- sr-users - kamailio.org

xtrange error but those rules that already work, some bug?
by PICCORO McKAY Lenz 23 Sep '19

23 Sep '19

i have many xtrange errors in the log.. but the code seems as i guess completed each rule without error. In the kamailio i have that error by example: Sep 23 13:38:29 ser-server-02 kamailio[2847]: 4(2855) ERROR: *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio-devel.cfg] l=692 a=25 n=save that line represents a supposed error in the following code, BUT that code seem work.. kamailio it’s working perfectly.. so i’m worry about it, what it happened? # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) sl_reply_error(); #!ifdef WITH_ASTERISK route(REGFWD); #!endif exit; } why that happened? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

kamailio run in docker not working
by Gaurav Bmotra 23 Sep '19

23 Sep '19

hi i m trying to run kamailio in docker this kamailio/kamailio-ci:master-alpine but inside the container kamailio not running if i rung this / # /etc/init.d/kamailio status sh: /etc/init.d/kamailio: not found plz help me thanks -- *Regards:* Gaurav Kumar

2 3

rate-limiting enabled on kamailio.org
by Henning Westerholt 22 Sep '19

22 Sep '19

Hello, just to inform you that rate-limiting on kamailio.org web server has been enabled. The motivation is to throttle robots and other automated services to not overload the server. If you notice any new errors in accessing the service (like page not found, or a 429 Too Many Requests) let us know. Cheers, Henning -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

1 0

TLS in-dialog set_forward_no_connect()and upstream TLS LCR gateway
by Anthony Joseph Messina 21 Sep '19

21 Sep '19

In preparation for the 5.3 release, I've been testing the following configuration change for TCP/TLS connections: https://github.com/kamailio/kamailio/commit/ 8bba208fe6ae7ccb4c92362b8c33f1530b9f56da route[REQINIT] { # no connect for sending replies set_reply_no_connect(); if(has_totag()) { # no connect for requests within dialog set_forward_no_connect(); } This change creates issues when a UAC TLS INVITE routes to an upstream gateway using TLS to port 5061 (via the LCR module). Kamailio sends the initial outbound TLS connection from a local ephemeral port. The TCPOPS tcp_keepalive_enable function issues keepalives from the local ephemeral port to the gateway port 5061: https://kamailio.org/docs/modules/stable/modules/ tcpops#tcpops.f.tcp_keepalive_enable Even so, the TLS connection eventually times out, after which in-dialog requests from the UAC are no longer able to reach the upstream gateway. ERROR: tm [../../core/forward.h:293]: msg_send_buffer(): tcp_send failed WARNING: tm [t_fwd.c:1570]: t_send_branch(): sending request on branch 0 failed ERROR: sl [sl_funcs.c:372]: sl_reply_error(): stateless error reply used: Unfortunately error on sending to next hop occurred (477/SL) I figure I must be doing something wrong with my TCPOPS here. Is a TLS connection to an upstream gateway supposed to be maintained throughout the duration of a call? -- Anthony - https://messinet.com F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 6

Info: initialization of libssl 1.1.+ with PTHREAD_PROCESS_SHARED attribute for pthread mutex and rwlock
by Daniel-Constantin Mierla 21 Sep '19

21 Sep '19

Hello, after couple of months waiting for feedback about the workaround with the preloaded shared library to set the PTHREAD_PROCESS_SHARED attribute for pthread mutex/rwlock in order to avoid deadlocks and as it seems to be the solution, I just pushed a few commits to move that code in the core of kamailio. It this way the initialization with the PTHREAD_PROCESS_SHARED attribute is done by default when libssl v1.1 is detected at compile time, without the need of using the approach with a preloaded library. The shared library is still kept in the tls module: * https://github.com/kamailio/kamailio/tree/master/src/modules/tls/utils/open… Probably is going to be removed in the near future. I am asking the community members that use Kamailio on systems with libssl v1.1+ to test the master branch and give feedback. The master branch is anyhow in testing phase for preparing the next major release v5.3.0, so it will help for that as well. If no issue is going to be reported soon, the commits will be backported to stable branches, to re-enable smooth use of Kamailio with libssl v1.1+. When libssl v1.1+ is detected, the core links with pthread, but that was already present in the Makefiles.defs for different compile options. The pthread more or less a default library on linuxes/unixes, so this should not introduce any major dependency to the Kamailio core. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

6 21

Possible locking issue in usrloc module in file udomain.c
by Rupesh Kumar 18 Sep '19

18 Sep '19

Hi , Currently I am running kamailio 4.2 and now upgrading to kamailio 5.2 and was just going thru the changes. In the file modules/usrloc/udomain.c inside the function free_udomain( ) , it appears that we are not protecting via lock_ulslot and unlock_ulslot doing free. Could it cause some race condition ? May be its intentional but going thru the github history its there since day 1. I checked all branches in kamailio 4.x and we have the protection. In 4.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { lock_ulslot(_d, i); deinit_slot(_d->table + i); unlock_ulslot(_d, i); } shm_free(_d->table); } shm_free(_d); } In 5.x void free_udomain(udomain_t* _d) { int i; if (_d->table) { for(i = 0; i < _d->size; i++) { deinit_slot(_d->table + i); } shm_free(_d->table); } shm_free(_d); }

2 2

How to distinguish is from Asterisk or from the client
by przeqpiciel 18 Sep '19

18 Sep '19

Hi, I looking information how to distinguish whether the packet is directed to the Astersik (my platform) or to the customer (softphone / sip client ).

5 5

NAT problem
by Arnout Van Den Kieboom 18 Sep '19

18 Sep '19

Hi, First of all, I'm really new to Kamailio. So sorry if I ask a stupid question, or perhaps a really weird one. I started using kamailio in a test environment about a week ago. for setup i have the situation for inbound calls: Carrier --> FW (NAT) --> Kamailio for outbound to an asterisk box i have: Kamailio --> FW (NAT) --> Asterisk for outbound to users i have: Kamailio --> FW (NAT) --> sip-phone (yealink) or grandstream During that time i was able to : Set up calls to users, Use the dispatcher module use the avpops module to do data lookups. Get calls from asterisk to kamailio user to work. Also calls between users (even though behind nat) are having good audio. I have installed rtpproxy and it works nicely. There is just one situation where it fails: Carrier --> FW (NAT) Kamailio --> asterisk There is only one way audio (from asterisk to carrier) but not from carrier to asterisk. I believe I need to do something with the rtp proxy module ... and tried different things (forcing the r flag, forcing the w flag) Any idea where I might need to start looking? It's been driving me crazy... Thanks Pemp

4 7

app_lua: invalid Lua environment attributes or parameters
by Yuriy Gorlichenko 18 Sep '19

18 Sep '19

Hi! Im using kamailio 5.1.8 with app_lua and getting this message during startup of kamailio. Looks like I forgot to add something at the module setup, but cant figure out what is wrong. My setup of app_lua is: modparam("app_lua", "load", "/etc/kamailio/extended/main.lua") modparam("app_lua","reload",1) modparam("app_lua","register","textops") app_lua [app_lua_sr.c:1599]: sr_kemi_lua_exec_func_ex(): invalid Lua environment attributes or parameters starting kamailio service with kamailio -f /etc/kamailio/extended/main.lua Also here is little bit confusing me that in case of use "KSR" everywhere I'm getting message about "sr"... Thx in advice

2 5

dedicated bearer - Kamailio settings
by Jack R 18 Sep '19

18 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu. I have done the attached (image) settings to create dedicated bearer in nextepc as advised by this link: https://github.com/open5gs/nextepc/issues/180 Do I need to do some setting in Kamailio server side (maybe in pcscf cfg files etc.to take some information from PCRF) to make dedicated bearer work end to end or that is all (just configure the new apn in the phone) rest will be taken care itself? Thanks and Regards, Jack.

2 1

core dump if silo table > ~5000 records
by David Villasmil 18 Sep '19

18 Sep '19

Hello guys, i've got a kamailo: version: kamailio 5.0.0 (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. id: unknown compiled on 19:50:58 Mar 28 2017 with gcc 4.9.2 which was core dumping every 20 seconds and we had no idea why. We emptied out the silo table, which had: mysql> select count(*) from silobackup; +----------+ | count(*) | +----------+ | 5240 | +----------+ records. Once we truncated it, kamailio started working again. Is there a limit to how many records msilo can handle? I will look into the data in the table, but in case there's a limit, i'd like to know. core available on request. Thanks Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 3

Kamailio+B2BUA on IPv4-only and NAT64 Apple app store tests
by Anthony Alba 17 Sep '19

17 Sep '19

Hi folks, I'm wondering what you do to pass Apple app store tests (IPv6-only NAT64/DNS64 scenario) with the iOS app likely to put IPv6 addresses in the SDP, when Kamailio and your B2BUA(Asterisk, FreeSWITCH,...) are IPv4-only endpoints. By "pass" I mean media flowing + SIP working - not sure the app store gatekeepers would actually test media, but it would be nice to have a robust solution. Strictly speaking, this is not a Kamailio's problem but a media SDP/<your-rtp-proxy-of-choice> problem. But since everything is so closely intertwined I'm wondering where you choose to the location the "solution". I have noticed that if Kamailio is IPv4-only and it "sees" IPv6-addresses(Contact, Via headers??) it will try to resolve the IPv6-address as a DNS name! The Wikipedia article on NAT64 mentions 464XLAT(RFC 6877) and Port Control Protocol (RFC 7225), but those seem to be deep rabbit-holes... Cheers Anthony Alba

4 5

Kamailio merchandising shop
by Henning Westerholt 16 Sep '19

16 Sep '19

Hello, there is now a Kamailio merchandising shop available: https://skalatan.de/merchandising/ You find there many options e.g. different Kamailio T-Shirts, Hoodies, Bags, Caps, Mugs, etc.. It provides shipping in many countries world-wide (based from Europe and also USA). Feedback via (direct) e-mail welcome. Best regards, Henning

1 1

3GPP compliance
by Tihomir Culjaga 15 Sep '19

15 Sep '19

Hi, is there any 3GPP compliance list for Kamailio IMS ? Regards, Tihomir.

2 1

RUN kamailio in kubernetes
by Gaurav Bmotra 14 Sep '19

14 Sep '19

hi i want to run kamailio in kubernetes deployment and i want to run this kamailio/kamailio-ci:5.2 image in to the kubernetse .. can anyone help thank you -- *Regards:* Gaurav Kumar

5 6

Problem with incoming audio from Asterisk
by przeqpiciel 14 Sep '19

14 Sep '19

Hi, After a month of fight with SIP signalling I made it but now I have problem with incoming audio from Asterisk. Kamailio and Asterisk are their own public IPs I dont use rtpproxy. When Asterisk sends rtp I'm able to see RTP on my router but it not going to my computer. Maybe somebody have idea why ?

2 1

Using $shv within current call only
by Zhan Bazarov 13 Sep '19

13 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } it nicely work for outgoing request, but doesn't for reply.... reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var/$avp... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this, how we can assign $shv within current call only? Many thanks!

3 2

Issue with <null> and 0 when comparing vars
by Joel Serrano 13 Sep '19

13 Sep '19

Hello, I'm running into what I believe a weird situation when comparing <null> to 0. I have the following lines: xlog("L_NOTICE", "DEBUG: sel(contact.expires)=$sel(contact.expires) - hdr(Expires)=$hdr(Expires)\n"); if ($hdr(Expires)==0 || $sel(contact.expires)==0) { xlog("L_NOTICE", "[end] removed user location\n"); } else { xlog("L_NOTICE", "[end] saved user location\n"); } I registered with 3 devices, one sends expires as header, another as contact param, and the last uses both header AND contact param. My surprise was this: 1- Both header and contact param: Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=600 Sep 12 19:37:48 csbc01 csbc[8036]: NOTICE: {1 4 REGISTER 02c3ea6ee623916d37532188031017c9@0:0:0:0:0:0:0:0} <script>: [end] saved user location 2- Only contact param: Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: DEBUG: sel(contact.expires)=600 - hdr(Expires)=<null> Sep 12 19:36:15 csbc01 csbc[8035]: NOTICE: {1 40629 REGISTER f2cbaa53-eabfd633(a)84.121.150.224} <script>: [end] removed user location 2- Only header: Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: DEBUG: sel(contact.expires)=<null> - hdr(Expires)=3600 Sep 12 19:29:33 csbc01 csbc[8033]: NOTICE: {1 402 REGISTER 2_1804628355(a)192.168.30.31} <script>: [end] removed user location I can workaround the header by using is_present_hf() before comparing, I can also probably parse the $ct to see if it has some Expires in it, but I really want to know if my understanding is incorrect and I'm doing something wrong or if this is a bug? Thanks! Joel.

