- sr-users - kamailio.org

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

09 Jan '20

Hello, Kamailio SIP Server v5.3.2 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.2. For more details about version 5.3.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2020/01/kamailio-v5-3-2-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Ask for Siremis document
by Thanh Tran 09 Jan '20

09 Jan '20

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

2 1

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

09 Jan '20

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

1 1

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

08 Jan '20

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

2 2

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

08 Jan '20

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

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

08 Jan '20

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

2 1

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

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

3 4

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

07 Jan '20

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

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

07 Jan '20

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

2 1

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

07 Jan '20

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

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Siremis & freeswitch
by John Tuxies 07 Jan '20

07 Jan '20

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

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

06 Jan '20

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

1 0

SNI Query
by mahesh b 06 Jan '20

06 Jan '20

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

2 4

Installation issue
by Hassen Bencheikh 06 Jan '20

06 Jan '20

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

5 4

MSRP proxy
by Sergey Safarov 06 Jan '20

06 Jan '20

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

2 1

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

06 Jan '20

Hello! Looks like $T(reply_reason) does not work well (although $T_reply_reason is ok). Kamailio script: failure_route[TRUNK_FAILURE] { ... xlog("L_WARN","rm=$rm / $T(reply_reason) \n"); ... } 0(8894) ERROR: tmx [t_var.c:696]: pv_parse_t_name(): unknown PV name reply_reason 0(8894) ERROR: <core> [core/pvapi.c:952]: pv_parse_spec2(): pvar "T" has an invalid name param [reply_reason] 0(8894) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [)/41] in [$T(reply_reason)] at [15 (5)] 0(8894) ERROR: xlog [xlog.c:511]: xdbg_fixup_helper(): wrong format[rm=$rm / $T(reply_reason) https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables $T(name) - pseudo-variable class to access TM attributes The *name* can be: - reply_reason - reply reason [kamailio]$ kamailio -v version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 18:04:26 Apr 3 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 1

Tweaking TCP settings
by Dovid Bender 06 Jan '20

06 Jan '20

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

2 1

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

06 Jan '20

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

3 8

dlg_set_timeout error
by Pepelux 03 Jan '20

03 Jan '20

Hi all I'm trying to set a dialog timeout when a call is established: event_route[dialog:start] { if(dlg_set_timeout("10")) { xinfo("10 secs"); } } But when the time expired I obtain this error: Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: CRITICAL: dialog [dlg_timer.c:199]: update_dlg_timer(): Trying to update a bogus dlg tl=0x93947b30 tl->next=(nil) tl->prev=(nil) Jan 3 11:24:20 test /usr/sbin/kamailio[18436]: ERROR: dialog [dlg_hash.c:1267]: update_dlg_timeout(): failed to update dialog lifetime # kamailio -v version: kamailio 5.3.1 (i386/linux) flags: USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS 1024, MAX_RECV_BUFFER_SIZE 262144, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 6.3.0 Any idea? Thanks and Happy New Year

3 6

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

03 Jan '20

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

5 10

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

02 Jan '20

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

2 2

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

31 Dec '19

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

3 2

Calculate call duration
by Duarte Rocha 31 Dec '19

31 Dec '19

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

3 2

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

25 Dec '19

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

4 3

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

24 Dec '19

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

2 6

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

24 Dec '19

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

2 1

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

24 Dec '19

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

1 0

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

24 Dec '19

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

2 2

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

22 Dec '19

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

2 1

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

22 Dec '19

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

3 5

Need support
by Srikanth Magloori 21 Dec '19

21 Dec '19

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

4 3

Questions about Presence
by Володимир Іванець 20 Dec '19

20 Dec '19

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

2 2

Kamailio Product Support
by Simal Mathai 20 Dec '19

20 Dec '19

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

2 1

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

20 Dec '19

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

3 5

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

19 Dec '19

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

4 6

Kamailio SIP TLS Troubleshooting
by Sudhakar Parasher 19 Dec '19

19 Dec '19

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

4 10

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

19 Dec '19

Hello! The configuration below leads to a CRITICAL message in the console. Please let me know how to avoid this issue. event_route[htable:mod-init] { route(AUTORUN); ... } route[AUTORUN] { if ( ds_select("9", "4") ) { xlog("L_WARN", "[$cfg(route)] *ds_select* $xavp(_dsdst_[0]=>uri) $xavp(_dsdst_[1]=>uri) $xavp(_dsdst_[2]=>uri)\n"); } ... } dispatcher.list # group 9 9 sip:192.168.20.125:5070 9 sip:192.168.20.160:5070 9 sip:192.168.20.7:5070 9 sip:192.168.20.107:5070 9 sip:192.168.20.190:5070 0(6088) DEBUG: dispatcher [dispatch.c:2054]: ds_manage_routes(): set [9] 0(6088) DEBUG: dispatcher [dispatch.c:2158]: ds_manage_routes(): using alg [4] hash [0] 0(6088) DEBUG: dispatcher [dispatch.c:2199]: ds_manage_routes(): selected [4-9-2/0] <sip:192.168.20.190:5070> 0(6088) DEBUG: dispatcher [dispatch.c:2218]: ds_manage_routes(): using first entry [9/0] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/1] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/2] 0(6088) DEBUG: dispatcher [dispatch.c:2238]: ds_manage_routes(): using entry [9/3] 0(6088) DEBUG: dispatcher [dispatch.c:2272]: ds_manage_routes(): using default entry [9/4] 0(6088) DEBUG: dispatcher [dispatch.c:2016]: ds_select_dst_limit(): selected target destinations: 5 0(6088) WARNING: <script>: [AUTORUN] *ds_select* sip:192.168.20.190:5070 sip:192.168.20.107:5070 sip:192.168.20.7:5070 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list 0x7fe5f7d4f420 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/usr_avp.c:636]: destroy_avp_list(): destroying list (nil) 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list 0x7fe5f7d4faf0 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4faf0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f168), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f378), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f580), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f790), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/xavp.c:495]: xavp_destroy_list(): destroying xavp list (nil) 1(6089) CRITICAL: <core> [core/mem/q_malloc.c:514]: qm_free(): BUG: freeing already freed pointer (0x7fe5f7d4f9a0), called from core: core/xavp.c: xavp_free(65), first free core: core/xavp.c: xavp_free(65) - ignoring 1(6089) DEBUG: <core> [core/receive.c:457]: receive_msg(): cleaning up ... version: kamailio 5.2.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 19:29:40 Jun 20 2019 with gcc 4.8.5 -- BR, Denys Pozniak

