_DELATED_DELETE is about to be deleted13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 464 = [sip:alice@ims.mnc001.mnc001.3gppnetwork.org] (sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP) which expires in 50.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:301]: mem_timer_udomain(): We have a [3gpp=0] contact in the new contact list in slot 478 = [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] (sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP) which expires in 56.000000 seconds and has a ref count of 1 (state: Contact valid) 13(18) DEBUG: ims_usrloc_scscf [udomain.c:353]: mem_timer_udomain(): *** mem_timer_udomain - checking contacts - FINISHED *** 13(18) DEBUG: ims_usrloc_scscf [udomain.c:359]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - START *** 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:alice@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:alice@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:alice@192.168.164.203:44640;rinstance=e35e9704884c3208;transport=UDP, Ref [1] (expires in 50 seconds) (State: 0) 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:372]: process_impurecord(): Checking validity of IMPU: <sip:bob@ims.mnc001.mnc001.3gppnetwork.org> registration subscriptions 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:404]: process_impurecord(): Public Identity sip:bob@ims.mnc001.mnc001.3gppnetwork.org, Barred: [0], State: [registered], contacts [1], 3gppcontacts [0] 3(18) DEBUG: ims_usrloc_scscf [impurecord.c:441]: process_impurecord(): Contact #0 - sip:bob@192.168.164.248:50448;rinstance=9e742618a5f6bcae;transport=UDP, Ref [1] (expires in 56 seconds) (State: 0) 3(18) DEBUG: ims_usrloc_scscf [udomain.c:384]: mem_timer_udomain(): *** mem_timer_udomain - checking IMPUs - FINISHED *** 3(18) DEBUG: ims_usrloc_scscf [udomain.c:406]: mem_timer_udomain(): deleting contact [sip:bob@ims.mnc001.mnc001.3gppnetwork.org] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:598]: delete_scontact(): Deleting contact: [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [usrloc_db.c:370]: db_delete_ucontact(): Deleting ucontact [sip:bob@192.168.164.248:50448;rinstance=989f943bd5098a11;transport=UDP] 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:310]: mem_delete_ucontact(): Checking if dialog_data is there and needs to be torn down 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:314]: mem_delete_ucontact(): first dialog is not 0 13(18) DEBUG: ims_usrloc_scscf [impurecord.c:317]: mem_delete_ucontact(): Going to tear down dialog with info h_entry [1694] h_id [3889] 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=3889 found on entry 1694 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7fb09414ce40 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 18) called from ims_dialog: dlg_req_within.c: build_extra_hdr(229) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 24) returns address 0x7fb09b640c30 frag. 0x7fb09b640bf8 (size=64) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) called from ims_dialog: dlg_req_within.c: build_dlg_t(81) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 248) returns address 0x7fb09b74b100 frag. 0x7fb09b74b0c8 (size=248) on 1 -th hit 13(18) DEBUG: <core> [core/mem/q_malloc.c:367]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) called from core: core/parser/parse_rr.c: do_parse_rr_body(68) 13(18) DEBUG: <core> [core/mem/q_malloc.c:412]: qm_malloc(): qm_malloc(0x7fb09b5ae010, 72) returns address 0x7fb09b74b048 frag. 0x7fb09b74b010 (size=80) on 1 -th hit 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7fb09414ce40 with 1 -> 3 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

2 2

Question regarding priority in dispatcher module
by Floimair Florian 05 Dec '19

05 Dec '19

Dear all! After reading the documentation on dispatcher module I still have a question that isn't really documented. Regarding the priority value of an entry in a dispatcher list: Is a lower numerical priority value regarded as higher priority or is it the other way around? Hopefully someone can help me to clarify this. With best regards FLORIAN FLOIMAIR Software Development Commend International GmbH Saalachstrasse 51 5020 Salzburg, Austria commend.com LG Salzburg / FN 178618z

2 2

Re: [SR-Users] RTPengine with rejected RE-INVITE when port is changing
by Daniel Tryba 05 Dec '19

05 Dec '19

On Thu, Dec 05, 2019 at 09:37:51AM +0000, Laurent Schweizer wrote: > Hello, > > I already see old post about this : > https://opensips.org/pipermail/users/2014-November/030451.html > > but I???m interested to know if now they is a solution ???? > > so the issue is a RE-INVITE rejected (415 Unsupported Media or 488 ) and the RTP port is changing. > Is that case the old RTP relay (I???m using RTPengine) is destroyed after 30sec. > Any way to restore them ? I'm confused: - opensips != kamailio - rtpproxy != rtpengine But RFC 3261 states (page 16): During the session, either Alice or Bob may decide to change the characteristics of the media session. This is accomplished by sending a re-INVITE containing a new media description. This re- INVITE references the existing dialog so that the other party knows that it is to modify an existing session instead of establishing a new session. The other party sends a 200 (OK) to accept the change. The requestor responds to the 200 (OK) with an ACK. If the other party does not accept the change, he sends an error response such as 488 (Not Acceptable Here), which also receives an ACK. However, the failure of the re-INVITE does not cause the existing call to fail - the session continues using the previously negotiated characteristics. Full details on session modification are in Section 14. So if this happens with kamailio/rtpengine that is a bug in (I presume) rtpengine.