2 2

Testing server up with SIP OPTIONS/netcat
by Andrew White 13 Sep '19

13 Sep '19

Hi all, As part of a custom monitoring solution, I’m writing a simple script that sends an OPTIONS via netcat to a Kamailio server. However the server doesn’t appear to like it. Here’s the text - options.sip: OPTIONS sip:10.0.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.20:5060 From: <sip:monitoring@10.0.0.20> To: sip:monitoring@10.0.0.10 Call-ID: monitoring(a)10.0.0.20 CSeq: 1 OPTIONS I’m running the following command from the 10.0.0.20 machine: nc -u 10.0.0.10 5060 < options.sip And I get the following error in the Kamailio logs: ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] WARNING: <core> [core/receive.c:319]: receive_msg(): parsing relevant headers failed ERROR: <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: pv [pv_core.c:1893]: pv_get_hdr(): error parsing headers ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} maxfwd [mf_funcs.c:51]: is_maxfwd_present(): parsing MAX_FORWARD header failed! ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:56]: parse_cseq(): method terminated unexpectedly ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [CSeq: 1 OPTIONS] ERROR: {1 <null> monitoring(a)13.236.91.139} <core> [core/msg_translator.c:2361]: build_res_buf_from_sip_req(): alas, parse_headers failed As far as I can see, the message itself is valid. It appears parse_cseq() is expecting a higher cseq, or more headers? I’ve looked at another OPTIONS message we get regularly from one of our upstream, and it’s almost identical to this one. Can anyone point me the right way here? Thanks! Andrew

3 2

Presence and BLF
by Bill Neely 12 Sep '19

12 Sep '19

Hi all: I am trying to set up presence and BLF notifies on Kamailo 5.2.4 I am following the instructions laid out in http://kb.asipto.com/kamailio:presence:k43-blf Don't know if these instructions are valid, but I am definitely not getting any light action (using Grandstream 2130 and Grandstream Wav for testing.) Entries in active_watchers appear to be correct. Here is my cfg file: #!KAMAILIO # # Kamailio (OpenSER) SIP Server v5.1 - 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" #!define WITH_DEBUG #!define WITH_NAT #!define WITH_AUTH #!define WITH_MYSQL #!define WITH_USRLOCDB #!define WITH_PRESENCE ####### 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 FLT_DLG 9 #!define FLT_DLGINFO 10 #!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=3 log_stderror=no #!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 */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:62.151.xx.xx:50060 # allow http requests listen=tcp:127.0.0.1:8080 tcp_accept_no_cl=yes #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" 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 = "104.192.xx.xx" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib64/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" loadmodule "presence_dialoginfo.so" loadmodule "pua.so" loadmodule "pua_dialoginfo.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 # load the module loadmodule "xhttp.so" loadmodule "json.so" loadmodule "pua_json.so" loadmodule "dialog.so" # ----------------- 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) modparam("presence", "server_address", "sip:62.151.xx.xx:50060" ) modparam("presence", "send_fast_notify", 0) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 2) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) # ----- presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_single_dialog", 0) # ----- pua params ----- 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) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) #!endif # ----- dialog params ----- modparam("dialog", "db_url", DBURL) modparam("dialog", "enable_stats", 1) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_flag", FLT_DLG) #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:22222") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/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 { xlog("L_ALERT","ALERT: REQUEST ROUTE ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^6 $rm from $fu (IP:$si:$sp) $du\nMESSAGE BUFFER:\n$mb\n\n\nRU:$ru\n"); $var(i)=0; while($var(i)<$branch(count)) { xlog("$(branch(uri)[$var(i)])\n"); $var(i) = $var(i) + 1; } # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # handle registrations route(REGISTRAR); # 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"); xlog("L_ALERT","ALERT: line 492******************************** $rm from $fu (IP:$si:$sp)\n"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_ALERT","ALERT: line 494******************************** $rm from $fu (IP:$si:$sp)\n"); record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } event_route[xhttp:request] { if ($hu != "/presence/") { xlog("L_INFO", "invalid request [$hu]"); xhttp_reply("404", "Not Found", "text/html", ""); return; } $var(call-id) = $(rb{json.parse,Call-ID}); xlog("L_INFO", "$var(call-id)|log|[$hu] payload $rb"); if ($(rb{json.parse,Event-Package}) == "dialog") { xlog("L_INFO", "$var(call-id)|log|received $(rb{json.parse,Event-Package}) update for $(rb{json.parse,From}) $rb"); pua_json_publish($rb); } xhttp_reply("200", "OK", "text/html", "$rb"); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if(is_method("INVITE|BYE|UPDATE|CANCEL|ACK")) { setflag(FLT_DLGINFO); dlg_manage(); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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("1511", "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] { xlog("ALERT:NATMANAGE********************* (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; #!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("corwie"); } 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 # Routing to foreign domains route[SIPOUT] { xlog("ALERT: SIPOUT (IP:$si:$sp)\n(DEST:$du @ $di:$dp)\n"); ; if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } route[TOAST] { rewritehostport("45.33.xx.xx:5060"); $fu = "sip:7028743833@62.151.182.83:5012"; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($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; } 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 }

2 2

Kamailio IMS setup - user online - unable to call
by Jack R 12 Sep '19

12 Sep '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Users alice and bob are online however unable to call. When we check the logs the FHoSS is hunting for values in sh_subscription table for alice / bob user. Can you please share: 1. A sample database table values to be inserted across all tables in FHoSS helpful for the call and 2, Maybe, if possible, some description to understand table structures/ importance of fields of FHoSS. Thanks and Regards, Jack.

2 1

Help with "routines:ssl3_read_bytes:sslv3 alert bad certificate" kamailio TLS error
by david＠aslo.us 12 Sep '19

12 Sep '19

Hello everyone, I am trying to configure TLS in kamailio (5.2.4) following this guide: [ http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ]( http://www.kamailio.org/dokuwiki/doku.php/tls:create-certificates ) Modules: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_PRESENCE #!define WITH_ALIASDB #!define WITH_IMC #!define WITH_TLS When i try to connect via command line, this is the result (just including relevant parts): $ openssl s_client -connect 192.X.X.X:5061 -tls1 CONNECTED(00000003) depth=1 C = XX, ST = XXXX, L = XXXXXX, O = XXX CA, CN = XXX CA verify error:num=19:self signed certificate in certificate chain verify return:0 --- No client certificate CA names sent --- SSL handshake has read 2550 bytes and written 336 bytes --- --- Start Time: 1567787935 Timeout : 7200 (sec) Verify return code: 19 (self signed certificate in certificate chain) --- read:errno=0 Now, when I setup my clients, they connect to the server, but they can't send messages or make calls. This is the TLS startup LOG: Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:372]: mod_init(): With ECDH-Support! Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_mod.c:375]: mod_init(): With Diffie Hellman Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: CRITICAL: tls [tls_init.c:671]: init_tls_h(): installed openssl library version is too different from the library the kamailio tls module was compiled with: installed "OpenSSL 1.1.1 11 Sep 2018" (0x1010100f), compiled "OpenSSL 1.1.0k 28 May 2019" (0x101000bf).#012 Please make sure a compatible version is used (tls_force_run in kamailio.cfg will override this check) Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:680]: init_tls_h(): tls_force_run turned on, ignoring openssl version mismatch Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 12582912 and 6291456 bytes Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold1 has been changed to 12582912 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 6291456 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [main.c:2669]: main(): processes (at least): 24 - shm size: 67108864 - pkg size: 8388608 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSs<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSs<default>: certificate='/etc/certs/192.X.X.X/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSs<default>: ca_list='/etc/certs/demoCA/cert.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSs<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSs<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSs<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSs<default>: private_key='/etc/certs/192.X.X.X/key.pem' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSs<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSs<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: NOTICE: tls [tls_domain.c:1087]: ksr_tls_fix_domain(): registered server_name callback handler for socket [:0], server_name='<default>' ... Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:707]: set_verification(): TLSs<default>: No client certificate required and no checks performed Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:303]: ksr_tls_fill_missing(): TLSc<default>: tls_method=12 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:315]: ksr_tls_fill_missing(): TLSc<default>: certificate='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:322]: ksr_tls_fill_missing(): TLSc<default>: ca_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:329]: ksr_tls_fill_missing(): TLSc<default>: crl='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:333]: ksr_tls_fill_missing(): TLSc<default>: require_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:340]: ksr_tls_fill_missing(): TLSc<default>: cipher_list='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:347]: ksr_tls_fill_missing(): TLSc<default>: private_key='(null)' Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:351]: ksr_tls_fill_missing(): TLSc<default>: verify_certificate=0 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:354]: ksr_tls_fill_missing(): TLSc<default>: verify_depth=9 Sep 6 16:41:57 aslo-kamailio /usr/sbin/kamailio[5845]: INFO: tls [tls_domain.c:710]: set_verification(): TLSc<default>: Server MAY present invalid certificate Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5862]: INFO: jsonrpcs [jsonrpcs_sock.c:443]: jsonrpc_dgram_process(): a new child 0/5862 Sep 6 16:41:58 aslo-kamailio /usr/sbin/kamailio[5866]: INFO: ctl [io_listener.c:214]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) This is my tls.cfg file: [server:default] method = TLSv1 verify_certificate = no require_certificate = no private_key = /etc/certs/192.X.X.X/key.pem certificate = /etc/certs/192.X.X.X/cert.pem ca_list = /etc/certs/demoCA/cert.pem #crl = /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 verify_certificate = no require_certificate = no These are the relevant parts of my kamailio.cfg: # alias="sip.mydomain.com" alias=192.X.X.X:5060 alias=192.X.X.X:5061 /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ listen=udp:192.X.X.X:5060 listen=tcp:192.X.X.X:5060 listen=tls:192.X.X.X:5061 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") modparam("tls", "tls_force_run", 1) #!endif These are the errors that show up everytime i try to connect with a client: Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:42 aslo-kamailio /usr/sbin/kamailio[5870]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:43 aslo-kamailio /usr/sbin/kamailio[5874]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14094412:SSL routines:ssl3_read_bytes:sslv3 alert bad certificate Sep 6 16:53:44 aslo-kamailio /usr/sbin/kamailio[5875]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f7c4e3ddd00 r: 0x7f7c4e3ddd80 (-1) Any help would be greatly appreciated. Regards.

2 1

Kamailio status get Active (Exited)
by Gaurav Bmotra 11 Sep '19

11 Sep '19

hi i m using kamailio 5.1, with -m 512 -M 8 it stop registering user SIP anf in status it shows ------------------------------ ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2019-09-07 11:05:16 UTC; 3 days ago Docs: man:systemd-sysv-generator(8) Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17708]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17677]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17714]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17685]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17681]: {1 1094155754 NOTIFY TwGwmzx4wINnraJs1Gh3Kg..} INFO: <core> [main.c:819]: sig_usr(): s… received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17704]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17689]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17693]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17691]: INFO: <core> [main.c:819]: sig_usr(): signal 15 received Sep 11 06:38:36 talk /usr/local/sbin/kamailio[17674]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Hint: Some lines were ellipsized, use -l to show in full. --------------------------------------------------------------------- plz help thank you -- *Regards:* Gaurav Kumar

2 1

Can't connect to db
by Andrew Chen 10 Sep '19

10 Sep '19

Hi all, I just installed kamailio 5.2.4 on two separate aws instances in two separate regions. Both regions uses the same configuration template and share the same db in AWS RDS. The issue I see is this: - SJO location was able to launch fine without any issues. - ASH location could not start up kamailio process due to db connection failure: Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8043]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Too many connections Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: carrierroute [db_carrierroute.c:141]: carrierroute_db_open(): can't connect to database. Sep 10 21:24:13 ashmainkama51 /usr/sbin/kamailio[8045]: ERROR: <core> [core/sr_module.c:847]: init_mod_child(): error while initializing module carrierroute (/usr/lib/x86_64-linux-gnu/kamailio/modules/carrierroute.so) (idx: 30 rank: 18 desc: [tcp receiver (generic) child=9]) - In ASH instance, I can manually mysql to the RDS host using the credentials in the config file. - I checked db user permissions and it's opened for the subnet in our ASH location (which is also proven using manual connection). So I don't get why one works and the other doesn't Thoughts? -- Andy Chen Sr. Telephony Lead Engineer -- *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

LCR lcr_id short or int?
by Grant Bagdasarian 10 Sep '19

10 Sep '19

Hello, The database tables definition for 5.2.x mentions the lcr_id parameter being of unsigned short type, while in the code I only see definitions of type int. Is it safe to assume the parameter is of type int, so the database column can also be set to an integer type? Regards, Grant

2 7

Share variable into reply route.
by Zhan Bazarov 09 Sep '19

09 Sep '19

Hello, we ran into a trouble with sharing variable into reply route... We are going to match asterisk's uniqueid with rtpengine records like this: route[from_asterisk]{ $var(UniqueId)=$hdr(X-UniqueId); $var(rtp_flags) = $var(rtp_flags) + " label={uid:$var(UniqueId),stream:outgoing} record-call=on"; rtpengine_manage($var(rtp_flags)); } reply_route[from_pstn]{ rtpengine_manage("replace-origin replace-session-connection ICE=remove RTP/AVP rtcp-mux-demux label={uid:$var(UniqueId),stream:outgoing"); } and it doesn't work... The one possible way what we found its using $shv instead of $var... But its unsecure way, cuz it may confus our bussines logic... Do you have any idea how to implement this? Many thanks!

3 2

How to install kamailio symbols for debugging
by YASIN CANER 09 Sep '19

09 Sep '19

Hello, I want to debug kamailio with gdb or try to get core.dump file but i failed. GDB wants to kamailio symbols. How to install them? on Ubuntu - raspberry PI Best regards. Yasin CANER

4 3

Error Users - 604 Does not exist anywhere - HSS User Unknown
by Jack R 08 Sep '19

08 Sep '19

Hi, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. 1. 2 Sip client registers successfully (shows Online) as 'bob(a)net1.test' and 'alice(a)net1.test' 2. When we check server logs: Invite request shows 'bob@IP address' i.e. ' bob(a)10.0.0.10' 3. However, in HSS it is registered as 'bob(a)net1.test'. 4. When the request from SCSCF goes to ISCSF it gives Error: Users - 604 Does not exist anywhere - HSS User Unknown. As it compares 'bob(a)10.0.0.10' with 'bob(a)net1.test'. Please help and advise how everywhere (SCSCF, PCSCF) during query/comparisons it can take 'bob(a)net1.test' and not 'bob(a)10.0.0.10' Thanks and Regards, Jack.

1 1

Kamailio - PfSense Firewall - inbound/outbound rules
by Youssef Boujraf 06 Sep '19

06 Sep '19

Dear All, I am evaluating the kamailio proxy 5.2 server with rtpproxy on an internal network. Everything are ok. Now, I would like to evalute Kamailio proxy from outside (internet) but I have some difficulties to setup my PfSense firewall to enable inbound & outbound rules, network settings and kamailio.cfg. The issues are : Setup correctly Network files : * /etc/hostname 127.0.0.1 localhost ip_private codecsip.mydomain codecsip ip_public codecsip.mydomain codecsip * /etc/hosts file codesip * dns * /etc/resolv.conf nameserver 8.8.8.8Setup /etc/kamailio/kamailio.cfg :#!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_NAT #!define WITH_TLS /* 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="codecsip.mydomain" #!ifdef WITH_NAT # ----- rtpproxy params ----- # modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") modparam("rtpproxy", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock") Setup etc/default/rtpproxy : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS="-l <IP-address>" Where <IP-address> is the external IP address of your host. Regarding the firewall rules, it will be great to identify which rules I need to create : Inbound and outbound rules and NAT 1:1 ? If anyone have any information it could be great Best regards, Youssef

2 1

Kamailio HA
by Christian Attanasio 06 Sep '19

06 Sep '19

Hi everyone I use kamailio 5.1 with rtpengine and I would like to set up a simple warm standby or hot standby system for my sip server. on the web I found various discussions and posts that led me to deepen them the dispatcher, dmq, dmq_usrloc, db_cluster and p_usrloc modules. now, though, I'm a little confused: what approach do you recommend for this system? I thought of using two identical servers: Kam01 10.10.10.2 Kam02 10.10.10.3 Hostname cluster.mydomain.com each with its own mariadb and implementing data replication (with the dmq module? Or db_cluster and sqlops?). in this scenario is it necessary to also use the dispatcher module or implement a high available system with keepalives and floating ip? thanks, Christian

1 0

Kamalio & Asterisk integration erroneous behavior, setup issue
by Aristeidis Tsitras 05 Sep '19

05 Sep '19

new to the area and trying to setup Kamailio with Asterisk in a single machine. All users will register to Kamailio's port and in case of need for media, it will be forwarded to Asterisk, that is my intention. All of my work is based on the following link https://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb. Here is what i have done: - Debian 8, 64 bit machine with mysql and odbc - *root@kamast: ~ $ lsb_release -a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 8.11 (jessie) Release: 8.11 Codename: jessie root@kamast: ~ $ uname -a Linux kamast 3.16.0-10-amd64 #1 SMP Debian 3.16.72-1 (2019-08-13) x86_64 GNU/Linux root@kamast: ~ $ * - Kamailio 5.2 installed from Kamailio's deb repository - Asterisk 13LTS installed from source - Used the same passwords such as kamailiorw and asterisk_password, since this is a test system, for proof of concept. I did import to the mysql>asterisk database 3 users 2200, 2201 and 2202. Then created in sip.conf the same 3 users with the same credentials. Then on 3 PCs i used softphones (Jitsi, Zoiper) and registered each account to a softphone. Problems: - Cannot see the users in the Asterisk's cli, sip show peers - I can see users only in Kamailio with kamctl ul show - A call between the extensions goes to voicemail. It never reaches the other destination eg 2200 calls 2201 and in Asterisk's console i am getting a message that 2201 is absent and it goes to voicemail. The same with any other extension. Attached you can find: 1. Kamailio.cfg 2. Asterisk's sip.conf 3. Asterisk's extension.conf 4. The import that i have done to mysql for the user creation. I would appreciate if someone could point me to the error and help me fix it please?

3 7

$Au being set to $fu instead of $fU
by Joel Serrano 05 Sep '19

05 Sep '19

Hi, Testing on latest master deb nightly build from last night I've noticed that when $au is not available, $Au is being set to $fu (uri) instead of $fU (username). Docs: https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables ... $Au - Acc username: username for accounting purposes. It's a selective pseudo variable (inherited from acc module). It returns auth username ($au) if exists or From username ($fU) otherwise. ... I have the following log in several places in a test box: xlog("L_NOTICE", "[DEBUG] Au=$Au au=$au aU=$aU fU=$fU fu=$fu - \n"); And when I run through them (on both request and reply routes) I get the following: Aug 8 14:37:43 csbc01 csbc[9478]: NOTICE: {1 21564 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9479]: NOTICE: {1 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=1002366 aU=1002366 fU=1002366 fu=sip:1002366@some.domain - Aug 8 14:37:43 csbc01 csbc[9484]: NOTICE: {2 21565 REGISTER d8df9be1-1702e20b(a)A.B.C.D} <script>: [DEBUG] Au=1002366(a)some.domain au=<null> aU=<null> fU=1002366 fu=sip:1002366@some.domain - Those 3 logs belong to the same flow from different places in the config script. I would expect $Au to never have a "@some.domain" in it, so I assume in some place the replacement is not being done with $fU but with $fu. @Devs: I'm happy to create a GH issue for tracking etc if required, please let me know! Cheers, Joel.

2 3

How to detect miss call and accomplished call
by Anuran Barman 05 Sep '19

05 Sep '19

Hi, I want to show call logs in my SIP app. How can I do this with Kamailio? I want to do it from the *kamailio.cfg *file. Is it possible to do it from where? I guess Dialog module should be used for this, but can not figure out how to do it. I want minimalist feature like who call whom with timestamp, thats all. How can I do this?

3 9

Kamailio-IMS system
by Hạnh Nguyễn 05 Sep '19

05 Sep '19

Dear Mrs/Mss, I am Hanh, I am from Viet Nam. I am writting this email because I have an issue when I download The log-file and the config-files on website https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/. When I click on https://www.dropbox.com/s/ehkzi9pbhlp7njc/Kamailio-IMS%20config-files.zip?d… to download The log-file which I really need on this time, the page displays 404 Error - Link not found. So I can not download. I hope I will receive a new link from you soon. I look forward to hearing from you. Sincerely, Hanh Nguyen

2 1

Planning the next major release - v5.3.0
by Daniel-Constantin Mierla 04 Sep '19

04 Sep '19

Hello, it's about the time to set the milestones to the next major release - v5.3.0. There are 6 new modules and many other enhancements to existing components in git master branch, also during the last irc devel meeting it was proposed to freeze after summer holidays. At this moment I would propose Wednesday, September 4, 2019 to freeze the development. Then the release should be about one month later, after testing interval. In case there are people willing a bit more time for development, we can freeze one week later, September 11, or another date -- your suggestions here. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

3 5

Using Kamailio IMS for making a call
by supreeth herle 04 Sep '19

04 Sep '19

Hello Everyone, I have successfully setup a VM in OpenStack with Kamailio IMS, rtpengine and FoHSS using the instructions mentioned in the guide which i wrote (attached to this mail), all running in the machine. I am using the existing alice and bob IMPUs created in FoHSS. But, when I try to call each other the call doesnt go through and I observe in the wireshark traces that S-CSCF is replying to I-CSCF that "403, Forbidden - Domain not served". Can somebody please tell me do I need any additional components to make a call or am i missing some configuration? It would be great if someone can help me troubleshooting this issue. I have attached the pcap file for your reference. Many thanks in advance. Regards, Supreeth

1 0

Kamailio stops processing requests over UDP
by Juha Heinanen 04 Sep '19

04 Sep '19

I'm returning to an old issue, where Kamailio stops processing incoming requests. Now it turned out that only requests over UDP are not processed. Requests over TCP work fine. Debug log doesn't show anything about the requests over UDP, but wireshark sees them coming. core.ps shows that main process - attendant and all udp receiver processes are alive. Any ideas what could cause Kamailio (5.2 version) to stop processing requests over UDP? What additional info should I try to get? -- Juha

4 21

Hello Group - I have a "challenge"
by Kevin Greene 04 Sep '19

04 Sep '19

I recently did the upgrade from v4.4 to 5.2 (via Yum). When starting Kamailio, I get these three errors. Not sure if they are related or not but I do not think so. Have spent a lot of time searching on Google for the answer and have not found the solution. [root@ids kamailio]# kamctl restart INFO: Stopping Kamailio : INFO: stopped INFO: Starting Kamailio : INFO: started (pid: 44608) [root@ids kamailio]# tail -f /var/log/sipserver.log Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/pvapi.c:1119]: pv_parse_spec2(): wrong char [t/116] in [$mt] at [2 (0)] Sep 4 00:39:06 ids /usr/sbin/kamailio[44608]: ERROR: <core> [core/dprint.c:451]: log_prefix_init(): wrong format[{$mt $hdr(CSeq) $ci} ] Sep 4 00:39:06 ids /usr/sbin/kamailio[44626]: ERROR: <core> [core/receive.c:319]: receive_msg(): no request_route {...} and no other config routing engine registered The last error is repeated many times (each time a remote user tries to connect I think). I can send any additional files if needed. Thanks. I am burning out on this upgrade. Kevin

2 1

Unexpected BYEs from dialog module
by Alex Balashov 03 Sep '19

03 Sep '19

Hi, I am running 4.3.5:1b0c0a, which I am aware is an EOL'd release train, and have a problem with the dialog module I am baffled by. On many calls - I can't find any correlation to particular kinds of endpoints - I see spoofed BYEs come out of Kamailio after a minute and a half, as if the call had hit a dialog timeout or the dialog module's dead peer detection (OPTIONS pinging) kicked it out. The thing is, neither of those explanations seem to be borne out by the parameters of the calls under investigation. The dialog timeout on these calls is set to 7200, and the dialog keepalives aren't enabled in either direction. If they were, it'd be logged. Calls that are terminated in this manner also see the following log message: WARNING: dialog [dlg_req_within.c:214]: bye_reply_cb(): inconsitent dlg timer data on dlg 0x7f9997317e48 [3390:9644] with clid '1996679936_133218050(a)x.x.x.x' and tags '2bb17663-co4006-INS001' 'as7925780d' But I don't think this is unusual for a dialog-spoofed BYE; presumably this is due to a 200 OK for the BYE from both ends. So anyway, I am trying to track down anything else that could inadvertently cause dialog to hang up calls relatively quickly in that release, or inadvertently set the dialog timeout parameter to something lower than is apparent from the logging and the config. If anyone has any pointers, that would not go unappreciated! Many 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

crash/mem leaks on low memory
by Gaurav Bmotra 03 Sep '19

03 Sep '19

hi i m using kamailio 5.1 on Ubuntu 18.04 i m geting this error in logs and it stop registration form soft-phone plz help me -------------------------------------------------------------------------------------------------------- WARNING: tls [tls_init.c:778]: init_tls_h(): openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory th -------------------------------------------------------------------------------------------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Tue 2019-09-03 12:09:19 UTC; 26min ago Docs: man:systemd-sysv-generator(8) Process: 27237 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to 7077888 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 536870912 - pkg size: 8388608 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7f0b41b001f4, 16) on 172.31.35.49:5091 : Address already in use Sep 03 12:09:18 talk /usr/local/sbin/kamailio[27270]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized -- thank you *Regards:* Gaurav Kumar