3 2

siputils: how does sip_p_charging_vector generate the header
by George Diamantopoulos 05 Dec '19

05 Dec '19

Hello, I've been looking at the sip_p_charging_vector() function of siputils. From the documentation it is not evident how the PCV header is generated. Are the exported $pcv pseudovariables writable, so that the function canl use those when generating the header? Thanks! BR, George

2 3

anycast + dmq_usrloc
by Nuno Miguel Reis 05 Dec '19

05 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replication the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: *anycast node that actually gets the SIP REGISTER:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *anycast node that gets the USRLOC update via KDMQ:* $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } *the actual KDMQ message:* *KDMQ sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060> SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch=z9hG4bK6034.290e4186000000000000000000000000.0 To: <sip:usrloc@10.100.100.21:5060 <http://sip:usrloc@10.100.100.21:5060>> From: <sip:usrloc@10.100.100.22:5060 <http://sip:usrloc@10.100.100.22:5060>>;tag=3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 <0cfde4d32a31addc-8875(a)127.0.0.1> Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip:2.2.2.2:31171 <http://2.2.2.2:31171>","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0}* -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra mobile: +351 918519145 -- get my PGP key @ http://pgp.dei.uc.pt:11371/pks/lookup?op=get&search=0x750296EECEAD2B6D

2 2

anycast + dmq_usrloc
by Nuno Miguel Reis 04 Dec '19

04 Dec '19

Hi guys. I'm setting up an anycast scenario with several kamailios sharing an active state and I've noticed that the dmq_usrloc module is not replicating the usrloc socket_info attribute when a register comes in through one of the kamailio nodes. Is there a reason behind this? I'm thinking on adding the replication for that attribute in the code. Another way of address this woud be by having a modparam to replicate the socket. What do you think? This is what I'm seeing: anycast node that actually gets the SIP REGISTER: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 585 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: udp: 1.1.1.1:5060 [http://1.1.1.1:5060/] Methods: -1 Ruid: uloc-5de8056d-22ab-f Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } anycast node that gets the USRLOC update via KDMQ: $ kamcmd ul.dump { Domains: { Domain: { Domain: location Size: 32768 AoRs: { Info: { AoR: 309(a)nmreis.test HashID: -35263718 Contacts: { Contact: { Address: sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test Expires: 574 Q: -1.000000 Call-ID: 6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0 CSeq: 4 User-Agent: Jitsi2.11.5602Linux Received: sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] Path: [not set] State: CS_SYNC Flags: 1 CFlags: 192 Socket: [not set] Methods: -1 Ruid: uloc-5de7ff7b-4617-3 Instance: [not set] Reg-Id: 0 Server-Id: 0 Tcpconn-Id: -1 Keepalive: 1 Last-Keepalive: 1575487540 Last-Modified: 1575487540 } } } } Stats: { Records: 1 Max-Slots: 1 } } } } the actual KDMQ message: KDMQ sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] SIP/2.0 Via: SIP/2.0/UDP 10.100.100.22;branch= z9hG4bK6034.290e4186000000000000000000000000.0 To: < sip:usrloc@10.100.100.21:5060 [http://sip:usrloc@10.100.100.21:5060/] > From: < sip:usrloc@10.100.100.22:5060 [http://sip:usrloc@10.100.100.22:5060/] >;tag= 3393f0703fb0ccaca74109ff37de39f5-1e4c092b CSeq: 10 KDMQ Call-ID: 0cfde4d32a31addc-8875(a)127.0.0.1 [mailto:0cfde4d32a31addc-8875@127.0.0.1] Content-Length: 433 Max-Forwards: 1 Content-Type: application/json {"action":1,"aor":"309(a)nmreis.test","ruid":"uloc-5de8056d-22ab-f","c":"sip:309@192.168.150.158:55060;transport=udp;registering_acc=nmreis_test","received":"sip: 2.2.2.2:31171 [http://2.2.2.2:31171/] ","path":"","callid":"6e7635ce4604e16400ee763b2ead55b2@0:0:0:0:0:0:0:0","user_agent":"Jitsi2.11.5602Linux","instance":"","expires":1575487600,"cseq":2,"flags":1,"cflags":192,"q":-1,"last_modified":1575487000,"methods":4294967295,"reg_id":0,"server_id":0} Looking forward to hear from you. BR, -- Nuno Miguel Reis Departamento de Engenharia Informática Faculdade de Ciências e Tecnologia Universidade de Coimbra

1 0

Vendor Specific RBT Call Flow
by Hamid Hashmi 04 Dec '19

04 Dec '19

Dear Users Can the following call flow be implemented, if yes, can you give some suggestions with modules information? [cid:d15f60e7-73ec-4cd8-8954-92f472a94ef7] Regards Hamid R. Hashmi