3 3

evapi mismatch version
by bill＠novatrope.us 02 Sep '19

02 Sep '19

Hi All Installed Kam 5.2.4, but now getting version mismatch on evapi. ERROR: <core> [core/sr_module.c:348]: version_control(): module version mismatch for /usr/lib64/kamailio/modules/evapi.so; core: kamailio 5.2.4 (x86_64/linux); module: kamailio 5.1.6 (x86_64/linux) Anybody know how to get the correct evapi version. I am running centos 7 Bill

3 2

Issue with uac_replace
by David Costa 01 Sep '19

01 Sep '19

Greetings, I have a Kamailio proxy with uac_replace for To and From Headers. So far everything has worked correctly. However, I now have a client who encodes my Calling Number too and I'm getting some issues in the restore mechanics. I have the restore setup as "auto" and it's info is stored in "vsf" and "vst" parameters of Record-Route. On replies sent by my client I have the following Record-Route and From Header (Note : My Kamailio has two IPs which makes it insert two RR headers ): Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> Record-Route: <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> From: "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this case the restore is done correctly When the client sends me a BYE request I have the following Route Header and To Header Route : <sip:xxx.xxx.xxx.xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_ o=128_11_Y;vsf=AAAAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMH DQgJDQECA3RyAwMcHwIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2>,<sip:xxx.xxx.xxx. xxx;r2=on;lr;ftag=99A730303736313103579CC4;tbk_i=1_2_Y;tbk_o=128_11_Y;vsf=AA AAABgGBAMDAAUBBQAJDndyAwMcHwIdHQsWHwUDNw--;vst=AAAAABgGBAMHDQgJDQECA3RyAwMcH wIdGgQeHAIFAnVzZXI9cGhvbmU-;did=f74.78e2> To : "+351411110097" <sip:I2116446I_500@xxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4 In this scenario, the To after restore is : To :"+351411110097" <sip:Q4525417L_<>Gxxx.xxx.xxx.xxx>;tag=99A730303736313103579CC4, which throws an internal error on Kamailio about bad URI and also makes our MediaGateway reply with a "Bad Request". Can this error be caused by the restore of the uac_replace mechanics? Both Record Route URIs are on different headers while the both Route URIs are on the same Route header. Can this be the reason? If you need more information please let me know. Best Regards David Costa

2 1

Re: [SR-Users] controlling presence with new module `pua_json`
by Henning Westerholt 30 Aug '19

30 Aug '19

Hello Bill, (please keep the list in CC) yes, this seems to be the case. Just looked briefly to the code, it does an assignment in some cases. E.g. if no Presentity* data is given it will just use the From* information. And then it should later output an error from pua_json_update_presentity(..) if there is not the necessary data. Do you see any error message like this? Another idea - does it works if you add more "missing" data in the curl requests? Cheers, Henning Am 28.08.19 um 21:16 schrieb bill(a)novatrope.us: > Henning: Good point. > > I do see a number of errors in the log. It looks like the json routine > is looking for data that is not in the http query as specified by the > author. > > > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Name: [update] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Event-Package: [dialog] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-User] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-User: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Presentity-Realm] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Presentity-Realm: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): From: > [sip:1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-User: [1020101] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Realm: [<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To: > [sip:1020108@<IP of server>:50060] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > To-User: [1020108] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-URI] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-URI: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Call-ID: [1020101@<IP of phone>] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [From-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > From-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [To-Tag] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): To-Tag: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): Json-c > error - failed to extract field [Direction] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): > Direction: [] > 17(125579) DEBUG: json [json_mod.c:70]: _json_extract_field(): State: > [Confirmed] > 17(125579) DEBUG: presence [event_list.c:348]: search_event(): start > event= [dialog/5] > > On 8/28/19 12:00 PM, Henning Westerholt wrote: >> Hello Bill, >> >> just giving some generic debug pointers here: >> >> - do you see any errors in the kamailio log file? >> >> - do you see any SIP traffic (e.g. by observing with ngrep, wireshark >> etc..) to the phone after your curl command? >> >> Cheers, >> >> Henning >> >> Am 28.08.19 um 19:04 schrieb bill(a)novatrope.us: >>> Hi all: >>> >>> I am trying to use the procedure for setting BLF lights described in >>> https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… >>> >>> >>> WIthout success. >>> >>> My configuration is 1 Grandstream GXP2130 phone behind a NAT on a >>> public IP. (IPP in example below) >>> >>> Kamailio is running on public IP (IPK in example) listening on port >>> 50060 >>> >>> The command I am sending: >>> >>> curl -d >>> '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' >>> >>> http://localhost:8080/presence/ >>> {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} >>> >>> >>> >>> >>> Kamailio appears to be processing the command OK, so I think I am just >>> not setting it up properly> >>> >>> Anybody having any luck with this procedure, let me know how you are >>> getting it to work. >>> >>> >>> Bill >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 1

Info: Upgrading kamcli to work with Python3
by Daniel-Constantin Mierla 30 Aug '19

30 Aug '19

Hello, with the perspective that Python2 is going to be removed from the major Linux distributes starting with the beginning of next year, the work to upgrade kamcli to run with Python3 has been started. >From now on, master branch of kamcli is expected to work only with Python3: * https://github.com/kamailio/kamcli There is still work to be done to upgrade all the code, but any use of it should be done with Python3 and eventual issues to be filed on bug tracker. Of course, pull requests are welcome as always! Who wants to use the version for Python2 has to use the code from git branch v1.2-python2. That branch is probably not going to receive any new feature, it will be kept for use with older systems (as well as a running option during the upgrade period to Python3). Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

IMC module not sending messages to members of a conference room
by david＠aslo.us 29 Aug '19

29 Aug '19

2 1

Kamailio 5.0.8 | authentification issue only with TCP/TLS
by Laurent Schweizer 29 Aug '19

29 Aug '19

Dear all, I have a kamailio running in version 5.0.8 and since fee weeks we have an issue with different users connected in TCP or TLS, sometimes authorization like for REGISTER are rejected and after a moment (can be few minute or hours) it work again and of course no change was done in the password .... We see this issue with different device, snom swyx, ... and on UDP we have no issue. I can see that when the Register is rejected it's with the error -2, so wrong password... # Authentication route route[AUTH] { if (is_method("REGISTER")) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { switch($retcode) { case -1: sl_send_reply("503","Service not available"); exit; case -2: xlog("L_WARN", "auth error -2 username $au - src ip: $si \n"); auth_challenge("$fd", "0"); exit; I have attached an example of a trace where we can see a first REGISTER accepted and less than 2 minutes after a new one is rejected. ( in between they is a REGISTER without any Authorization header) Any idea ? BR Laurent

2 3

Connect Kamailio to ITSPs?
by Aingaran Thirunadarajah 29 Aug '19

29 Aug '19

Hello, Newbie here! Have Kamailio set up on a 512MB VPS. Created a couple of users, registered with Zoiper and can call between them. I am looking for documentation to connect the server to other VoIP providers (Signalwire, Anveo, VOIP.ms etc.) to make & receive calls from the external network (DIDs) / PSTN Can someone point me in the right direction? Thanks in advance.

2 1

Define properly IP in record-route header
by przeqpiciel 29 Aug '19

29 Aug '19

Hi, I have some issue and dont have enought knowledge to fix it. I have simple infrastructure. Internet -> router (NAT) -> kamailio -> asterisk Kamailio and Asterisk are inside the network and Kamailio places his own private address in the "record-route" headers instead of who's public. There is configuration to said to him what IP is proper ?

3 8

kamailio+asterisk VoiceMail implementation
by Zhan Bazarov 28 Aug '19

28 Aug '19

Hello, what's best practice for VoiceMail implementation in Kamailio+Asterisk's scheme? should I forward register into asterisk? Or I can keep register on kamailio? Could you, please, shed any lights on this

2 1

Disable GW's on runtime
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, I have two Kamailio's working on a HA scenario. Both Kamailio's have DB replication and each machine has one HA IP. In order to set this up, i have every GW on the DB duplicated, one with Socket IPA and another with socket IPB. I have both GWs in a active state configurated in the replicated DB. They also both have the same URI and the same group id Example : GW1 : Socket = IPA ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 GW2 : Socket = IPB ; GroupID = 1 ; Uri = sip:1.1.1.1:5060 In order to avoid errors, i want to disable on runtime the GWs that have a socket that doesn't belong, at the moment, to that machine (GW with Socket IPB on machine with IPA). To do that i would use the dispatcher.set_state, however it works with group id and address, which in this case it is the same. Is there other method to disable the GWs on runtime using other arguments like duid or socket ? I thought about using probing for this, since it will disable automatically the GWs with non-existent socket, but not all my clients support probing, so it isn't feasible. Thanks in advance

2 5

UAC ha1 password
by Alexandru Covalschi 28 Aug '19

28 Aug '19

Hello, I'm trying to make Kamailio to SIP register on a remote server. However storing plaintext password looks very unsecure. Is there a possibility to store password for uac module using ha1 hash instead of the plaintext password? I see there's a row for it in the database, but in the source code it's not used and it seems i can't neither use it in db neither set up in-memory as avp. Maybe there's a workaround to directly access the in-memory uac registration htable? ________________________________ Regards, Alexandru Covalschi VoIP Engineer and System Administrator tel: +37367367850

3 5

controlling presence with new module `pua_json`
by bill＠novatrope.us 28 Aug '19

28 Aug '19

Hi all: I am trying to use the procedure for setting BLF lights described in https://blog.voipxswitch.com/2018/02/22/kamailio-controlling-presence-with-… WIthout success. My configuration is 1 Grandstream GXP2130 phone behind a NAT on a public IP. (IPP in example below) Kamailio is running on public IP (IPK in example) listening on port 50060 The command I am sending: curl -d '{"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"}' http://localhost:8080/presence/ {"Call-ID":"1020101@<IPP>","Event-Category":"presence","Event-Name":"update","Event-Package":"dialog","Expires":"3600","To":"sip:1020108@<IPK>:50060","To-User":"1020108","To-Realm":"<IPK>:50060","From":"sip:1020101@<IPP>","From-User":"1020101","From-Realm":"<IPP>","State":"Confirmed"} Kamailio appears to be processing the command OK, so I think I am just not setting it up properly> Anybody having any luck with this procedure, let me know how you are getting it to work. Bill

2 1

[tm-local-request] usage
by Duarte Rocha 28 Aug '19

28 Aug '19

Greetings, Does "event_route[tm:local-request]" works like onsend route ? I'm having some problems and i'm using local-request to debug and i found this on the documentation : "Executed after the tm module has sent a local generated, transaction stateful request." I'm logging all the requests that go through this route. Is this really the last place before kamailio send the request? If the request is logged here, is there anything more in kamailio that can stop the request (pipelimit, etc) ? If a packet gets logged on this route does that mean that it was already sent? Best Regards, Duarte Rocha

3 4

Kamailio 4.4.7 - LCR modules
by Laura 28 Aug '19

28 Aug '19

Hi guys, on our enviromet we have 130+ gateway configured over the LCR_gw tables. when we try to reload it.. we get error.. "Aug 28 09:40:36 vm-nextaitz-02 /usr/sbin/kamailio[3800]: ERROR: lcr [lcr_mod.c:1511]: reload_tables(): too many gateways" We reached some top values.. how can we modify it ? Regards

2 2

Dispatcher - FQDN with multiple IPs
by Sergiu Pojoga 27 Aug '19

27 Aug '19

Hi ppl, The problem I'm facing is that function *ds_is_from_list* seems to return false when a gateway's address is a FQDN that resolves into multiple IP addresses (I'm talking about A records), consequently in-dialog requests like BYE and re-Invites get blocked since they don't pass the *ds_is_from_list* validation. Am I missing some parameter or something? Thanks, --Sergiu

3 3

RFC: about sl and tm local-response event routes
by Daniel-Constantin Mierla 27 Aug '19

27 Aug '19

Hello, just discovered what I consider to be an inconsistency in naming and behaviour for event_route blocks for local-request and local-response and starting a discussion here to see how to move on. The event_route[tm:local-request] is executed before sending there local generated request out (allowing also to change its content, drop, etc...). This event route is quite popular event route used when the local generated requests need to be checked or updates. The event_route[tm:local-response] is executed after the response is sent out, obviously no possibility to change anymore the content. I haven't checked the code for event_route[sl:local-response], but based on commit message should be the same. I haven't used the local-response so far at all, today after a discussion on sr-users I wanted to enable kemi callback for tm:local-response and I noticed that behaviour in the code, even I expected to be like local-request (before sending out). I am not sure how much used are the event routes for tm:local-response and sl:local-response, I haven't seen any questions about them so far on mailing lists, that's why I am asking here if would make sense to rename them like tm:local-response-sent and sl:local-response-sent to properly reflect when they are executed. I am expecting that they are very few used so far, so no big head ache with upgrades and bringing some consistency around (this change to be part of next major release). It can stay like now with proper documentation, however in the future if one want and event route for local responses before being sent out, there will be more confusion, imo ... Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

4 5

Too many hops 483 - unable to setup call
by Jack R 27 Aug '19

27 Aug '19

Hi, I installed Kamailio SIP server: Following article: https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… By following step # 12 (as per above article) when we initiate the call we get - Too many hops 483 Created two user earlier: $ kamctl add test testpasswd $ kamctl add test2 testpasswd syslog show: Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56286]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: nathelper [nathelper.c:780]: add_contact_alias_0(): failed to get contact uri Aug 27 09:36:50 localhost /usr/sbin/kamailio[56275]: ERROR: ims_registrar_pcscf [save.c:268]: save_pending(): No contact headers Aug 27 09:36:50 localhost /usr/sbin/kamailio[56276]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (139.54.102.135:5060) to sip:test1@139.54.102.135, ee24f837b04b07334e25befe81217f8f(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: <script>: REGISTER (sip:test1@139.54.102.135 (45.45.0.16:42965) to sip:test1@139.54.102.135, 04e06b6352f7a2b8f97c1a3733b3334a(a)45.45.0.16) Aug 27 09:37:05 localhost /usr/sbin/kamailio[56278]: ERROR: ims_registrar_pcscf [save.c:339]: save_pending(): Will save pending contact without security parameters Aug 27 09:37:05 localhost /usr/sbin/kamailio[56374]: ERROR: <script>: Diameter: AAR failed on subscription to signalling Server IP 139.54.102.135 where Kamailio is installed, UE IP address 45.45.0.16, 45.45.0.1 Followed this https://www.kamailio.org/wiki/tutorials/faq/main#why_the_sip_requests_are_r… Thought to test using SIPp - installed and ran default command: ./sipp –sn uac 127.0.0.1 and ./sipp –sn uac 139.54.102.135 Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 2-60232(a)139.54.102.135) Aug 27 08:51:32 localhost /usr/sbin/kamailio[56295]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist The user sipp and service does'nt exist Added user: kamctl add sipp testpasswd kamctl add service testpasswd Still same error: Aug 27 10:52:17 localhost /usr/sbin/kamailio[56289]: ERROR: ims_ipsec_pcscf [cmd.c:507]: ipsec_forward(): Contact doesn't exist Aug 27 10:52:17 localhost /usr/sbin/kamailio[56284]: ERROR: <script>: INVITE (sip:sipp@139.54.102.135:5061 (139.54.102.135:5061) to sip:service@139.54.102.135:5060, 977-61357(a)139.54.102.135) Please advise. Thanks and Regards.

1 1

kamailio stop registering agents
by Gaurav Bmotra 27 Aug '19

27 Aug '19

hi i m using kamailio 5.1,,, on ubuntu 18.4 installed form source .... after few days it stop registering client to kamailio.... it resolved after restart service ... same after few day loges ---- --------------------- kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Wed 2019-08-21 08:07:44 UTC; 5 days ago Docs: man:systemd-sysv-generator(8) Process: 1936 ExecStart=/etc/init.d/kamailio start (code=exited, status=0/SUCCESS) Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/cfg/cfg_ctx.c:595]: cfg_set_now(): tls.low_mem_threshold2 has been changed to Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [main.c:2636]: main(): processes (at least): 27 - shm size: 67108864 - pkg size: 838 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: dispatcher [dispatcher.c:813]: ds_warn_fixup(): failover functions used, but required AVP Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:153]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/udp_server.c:205]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: ERROR: <core> [core/tcp_main.c:2850]: tcp_init(): bind(c, 0x7fbabac2c1f4, 16) on 172.31.35.49:509 Aug 21 08:07:44 talk /usr/local/sbin/kamailio[1969]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 21 08:07:44 talk kamailio[1936]: * already running Aug 21 08:07:44 talk kamailio[1936]: ...done. Aug 21 08:07:44 talk systemd[1]: Started LSB: Start the Kamailio SIP proxy server. plz help thank you -- *Regards:* Gaurav Kumar

3 3

Re: [SR-Users] [sr-dev] Failed to catch some reply message with reply_route block
by Henning Westerholt 27 Aug '19

27 Aug '19

(moved the discussion to user list) Hello Salah, I see the this output in the log message: 6(26) INFO: <script>: on_reply_route [SIP/2.0 200 OK Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bKe68a.99dc492fe74dcfe1017828521a5fa362.0 2(22) INFO: <script>: on_reply_route [SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.32.0.6;rport=5060;received=172.32.0.6;branch=z9hG4bK178a.79cddb135aabfc42736fd9e9fd826ba8.0 and so on.. So you are seeing the replies correctly, or I wrong? Cheers, Henning Am 22.08.19 um 21:29 schrieb Salah Ahmed: Hello Henning, We already added xlog in the onreply_route and found all messages in between kamailio and UAS. The attached log files in first email has those logs. If any other specific log needed I am here to provide them. Thanks, Salah On Thu, Aug 22, 2019 at 2:24 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, ok, see if you are able to output some easy log with "xlog" in this route, to make sure there is no cfg problem somewhere. Cheers, Henning Am 22.08.19 um 21:15 schrieb Salah Ahmed: Hello Henning, Thanks for quick reply, We armed the t_on_reply just before t_relay() in the route block. t_on_reply("LOGRPL"); if (!t_relay()) { sl_reply_error(); } Thanks, Salah On Thu, Aug 22, 2019 at 2:03 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Salah, the replies are going from the UAS to the UAC over Kamailo as a proxy, with the exception of the hop-by-hop 100. So you should see the 200 OK in the Kamailio in reply_route and onreply_route. Maybe you can check if you armed the onreply_route with t_on_reply for the INVITE routing. Cheers, Henning Am 22.08.19 um 19:50 schrieb Salah Ahmed: Hello, We facing an issue on capturing sip message in Kamailio(Version: 5.2.3). The scenario is very simple. UAC Kamailio UAS |--------INVITE--------->| | | |-------INVITE------->| | |<------100 Trying----| |<-------100 Trying------| | | |<------200 Ok--------| |<-------200 Ok----------| | In this simple scenario, we can't catch any reply messages in between the Kamailio and UAC. We have tried reply_route, onreply_route, and onsend_route. But no one work to grep that reply on that side. onsend_route was bad try as its only for forwarded reply message. Is there any other magic to capture those replies. A debug=3 log message attached here. Thanks, Salah _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

4 11

exit stop procesing or not after a conditional? based on "exit" core docs seems does not explain behaviour
by PICCORO McKAY Lenz 26 Aug '19

26 Aug '19

in my kamailio-asterisk integration i have a exit after the authentication logic, BUT; there's another conditional after the "exit" sentence, so my ask, if there's a exit in that piece of code .. that next conditional will not be executed? Docs seems said "stop execution of script" so all the rest of the logic in file will not be routed? The documentation seems does not explain in good behavior what happened or the kamailio-asterisk wiki page does not have good explanations for that! it's that an error? https://www.kamailio.org/wiki/cookbooks/4.3.x/core#exit - - ``` if (from_uri==myself) {#!ifdef WITH_ASTERISK if (!proxy_authorize("$fd", "sipusers")) {#!else if (!proxy_authorize("$fd", "subscriber")) {#!endif proxy_challenge("$fd", "0"); exit; } if (is_method("PUBLISH")) { if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } } else { if ($au!=$fU) { sl_send_reply("403","Forbidden auth ID"); exit; } } consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; } ``` 8:52 am Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

reply cannot be forwarded - no 2nd via
by Ryan McEvoy 25 Aug '19

25 Aug '19

Hello, I have been having an issue using kamailio to load balance to a SIP/MRCP server. I believe that the issue actually lies with the reply being sent from the final recipient, but I wanted to be sure that I understood the problem and to see if there were any possible solutions within Kamailio. I am using the dispatcher module to dispatch SIP INVITES to media resources, in this case I am just using a single host in a single dispatcher group. The configuration that I have has worked just fine for other SIP/MRCP servers, which is why I think the issue is with this target in particular. The INVITE is routed correctly and contains the headers that it should. The invite comes originally from a Freeswitch, arrives at Kamailio, who should then dispatch to a host in the group. x.x.x.x:4901 is Kamailio x.x.x.x:8090 is Freeswitch y.y.y.y:8000 is the final target This is the invite, as received by the final target after dispatched via Kamailio INVITE sip: x.x.x.x:4901 SIP/2.0 Record-Route: <sip: x.x.x.x:4901;transport=tcp;lr> Via: SIP/2.0/TCP x.x.x.x:4901;branch=z9hG4bK3a9a.7f9a4b588cd75108cfea9a2bd9ffa829.0;i=3 Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S Max-Forwards: 69 From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ To: <sip: x.x.x.x:4901> Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE User-Agent: tts Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE Supported: timer, 100rel Content-Type: application/sdp Content-Disposition: session Content-Length: 281 v=0 o=IPSoft 3367790083977298880 2789991278648035021 IN IP4 x.x.x.x s=- c=IN IP4 x.x.x.x t=0 0 m=application 9 TCP/MRCPv2 1 a=setup:active a=connection:existing a=resource:speechsynth a=cmid:1 m=audio 32038 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=recvonly a=mid:1 The reply that the server sends back appears to be missing a second Via header, which is preventing Kamailio for routing it correctly (based on the error message from core/forward.c) This is the reply sent by the target destination SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.x.x:8090;branch=z9hG4bKStXp06Sv7jU7S To: <sip: x.x.x.x:4901>;tag=ndszib3k From: <sip: x.x.x.x:8090>;tag=4gHgNcNUDjvFQ Call-ID: 6f2a0e9c-3f9e-1238-2aa9-005056854777 CSeq: 8692231 INVITE Contact: "Voiceware MRCP Server" <sip:vwmrcp@ y.y.y.y:8000;transport=tcp> Content-Type: application/sdp Content-Length: 324 v=0 o=- 1566492047 1566492047 IN IP4 y.y.y.y s=Voiceware MRCP Server 2.15.0.7. Dec 28 2018 c=IN IP4 y.y.y.y t=0 0 m=application 8002 TCP/MRCPv2 1 a=channel:12951125463@speechsynth a=connection:new a=setup:passive a=cmid:1 m=audio 30158 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendonly a=ptime:20 a=mid:1 It is correct that there should still be a 2nd Via header in the 200 OK reply from the target host, right? If so, is there anything else that can be done in Kamailio to handle this, or is this mrcp server simply not complying properly with SIP? There’s no funky natting between these hosts or anything like that. Thanks in advance!