3 2

siremis help and usage index?
by PICCORO McKAY Lenz 04 Dec '19

04 Dec '19

it's there some help manual usage for siremis? the only KB source i have are asipto.. and only are install manuals!! Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

Re: [SR-Users] kamailio IMS supporting VOLTE
by Shubhendu Das 04 Dec '19

04 Dec '19

Hi , Does Kamailio has its own HSS supporting CX Interface ? BR, Shubhendu From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Mojtaba Sent: Saturday, November 23, 2019 4:37 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] kamailio IMS supporting VOLTE Hello, Running the IMS network for supporting VoLTE with Kamailio module and FHOSS is challenging. THe Kamailio works great in IMS network but You have to optimise your own database. Using FHOSS without any support or development is very risky. In general speaking, running VoLTE have three main section and the IMS network is one of them. You might have some challenges in EPC and RAN access. WIth Best Regards. Mojtaba On Thu, Nov 21, 2019 at 3:23 PM Shubhendu Das <shubhendu.das(a)azcom.in> wrote: Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- --Mojtaba Esfandiari.S -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 1

Fwd: app-ads.txt monetization proposal
by Joel Serrano 04 Dec '19

04 Dec '19

Not sure if this is BS or worth exploring, it's not my expertise area so I'm forwarding to you all just in case ---------- Forwarded message --------- From: Lorenzo from Instal <lorenzo.tacchio(a)instal.com> Date: Tue, Dec 3, 2019 at 8:56 AM Subject: app-ads.txt monetization proposal To: <joel(a)textplus.com> app-ads.txt monetization proposal Hello , I am Lorenzo, Senior Account Manager at Instal; we are currently buying your app's traffic via DSPs. We thought would be beneficial to work directly with you, *monetizing your inventory with no intermediate levels at all*. The key of our success is the *simple technical integration*. No SDK is required, we will add our ads.txt to your page and your app would be included in our media buying plan with budgets of *more than $1M per month*. Would you be interested to discover more in detail this opportunity? Please share your Skype ID or add me, *live:loresseintes*, to connect further! Kind Regards, Lorenzo Instal Srl, Via Ippolito Pindemonte, Florence, Italy Manage preferences <https://hs-3923849.s.hubspotemail.net/hs/manage-preferences/unsubscribe?d=V…>

1 1

secfilter usage
by Aymeric Moizard 03 Dec '19

03 Dec '19

Hi all, I'm testing and moving my kamailio script to use the newer secfilter module. In the past, I was replying "200 Ok" to "friendly scanners"... With the newer secfilter module, it looks like I can't send a reply with "sl_send_reply("200", "OK")" secf_check_ua(); if ($? == -2) { sl_send_reply("200", "OK"); exit; } I have read the code of the secfilter, but I was not able to see any specific code to silently discard the SIP request. I can see in the documentation about "secf_check_sqli_all", that the SIP message is supposed to be "dropped". I can see "w_check_sqli_all" returns 0 on detection and w_check_ua returns -2 upon detection. Are the message discarded because a negative value was returned? Would it be doable, using the secfilter, to still reply 200 Ok? Regards Aymeric -- Antisip - http://www.antisip.com

2 5

kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

Hi , I have a query regarding Kamailio open source SIP server. Does it have any release, where its supports the feature of IMS supporting VOLTE with real commercial mobile phones(IMSI based) ? If so, what is the release number ? Please also let me know about the supported mobile phones list. BR, Shubhendu -- _Disclaimer: This email and any files transmitted along with it may contain Azcom confidential and proprietary information. If you are not the intended recipient, you are notified that disclosing, copying, distributing or taking any action based on the contents of the information contained herein is strictly prohibited. If you are not an intended recipient of this transmission and you received it in error, please inform the sender by reply e-mail and destroy this and all other copies of this transmission to which you have access. The sender of this email or Azcom does not accept liability for any errors or omissions in the contents of this message that may occur as a result._

2 2

Contact header requirement
by David Villasmil 02 Dec '19

02 Dec '19

Hello guys, I'm connecting to a provider who requires the contact contains the proxy's IP... I know this is weird and totally off-rfc... I setup the topoh module and used modparam("topoh", "mask_ip", "[PROXY_IP]:5060") but this makes the contact: Contact: <sip:[PROXY_IP]:5060;line=sr-Z213nshDqt98-2qiIoT8GFf. y8PWype.SRy0nRo3nCT*> I know this is how it works, but they don't like the additional ";line=...." Has anyone faced this kind of thing? is there any other way to set this to something like: Contact: <sip:[PROXY_IP]:5060> that won't mess everything? Thanks all, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 7

$null + 0 is not int
by Juha Heinanen 02 Dec '19

02 Dec '19

Why is it that $null + 0 is 0 but not int? What is it? -- Juha $avp(test) = $null + 0; if (is_int("$avp(test)")) INFO(test $avp(test) IS int); else INFO(test $avp(test) IS NOT int); Dec 1 02:46:01 salmon /usr/bin/sip-proxy[10138]: INFO: test 0 IS NOT int

2 5

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users