2 1

Wrong ACL to Provider kamailio+asterisk+uac
by Zhan Bazarov 25 Aug '19

25 Aug '19

Hello! Please, is it possible to change from/to headers in ACK message to provider? this is exactly our case, please, look it when u get a chance. http://sip-router.1086192.n5.nabble.com/Wrong-ACK-to-Provider-td130050.html…

2 13

set_rtpengine_set question
by Juha Heinanen 24 Aug '19

24 Aug '19

When set_rtpengine_set() is called with two parameters before rtpengine_offer(), e.g. set_rtpengine_set("1", "2"); rtpengine_offer(); it was not clear to me from rtpengine README, if the sets need to reversed when set_rtpengine_set() is called before rtpengine_answer(), i.e., set_rtpengine_set("2", "1"); rtpengine_answer(); or can the order of the sets be the same as it was in case of rtpengine_offer()? -- Juha

1 1

Dealing with optional SRTP
by Mark Boyce 24 Aug '19

24 Aug '19

Hi all (Using Kamailio + RTPEngine, both current stable versions) Just wondering what current best practice is for dealing with “optional SRTP” such as those generated by Yealink devices. (I know this breaks the RFC but lots of things seem to use this method.) Where the device SDP message looks like; m=audio 50004 RTP/AVP 8 97 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:uhJOWOXPuzMeBmrbx2wAKohuveAvZeMEJTzfE7GT a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:yF+EhR3RtxSDCGtLCnEAl3Zrdut4b8cfFQP+hFqf An unencrypted RTP/AVP media line but with crypto lines showing that the device will support SRTP encryption but it is not mandatory. The system of offer SRTP and try and catch on not supported/acceptable here fail feels a little clumsy and relies on how gracefully the device sends “not acceptable here”. I’m wondering if I just short cut it and say that if a device offers a acceptable crypto line then Kamailio sends back a SAVP response. Naturally this question is then closely followed by how to do optional SRTP invites originating from Kamailio without just directly manipulating lines? Feels like there should be an RTPEngine setting for Optional? RTP/AVP+SAVP ? Thoughts? Mark

1 0

Kamailio as an Outbound Proxy - Strange Sip Flow
by Diogo Hartmann 23 Aug '19

23 Aug '19

Hello Kamailio experts, We have a Kamailio as outbound proxy in front of Asterisks PjSIP. There is a strange behavior after 200 OK: - ACK comes from Asterisk Public IP instead of coming from the internal IP address. - BYE can not be processed (404 not here) SNGREP print attached. Has anyone ever experienced this? I can provide more information if needed, just let me know what info. Thanks! Diogo Hartmann

2 5

Cant finish call
by przeqpiciel 23 Aug '19

23 Aug '19

Hi all, I am totally new with Kamailio but i have needs to build small cluster of Asterisk. Right now I could succesfully register to asterisk via kamailio and even make a call to hear tt-weasels but Asterisk cant close call My infrastructure diagram is here http://www.asciidraw.com/#Draw6816041626716434245/903175935 This is what probably I need is to know how to tell to Asterisk that send BYE to private IP of Kamailio instead public IP of Kamailio, am I right ?

1 0

TCP: Strange Device Behaviour
by Carsten Bock 23 Aug '19

23 Aug '19

Hi, I have a device here, which does some sort of strange things: I have SIP Message, e.g. with a length of 1100 Bytes and the devices adds "0" to the end to meet the MTU of the network (for a single TCP frame), some sort of TCP padding. The following TCP Frame contains only the next message. Looking at the trace in Wireshark, sngrep or similar, all of them try to re-assemble the next message in the TCP stream, but however don't decode them as SIP due to to first couple of zero's in the message. Looking at Kamailio, it seems not to have any issues with this and decodes/parses the message properly (they show up on Homer properly); however it also forwards the message INCLUDING the first couple of zero's even on UDP. To make it complitely irritating, the phone itself would ignore such packets, if we would forward them to the phone with starting zeros. Has anyone seen anything like this and is it standards-compliant? Any proper and easy way to get rid of the starting "zero's" from a message before forwarding these messages? I could always "hack" something into src/core/tcp_read.c, but I wonder if there is already a proper way? Any ideas anyone? Thanks, Carsten -- Carsten Bock I Managing Director ng-voice GmbH Millerntorplatz 1 I 20359 Hamburg I Germany www.ng-voice.com Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49 (0)40-52 47 593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock, Dr. David Bachmann Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

1 0

About STIR/SHAKEN - Caller Identity
by Daniel-Constantin Mierla 23 Aug '19

23 Aug '19

Hello, at couple of events I participated during the past few months, I was asked about support of STIR/SHAKEN (caller identity authentication/verification), which is a hot topic these days at least in USA, aiming to combat "fraudulent" robo-calling. Therefore I thought of share some details with everyone in the community about the state in Kamailio, writing to both devs and users, the information being relevant for everyone. We already have the (related) module named auth_identity, available since 2008 (iirc): - https://www.kamailio.org/docs/modules/stable/modules/auth_identity.html But it implements the previous iteration of the specs for caller identity, respectively RFC 4474: - https://tools.ietf.org/html/rfc4474 However, that RFC is obsoleted by 8224 (the latest core specs for STIR/SHAKEN): - https://tools.ietf.org/html/rfc8224 Then, there are also RFCs 8225 and 8226 to add to the core specs. Should anyone be interested to implement STIR/SHAKEN specs in a modules, I would suggest to start from auth_identity -- might not be much work to update it to become conform with latest specs (a new module can be created, of course, even when starting from auth_identity). However, these specs are about signing the SIP request (the INVITE) with special PKI certificate. It can be done easily with embedded scripts such as Lua or Python (inline execution in native kamailio.cfg or using kemi scripts). At Kamailio World 2019, one of the participants I discussed with told me they already implemented using Lua. That's it for a starting point, if anyone wants to discuss more, just reply to sr-users and add your comments or ask the questions. If someone wants to go ahead and work on a C module, announce yourself to avoid duplicate work of others, and use sr-dev if you need assistance on module development. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 2

Unable to start kamailio_pcscf.service
by Jack R 23 Aug '19

23 Aug '19

1. Followed steps from Step # 3 to Step # 16 : https://github.com/open5gs/nextepc/blob/master/docs/_docs/guide/04-setting-… 2. When I run last command >systemctl start kamailio_icscf kamailio_pcscf kamailio_scscfthe root@ubuntu18-04:/etc/init.d# *systemctl start kamailio_icscf kamailio_pcscf kamailio_scscf* Job for kamailio_pcscf.service failed because the control process exited with error code. See "systemctl status kamailio_pcscf.service" and "journalctl -xe" for details. 3. All services start but not kamailio_pcscf: root@ubuntu18-04:/etc/init.d# *systemctl status kamailio_pcscf.service* ● kamailio_pcscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_pcscf; generated) Active: failed (Result: exit-code) since Mon 2019-08-19 09:05:30 UTC; 45min ago Docs: man:systemd-sysv-generator(8) Process: 52271 ExecStart=/etc/init.d/kamailio_pcscf start (code=exited, status=1/FAILURE) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: pv [pv_shv.c:60]: shvar_init_locks(): locks array size 16 Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: ERROR: bad config file (1 errors) Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: 0(52294) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 19 09:05:30 ubuntu18-04 kamailio_pcscf[52271]: * Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Control process exited, code=exited status=1 Aug 19 09:05:30 ubuntu18-04 systemd[1]: kamailio_pcscf.service: Failed with result 'exit-code'. Aug 19 09:05:30 ubuntu18-04 systemd[1]: Failed to start LSB: Start the Kamailio SIP proxy server. lines 1-16/16 (END) Please advise. Thanks and Regards.

2 10

basic Kamailio frontend to Asterisk
by Travis Ryan 22 Aug '19

22 Aug '19

All, I would like to use Kamailio as a frontend/proxy for an asterisk machine/cluster. I'm very well versed in Asterisk, but VERY noob to Kamailio. I've watched so many Kamailio presentations and feel like I've scoured the web for any article that could help me. There is alot out there, but I'm not finding my exact situation. I believe I understand the dispatcher stuff that I'll need down the road, but I'm having issues understanding the setup of my SIP trunk and Kamailio. I used UAC to setup the trunk connection in Kamailio and it sort of worked. I'm not sure what I do for Asterisk. I'd suspect that in this situation Kamailio would handle the registration and trunk connection for me. What role is Kamailio to my Asterisk? Just an Outbound proxy? Do I need to still register the trunk from each Asterisk box "thru" the Kamailio proxy, etc? Also, I'm merely accepting outside calls and then validating the caller and bridging them back out to the PSTN, so I don't have any local SIP clients, etc., so no need to register the sip devices, etc. Thanks! Travis

4 5

Docker image for 5.2.4
by Colin, Andrew 22 Aug '19

22 Aug '19

Hey All Who updates the images on https://hub.docker.com/r/kamailio/kamailio/tags i dont see the latest image kamailio:5.2.4-stretch --

1 0

{param.count,delimiter} issue
by Juha Heinanen 20 Aug '19

20 Aug '19

Looks like there is bug in {param.count,delimiter} transformation: $var(test) = "a=1;b=2"; xinfo("param count = $(var(test){param.count,;})\n"); $var(test) = "a=1,b=2"; xinfo("param count = $(var(test){param.count,,})\n"); syslog: Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 2 Aug 19 11:27:27 salmon /usr/bin/sip-proxy[26513]: INFO: param count = 1 -- Juha

3 10

Slowly increasing PKG memory usage
by Andrew White 20 Aug '19

20 Aug '19

Hi all, I had a Kamailio crash the other day, and some debugging showed I ran out of PKG memory. Since then I’ve run a simple bash script to compile the amount of memory used by all child processes, effective /usr/local/sbin/kamcmd pkg.stats | grep real_used summed together. I’ve graphed out the data, and there’s a clear growth of PKG memory going on, mostly increasing during our busier daytime hours. https://i.imgur.com/UTzx2k1.png Based on this, I suspect either a module loaded or something within my app_ruby conf is leaking memory. I’ve been reading through https://www.kamailio.org/wiki/tutorials/troubleshooting/memory <https://www.kamailio.org/wiki/tutorials/troubleshooting/memory>, but I’m a bit nervous, as I’m not really a C/deep memory type of guy. I can see a GDB script I can attach to Kamailio, but is that going to use significant resources to run or impact the running process? Is there a newer/better/alternative way to do this, and to help me break this down? Thanks! Andrew

3 17

Evapi dispatcher is consuming all the private memory
by José Seabra 20 Aug '19

20 Aug '19

Hello there, I'm using kamailio 5.2.1 with evapi module to send and receive events from an external application. During my stress tests to kamailio i have been noticing that at some point kamailio is getting out of memory: edge-sip-proxy[20565]ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/parser/msg_parser.c:328]: parse_headers(): memory allocation error edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: pv [pv_core.c:704]: pv_get_callid(): cannot parse Call-Id header edge-sip-proxy[20565]: INFO: ESP_LOG: <null>: [evapi:connection-received] EVAPI msg received -> {"aor":"sip:FUZECI1-x1516@uc.fuze.domain ","contact":"<sip:FUZECI1-x1516@10.225.121.206:5060;transport=UDP>"} edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7fb8730b6010, 64); Free fragment not found! edge-sip-proxy[20565]: ERROR: ESP_LOG: <null>: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7fb8730b6010, 64) called from core: core/parser/msg_parser.c: parse_headers(325), module: core; Free fragment not found! The process id 20565 is running out of memory as you can see bellow: output of kamcmd pkg.stats entry: 14 pid: rank: -2 used: 6058696 free: 824 real_used: 8387784 total_size: 8388608 total_frags: 30 This process ID is related with Evapi Dispatcher kamcmd ps | grep 20565 20565 EvAPI Dispatcher After stop the sipp that was generating traffic to kamailio the evapi process still have all the memory consumed Can it be some memory leak in Evapi module? Best Regards José Seabra -- Cumprimentos José Seabra

2 12

HSS download - open source
by Jack R 20 Aug '19

20 Aug '19

Please could you advise where can I get open source HSS that works with latest Kamailio 5.2 release for OS: ubuntu18-04:~$ lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description: Ubuntu 18.04.3 LTS Release: 18.04 Codename: bionic If possible please share the steps to integrate with Kamailio. The link in this article for FHoSS repository is not working: https://www.kamailio.org/wiki/tutorials/ims/installation-howto If not Open source then maybe an evaluation software. Please advise. Thanks and Regards.

2 1

kamailio doesn't reply on internal interface
by Zhan Bazarov 19 Aug '19

19 Aug '19

Hello, we have interesting issue... Some time kamailio comes unreachable on private interface for asterisk, but port checked via telnet and everything is ok... ps. connection via public ip works good root@kamailio:/usr/local/etc/kamailio# lsof -i :5068 | grep LISTEN kamailio 6014 root 13u IPv4 8341589 0t0 TCP kamailio.local.ip:5068 (LISTEN) kamailio 6014 root 14u IPv4 8341590 0t0 TCP kamailio.public.ip:5068 (LISTEN) root@kamailio:/usr/local/etc/kamailio# telnet from asterisk machine: root@asterisk:/etc/asterisk# telnet kamailio.local.ip 5068 Trying kamailio.local.ip... Connected to kamailio.local.ip. Escape character is '^]'. But peer is unreachable in asterisk... More than, kamailio comes reachable when I change transport from TCP to UDP, or when I restart kamailio daemon(but it situation repeats every day).. Any ideas how I can fix it? thanks!

2 1

set_rtpengine_set(set1, set2) issue
by Juha Heinanen 19 Aug '19

19 Aug '19

My sip proxy has two rtpengine sets: # sip-proxy_ctl rtpengine.show all { url: udp:192.26.134.1:6050 set: 0 index: 0 weight: 1 disabled: 0 recheck_ticks: 0 } { url: udp:192.26.134.40:6050 set: 1 index: 1 weight: 1 disabled: 0 recheck_ticks: 0 } Config makes callBefore calling rtpengine_offer(), config calls set_rtpengine_set("1", "0"); rtpengine_offer("..."); As result, debug shows that sip proxy sends TWO offer commands to the rtpproxy in set 1 and NONE to the rtpproxy in set 0. Debug at set 1 engine: Aug 19 18:06:08 buster-10-0 rtpengine[1314]: NOTICE: [5e1407211cc50fd9]: Creating new call Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.000812 sec) Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Received command 'offer' from 192.26.134.1:33284 Aug 19 18:06:08 buster-10-0 rtpengine[1314]: INFO: [5e1407211cc50fd9]: Replying to 'offer' from 192.26.134.1:33284 (elapsed time 0.005247 sec) I haven't been able to figure out, why that is the case. It is a bug, or am I missing something? I added INFO call to set_rtpengine_set_f(): INFO("set1/set2 = %u/%u\n", selected_rtpp_set_1->id_set, selected_rtpp_set_2->id_set); return 1; and it correctly printed: Aug 19 18:06:08 salmon /usr/bin/sip-proxy[26911]: INFO: rtpengine [rtpengine.c:3574]: set_rtpengine_set_f(): set1/set2 = 1/0 Any ideas? -- Juha

1 0

MOH setup
by Maryam Baghdadi 19 Aug '19

19 Aug '19

Hello everyone, I need to setup Music on hold feature and I did all modifications in configuration file and did all steps for setup: 1- load MOH and all requirement modules . 2- rtpproxy setups and works correctly 3- mohqueue and mohqcalls tables created. 4- audio file with type index (novanet.8) exists 5- parameters of MOH module setup and mohq_process() function called in request_route()‌. 6- table mohqueue initiated like this‌: +----+------+----------------------------+-------------------+---------+-------+ | id | name | uri | mohdir | mohfile | debug | +----+------+----------------------------+-------------------+---------+-------+ | 1 | main | sip:main at 191.38.245.2 | /var/kamailio/MOH | novanet | 1 | +----+------+----------------------------+-------------------+---------+-------+ Kamailio starts successfully and no error appears , call setup but on hold doesn't work and no audio file plays. I review all previous mailing lists responses but no one answers completely and before me others had this problem.

4 5

Replacing Caller ID
by Pete Kay 18 Aug '19

18 Aug '19

Hi Could anyone tell me if there is a module for replacing the caller ID in the SIP messages? I need to replace the caller ID when the call comes in and leaves Kamailio. When the egress sends back any response, I need to revert the caller ID back to its normal form. I have a long list of numbers to be used as caller-ID, and I need to keep modifying them without having to restart Kamailio. Do I need to write my own modules for maintaining the list of numbers to be used as caller-id? I am running at 2000 CPS, so I think I need to implement some sort of hash table in order for the caller id to be retrieved in round roubin fashion very quickly. Is there any modules for this purpose or I need to write a cusom one? Pete

2 1

Re: [SR-Users] Wildcarded IMPU
by Henning Westerholt 17 Aug '19

17 Aug '19

Hello Maryam, (please keep the list in CC) understand. There is work ongoing to update the IMS modules (e.g. refer to https://www.kamailio.org/wiki/features/new-in-devel). I don't know about a current effort to add wildcarded AVPs to them, though. If you like to implement something in this regards, it is always good to announce it quickly on the developer list sr-dev to get some feedback and to to synchronize with other developers. Cheers, Henning Am 17.08.19 um 08:04 schrieb Maryam Baghdadi: Hello Henning, Thank you for your response . I read all ETSI‌ standards , latest version of standards are according to Release15 IMS but IMS modules in kamailio support Release 7 IMS. for example there is Wildcarded_IMPU AVP in AUR in R15 IMS which isn't implemented in IMS_icscf module in kamailio . I know all AVP lists are exist in kamailio and I can add my requirements to modules code . I just wanted to know ims modules in kamailio will be updated or not? On Wed, Aug 14, 2019 at 11:55 PM Henning Westerholt <hw(a)skalatan.de<mailto:hw@skalatan.de>> wrote: Hello Maryam, Kamailio supports AVPs and also an extended version called XAVP. About your question - I quickly looked into ETSI TS 129 329 and TS 129 229 but they did not define how this actually works.. Can you elaborate what you actually are looking for? Cheers, Henning Am 05.08.19 um 12:41 schrieb Maryam Baghdadi: hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks _______________________________________________ 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://skalatan.de/services -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

uac_replace_from() and multiple branches
by John Quick 16 Aug '19

16 Aug '19

v5.2.3 of Kamailio. Serial forking is used to route calls to a multiple destinations as described in documentation for the TM module. Also using the DIALOG module - my script calls dlg_manage() before loading the serialised branches. I need to be able to substitute the username part of the From header and have been trying this with calls to uac_replace_from(). modparam("uac", "restore_mode", "auto") modparam("uac", "restore_dlg", 1) There is no problem if all branches use one substituted From username, in which case my script calls uac_replace_from() in the main route. uac_replace_from("sip:" + $avp(route:src) + "@" + $fd); However, I may need to substitute a different value for each branch. Can this be done? I tried moving the call to uac_replace_from() into the branch route, but then I get an error when the final BYE is received: ERROR: uac [replace.c:281]: replace_uri(): decline FROM replacing in sequential request in auto mode (has TO tag) Is there a simple solution to this please? Hoping I can still use UAC auto restore mode because manual restore looks quite tedious/complex. Thanks. John Quick Smartvox Limited www.smartvox.co.uk

3 3

pike parameters doup when have dinamyc ip clients and scanners
by PICCORO McKAY Lenz 15 Aug '19

15 Aug '19

I read documentation for pike usage.. and have a doub.. what it's the best for very dinamyc ip of my devices.. i mean, i'm just curious of that very ironic and problematic scenario: for the scanners i setup fail2ban but only when the scaning are detecte.. but if i have the pike option how this: # this it's my setup for pike due the dinamyc ip and devices over the internet: modparam("pike", "sampling_time_unit", 4) modparam("pike", "reqs_density_per_unit", 80) modparam("pike", "remove_latency", 60) ... route { if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike block $rm from $fu (IP:$si:$sp)\n"); exit; } ... } I put the remove latency in 60, so then due are dinamycally must remian in memory more (due any one will be a possible clilent), just ban if there are 180 (60*3) request each 4 seconds, it's a good configuration or maybe i'm wrong please help me! -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 6

RFC 3860 support - Kamailio is sending SUBSCRIBE back with extra Via header
by Praveen Vatti 15 Aug '19

15 Aug '19

Hi Kamailio support, Basically, I am looking for RFC3860 support - i.e., accept SUBSCRIBE for reg event and send 200OK, NOTIFY to client. With some browsing on Internet I came across Kamailio, which is supporting RFC3860. I have installed and attached is my CFG file. I am sending a SUBSCRIBE request with reg event to Kamailio and it is sending back the same request with extra Via header. Via now contains Kamailio system IP and the source of SUBSCRIBE IP. Please help me in resolving this issue. I have loaded db_mysql, dialplan, tm, sl, presence, presence_reginfo modules. Thanks, Praveen

2 6

Wildcarded IMPU
by Maryam Baghdadi 14 Aug '19

14 Aug '19

hello, I need to know wildcarde_IMPU feature support in kamailio or not? I use kamailio as scscf and need to have support of wildcarded_impu in my scscf. is there any implementation of AVP_Wildcarded_IMPU in kamailio ? thanks

2 1

Uac_replace for header's parameters
by Duarte Rocha 14 Aug '19

14 Aug '19

Greetings, I'm using uac_replace_to() and uac_replace_from() to do some number manipulations. It allows me to modify and automatically restore Display Number and URI. This works nicely, however it only allows to manipulate URI's parameter. Is there any method to the same and include the Header's parameters too? Best regards, Duarte Rocha

2 1

Question regarding Dispatcher + ds_default_socket() modparam + Call-ID generation
by Joel Serrano 14 Aug '19

14 Aug '19

Hello, Simple doubt regarding dispatcher module when you have the ds_default_socket() modparam set: Say you have 2 kamailio boxes, active + passive, one shared IP via keepalived.... On the actrive node, you have dispatcher.send_ping to 1, and the passive has it set to 0. So far OK, only the active box is sending out probing OPTIONS requests, and it's using as outbound socket the virtual IP out of all the available ones (as defined with ds_default_socket() modparam) I have seen in a capture that the outbound OPTIONS look like: OPTIONS sip:190.14.203.219:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bK7abb.6fbaccc1000000000000000000000000.0 To: <sip:190.14.203.219:5060> From: <sip:dspinger@my.domain>;tag=e2bdd495733c59fd91487a137fccad4e-8a73 CSeq: 10 OPTIONS Call-ID: 2019f8491329c382-31419(a)A.B.C.181 Max-Forwards: 70 Content-Length: 0 A.B.C.180 -> VRRP A.B.C.181 -> Physical IP of the box My doubt is, shouldn't ds_default_socket also update the IP used to generate the Call-ID used in the OPTIONS request? Currently, I believe it's using the first defined listen= IP from config script as I switched the order the listen= are defined and I get the desired result: OPTIONS sip:186.188.220.174:5060 SIP/2.0 Via: SIP/2.0/UDP A.B.C.180 ;branch=z9hG4bKc97e.a8d9e1c2000000000000000000000000.0 To: <sip:186.188.220.174:5060> From: <sip:dspinger@my.domain>;tag=5e2e1773f812f6a7e4085c5d036e29d8-d323 CSeq: 10 OPTIONS Call-ID: 7d9a92c218fc1ba0-32111(a)A.B.C.180 Max-Forwards: 70 Content-Length: 0 Is this expected? Cheers, Joel.

3 4

Kamailio/Freeswitch auth user and UserID
by Petarr Jones 14 Aug '19

14 Aug '19

Hi I am new to Kamailio and would like some guidance on linking up with Freeswitch. I have an exiting hosted Asterisk PBX that uses a combination of extension and pbx username (eg 1001-126252581276) and I want to move it over to Kamailio/Freeswitch without re-configuring the SIP phones. I have tried Freeswitch "number-alias" but it does not seem like a good fit due to things like BLF not working and the added overhead etc. Is there a way to configure Kamailio to authenticate a user based on the above username and then send an extension to Freeswitch as the UserID instead, thereby separating the Kamailio authentication from the Freeswitch UserID? As I don't yet have a full understanding of the Kamailio/Freeswitch interaction, I don't know whether this will cause other issues? Thanks in advance Petarr

2 8

Re: [SR-Users] Dispatcher round-robin problems in 5.2
by JR Richardson 14 Aug '19

14 Aug '19

Hey Alex, After I recently upgraded, I had to include more response codes, 5.2 by default only 200 OK is accepted. I had some upstream carriers sending 405 and were taken off-line as a result until I added that code in modparm: modparam("dispatcher", "ds_ping_reply_codes", "class=2;code=405") Not sure if this applies to what you are doing, but my dispatcher round robin distribution acts as it did prior to upgrade to 5.2, pretty even, no anomalies with any nodes getting more calls than any other. JR JR Richardson Engineering for the Masses Chasing the Azeotrope Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- Alex

2 2

Wrong Record-Route and Via header fields when TCP is used
by Leonid Fainshtein 14 Aug '19

14 Aug '19

Hi, Kamailio server has two legs that are connected to different networks. I'm using Kamailio v.5.2.3 and the "enable_double_rr" is implicitly set to "1". The leg "A" IP address is 10.159.65.1 The leg "B" IP address is 192.168.0.31 The call is initiated from 10.159.65.18 According to the "rr" module documentation, function record_route() should insert two "Record_Route" header fields when a request is accepted on one leg is should go out via the second leg. This works as expected in case of UDP protocol: INVITE sip:2000@192.168.6.106:5460;transport=UDP SIP/2.0 Record-Route: <sip:192.168.0.31;r2=on;lr;did=e2c.a191> Record-Route: <sip:10.159.65.1;r2=on;lr;did=e2c.a191> Via: SIP/2.0/UDP 192.168.0.31;branch=z9hG4bKcfa5.d64ecbd87d5315b5993c4ccf16f86537.0 Via: SIP/2.0/UDP 10.159.65.18:5060;rport=5060;branch=z9hG4bK3a9e9a4d But when the TCP protocol is used then the outbound message looks like this: INVITE sip:2005@192.168.0.178:35058;transport=tcp SIP/2.0 Record-Route: <sip:10.159.65.1;transport=tcp;lr;did=bb6.7dc1> Via: SIP/2.0/TCP 10.159.65.1;branch=z9hG4bKc85a.14afc3867dd3220826f9b9940f78168f.0;i=3 Via: SIP/2.0/TCP 10.159.65.18:5060;rport=58616;branch=z9hG4bK1469331f There are two problems there: a) only one Record-Route with leg is inserted b) the added "Via" header field contains the leg "A" IP address instead of expected leg "B" IP address (192.168.0.31). In the LAN trace I see that in reality the message was sent from leg "B". Is it a bug? Best regards, Leonid Fainshtein

2 13

Kamailio v5.2.4 Released
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, Kamailio SIP Server v5.2.4 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.2.x. Deployments running previous v5.2.x versions are strongly recommended to be upgraded to v5.2.4. For more details about version 5.2.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/08/kamailio-v5-2-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

1 0

Planning the release of Kamailio v5.2.4
by Daniel-Constantin Mierla 14 Aug '19

14 Aug '19

Hello, I am considering to do soon a new release out of branch 5.2, respectively v5.2.4. I am aiming for August 12 or August 14, a matter of how backporting is finished, given that next week some devs travel to ClueCon. As usual, if you are aware of issues not reported yet on bug tracker or fixes not yet backported to branch 5.2, act during the next days to have enough time to approach them. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 3

Strange behavior for unregister() function. Wrong docs? Bug?
by Joel Serrano 13 Aug '19

13 Aug '19

Hello, I'm playing around with the unregister() function from the registrar module: https://www.kamailio.org/docs/modules/devel/modules/registrar.html#registra… Docs say that that it can be used in REQUEST_ROUTE or FAILURE_ROUTE. I needed it in REPLY_ROUTE, and before reading the docs I tried it and got this error: Aug 12 15:52:44 csbc01 kamailio: CRITICAL: <core> [core/cfg.y:3526]: yyerror_at(): parse error in config file /etc/kamailio/csbc/reply_routes.cfg, line 27, column 71: Command cannot be used in the block#012 Aug 12 15:52:44 csbc01 kamailio[12737]: ERROR: bad config file (1 errors) But, I have this working on another setup, I compared, and my difference is that I'm running unregister() inside a route, and calling that route from reply_route, and that works (and by works, I mean it also does what it is supposed to do, which is unregister the AoR). So this doesn't work: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } route(DEL_SIPSERVER); } ... But this works: ... onreply_route[MANAGE_REG_REPLY] { if(status=~"2[0-9][0-9]") { if(($sel(contact.expires)==0) || ($hdr(Expires)==0)){ route(UNREG_CUSTOMER); route(DEL_SIPSERVER); } ... ... route[UNREG_CUSTOMER] { if (unregister("location", "$sht(sipserver=>$fU::contact)")) { xlog("L_NOTICE", "[usrloc] removed user location\n"); } } ... So I'm in no-mans-land right now, docs say it's not allowed in REPLY_ROUTE, and if I try it it will fail, but if I try it inside a route, it woks (hack?). What is the correct approach here? Should unregister() be valid in reply_route (like other similar functions, for example: save()) and therefor docs should be updated and the error in the logs should also be updated, or if it's really not supported, shouldn't Kamailio not let me use it anywhere that a ends up being called from reply_route? I hope I have explained myself correctly. As always, please let me know if this should be a GH issue instead and I'll happily create it. Thanks, Joel.

3 5

Dispatcher round-robin problems in 5.2
by Alex Balashov 12 Aug '19

12 Aug '19

Hi, I've got a proxy that does very basic round-robin distribution of calls to an elastic pool of servers, defined in a simple SQLite database with equal priorities. Nothing too complicated, just a vanilla `ds_select_domain("1", "4")`. The only complication is that seldom are most of the gateways enabled. Probing stuff was turned on, but didn't do quite what was desired, so it was replaced with a system that manages activation and deactivation of the gateways externally. Nevertheless, these are the left-over probing settings: modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", "sip:pinger@domain") modparam("dispatcher", "ds_ping_interval", 15) modparam("dispatcher", "ds_probing_threshold", 256) modparam("dispatcher", "ds_inactive_threshold", 1) modparam("dispatcher", "ds_probing_mode", 0) After an upgrade from 5.1 to 5.2.3 (009f62), the round-robin has become extremely "sticky" and tendentious. That is to say, it directs _almost_ all calls to a single gateway -- the first gateway in the table -- though not 100%. There is also a marked tendency to send lots of consecutive calls to the same gateway, even if it's not the first one, though most of the time it is. Has anyone experienced similar upon upgrading to 5.2? Thanks much! -- 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 4

Kamailio DRA - Unable to get diameter_request() function to send
by Nick Jones 10 Aug '19

10 Aug '19

Hi Kamailio users, I'm trying to setup Kamailio as a Diameter Routing Agent, but I just can't seem to get the diameter_request() function to actually send a diameter request, I just get a warning about no JSON Response: *WARNING: ims_diameter_server [avp_helper.c:341]: addAVPsfromJSON(): No JSON Response* Inside my event route I've got one line to send the data to the peer "ims-hss.localdomain": *event_route[diameter:request]{ diameter_request("ims-hss.localdomain", $diameter_application, $diameter_command, $diameter_request); xlog("Forwarded Diameter Request");}* Hard as I try I can't seem to get this request forwarded to this peer and see that next xlog() line. In the config file I've included the cdp_check_peer & cdp_has_app checks to confirm the peer is up, and able to accept the request, (full config in pastebin below) I've defined the peers in the diametercfg.xml config file, and they're all showing as online when I do a "kamcmd cdp.list_peers": * FQDN: ims-hss.localdomain Details: { State: I_Open Disabled: False Last used: 0 Applications: { appid:vendorid: 16777216:10415 appid:vendorid: 16777216:4491 appid:vendorid: 16777216:13019 appid:vendorid: 16777216:0 appid:vendorid: 16777217:10415 appid:vendorid: 16777221:10415 } }* The source of avp_helper.c shows the warning is called if the length of the JSON is <= 0, but as I'm feeding back out what I've received it's not 0, an no recent major changes to the source, so I'm stumped as to why it's catching this. I'm running Kamailio 5.1.2 on Ubuntu 18.04 installed from Repos. Full Kamailio Config: https://pastebin.com/afgqUfWr Diameter CDP Config XML: https://pastebin.com/bVrBG8mG Relevant Syslog: https://pastebin.com/ZY8z2kd4 Kamcmd cdp.list_peers: https://pastebin.com/cKi4JAHC Any ideas as to what am I missing? Thanks in advance! Nick

1 0

need to Configure S-CSCF, I-CSCF and P-SCF
by ganesh nanaiah 10 Aug '19

10 Aug '19

Hi how do we configure the S-CSCF, I-CSCF and P-CSCF in kamalio 5.2.3 to authenicate with the HSS/PCRF, please point us to document to configure the IMS modules regards Ganesh

3 2

kamailio as UAC and NAPTR / sendsocket question
by Karsten Horsmann 09 Aug '19

09 Aug '19

Hi List, I try to register to Deutsche Telekom and there product Deutschland Lan siptrunk. Thats works find but i see an intressting behaviour on selecting the right outgoing interface. Kamailio is sending out with tcp the REQUEST via first private ip configured on that server (172.20.120.53). There is no listen directive for that. I forced NAPTR to use tcp or udp and i assume that kamailio got the right dns answers. On the list i read also that i can use force_send_socket to force the outgoing request. Now my idea - hey i use the $rP for the outgoing to select the right outgoing listen directive. $rP - reference to transport protocol of R-URI But to my surprise the logfile told me thats "UDP" - it sends out via TCP (thats okay). Whats an good transport selector variable from kamailio that works? event_route[tm:local-request] { if(!is_method("OPTIONS")) { xlog("L_INFO", "[tm:local-request] request [$rm] from [$fu] to [$ru] [$rP]\n"); } } INFO: <script>: [tm:local-request] request [REGISTER] from [ sip:+49XXXXXXXX@sip-trunk.telekom.de] to [sip:sip-trunk.telekom.de] [UDP] listen=tcp:2xx.xx.xx.xx:5060 listen=udp:2xx.xx.xx.xx:5060 listen=tls:2xx.xx.xx.xx:5061 advertise CFG_EXT_NAME:5061 listen=udp:172.20.120.55:5060 listen=udp:172.20.120.56:5060 listen=udp:172.20.120.57:5060 listen=udp:172.20.120.58:5060 listen=tcp:172.20.120.58:5060 use_dns_cache=on # use internal DNS cache use_dns_failover=on # depends on internal DNS cache dns_srv_loadbalancing=on dns_try_naptr=on dns_retr_time=1 # seconds before retrying a DNS request dns_retr_no=3 # number of DNS retransmissions dns_naptr_ignore_rfc=yes # ignore target NAPTR priority dns_tcp_pref=30 # TCP has second-highest priority dns_udp_pref=10 # use UDP with least priority tcp_connection_lifetime=3605 # set higher than registration expires #dont' restore modparam("uac","restore_mode","none") modparam("uac","restore_dlg",0) ## UAC REGISTER #!ifdef WITH_UAC_REGISTER modparam("uac", "reg_contact_addr", "CFG_PROD_IP") modparam("uac", "reg_timer_interval", 10) modparam("uac", "reg_retry_interval", 10) modparam("uac", "reg_db_url", DBURL) modparam("uac", "restore_mode", "none") modparam("uac", "auth_username_avp", "$avp(AVP_AUTH_USERNAME)") modparam("uac", "auth_password_avp", "$avp(AVP_AUTH_PASSWORD)") modparam("uac", "auth_realm_avp", "$avp(AVP_AUTH_REALM)") #!endif ip a l 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN group default qlen 1000 link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo valid_lft forever preferred_lft forever inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: ens192: <BROADCAST,MULTICAST,UP,LOWER_UP> mtu 1500 qdisc pfifo_fast state UP group default qlen 1000 link/ether 00:50:56:b5:c1:48 brd ff:ff:ff:ff:ff:ff inet 172.20.120.53/24 brd 172.20.120.255 scope global ens192 valid_lft forever preferred_lft forever inet 2xx.xx.xx.xx/29 scope global ens192 valid_lft forever preferred_lft forever inet 172.20.120.56/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.57/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.58/24 scope global secondary ens192 valid_lft forever preferred_lft forever inet 172.20.120.55/24 brd 172.20.120.255 scope global secondary ens192 valid_lft forever preferred_lft forever inet6 fe80::250:56ff:feb5:c148/64 scope link valid_lft forever preferred_lft forever default via 172.20.120.253 dev ens192 172.20.120.0/24 dev ens192 proto kernel scope link src 172.20.120.53 2xx.xx.xx.xx/29 dev ens192 proto kernel scope link src 2xx.xx.xx.xx -- Kind Regards Mit freundlichen Grüßen *Karsten Horsmann*

4 9

failure_route - no UAC support
by Attila Czébán 09 Aug '19

09 Aug '19

Hello, I'm trying to restart phones with local-requests (from config or kamctl cisco_restart). I can enable t_on_failure("REMOTE_AUTH") in event_route[tm:local-request] but when 401 Reply comes in I get this warning: "tm [t_reply.c:968]: run_failure_handlers(): no UAC support (1, 0)". kamailio -V version: kamailio 5.1.8 (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 with gcc 6.3.0 How can I make it work? Thank you! Regards, Attila

1 0

Kamailio auth via http post.
by Zhan Bazarov 09 Aug '19

09 Aug '19

Hello, Its possible to made authentication via http request to third party Web-API instead of storing users credentials into Dataqbase? Now I looking on auth_ephemeral, but not sure is that my case... Don't see when I have to put API url.. Could anyone, please, shed any light on this issue?

2 1

XMLRPC request URL
by David Villasmil 08 Aug '19

08 Aug '19

Hello guys, I can't find any reference to a pseudovariable containing the request url, i.e.: http://myserver/XMLRPC?server=1.2.3.4 I'd like to get that server variable and manipulate the dispatcher list... the http client can't do XMLRPC or add a body, just an https request.. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Breaking HTABLE changes from v5.2 to v5.3? Function renamed
by Joel Serrano 08 Aug '19

08 Aug '19

Hi all, Just noticed that when updating from 5.2 to 5.3 I saw some errors in Kam logfile regarding HTABLE module commands failing, the reason is the function 'sht_has_name' has been renamed to 'sht_match_name': https://www.kamailio.org/docs/modules/stable/modules/htable.html#htable.f.s… https://www.kamailio.org/docs/modules/devel/modules/htable.html#htable.f.sh… Not sure if updating this doc is automatic or manual but I think it's definitely worth mentioning: https://www.kamailio.org/wiki/install/upgrade/stable-to-devel Just FYI! Cheers, Joel.

2 1

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