2 2

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

19 Dec '19

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

2 1

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

18 Dec '19

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

6 11

kamcmd stuck
by Andrew Chen 17 Dec '19

17 Dec '19

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

2 2

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

16 Dec '19

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

3 6

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

16 Dec '19

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

2 4

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

16 Dec '19

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

2 1

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

15 Dec '19

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

2 1

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

15 Dec '19

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

1 0

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

14 Dec '19

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

3 3

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

13 Dec '19

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

4 16

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

12 Dec '19

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

3 2

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

12 Dec '19

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

1 0

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

12 Dec '19

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

2 4

xhttp_prom Metric Naming Issue
by ENGIN ESIN 12 Dec '19

12 Dec '19

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

2 1

Please Help
by Sharath 12 Dec '19

12 Dec '19

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

2 1

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

10 Dec '19

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

4 6

authcheck and fd variable
by Robert Lieback 10 Dec '19

10 Dec '19

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

2 2

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

09 Dec '19

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

1 0

duplicated registrations
by Liu, Charles 08 Dec '19

08 Dec '19

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

2 2

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

07 Dec '19

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

1 1

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

06 Dec '19

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

2 6

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

06 Dec '19

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

2 1

Scripts still valid?
by David Villasmil 05 Dec '19

05 Dec '19

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

3 4

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

05 Dec '19

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

2 2

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

05 Dec '19

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

2 2

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

05 Dec '19

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

3 2

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

05 Dec '19

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

2 3

anycast + dmq_usrloc
by Nuno Miguel Reis 05 Dec '19

05 Dec '19

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

2 2

anycast + dmq_usrloc
by Nuno Miguel Reis 04 Dec '19

04 Dec '19

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

1 0

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

04 Dec '19

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

3 2

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

04 Dec '19

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

3 2

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

04 Dec '19

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

2 1

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

04 Dec '19

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

1 1

secfilter usage
by Aymeric Moizard 03 Dec '19

03 Dec '19

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

2 5

kamailio IMS supporting VOLTE
by Shubhendu Das 03 Dec '19

03 Dec '19

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

2 2

Contact header requirement
by David Villasmil 02 Dec '19

02 Dec '19

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

3 7

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

02 Dec '19

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

2 5

Kamailio World 2020 - Call For Presentations
by Daniel-Constantin Mierla 02 Dec '19

02 Dec '19

Hello, I would like to announce that Call for Presentations at Kamailio World 2020 is now open. You can submit your proposal or see more details at: - https://www.kamailioworld.com/k08/call-for-speakers/ The 8th edition of the event takes place again in Berlin, Germany, during April 27-29, 2020. Expect over 150 participants, developers and community members as well as representatives of other popular open source VoIP projects such as Asterisk or FreeSwitch. Looking forward to meeting many of you there! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Possible conflict between t_newtran and setflag for ACC
by Patrick Wakano 01 Dec '19

01 Dec '19

Hello list, Hope you all doing well! I am using the ACC module and using the setflag() function as done in several examples. It works fine. However, I've added the t_newtran() function almost in the begging of the INVITE handler to help the retransmission detection and after that I noticed the ACC was not saving anything in DB. So after debugging I discovered that if I call the t_newtran() before setting the ACC flags, the module will not save the calls in DB, but if I call it after setting the ACC flags, it works.... So my question is, is this a bug or it is a expected side effect so when one is using t_newtran you must be careful and set all your transaction flags before? (ACC are the only transaction flags I am using so can't tell if other modules have the same problem) This is happening in Kamailio 5.2.2. Thank you! Kind regards, Patrick Wakano

3 6

tcp limit, too many open files
by David Villasmil 01 Dec '19

01 Dec '19

Hello guys, I'm facing an issue where, if i start kamailio as root, all works properly, but if i start it as kamailio, i get hundreds of: handle_new_connect(): error while accepting connection(24): Too many open files My limits is like: kamailio soft core unlimited kamailio soft data unlimited kamailio soft fsize unlimited kamailio soft memlock unlimited kamailio soft nofile unlimited kamailio soft rss unlimited kamailio hard stack 240 kamailio soft cpu unlimited kamailio soft nproc unlimited kamailio soft as unlimited kamailio soft priority -11 kamailio soft locks unlimited kamailio soft sigpending unlimited kamailio soft msgqueue unlimited kamailio soft nice -11 kamailio soft nofile 65536 kamailio hard nofile 65536 root soft core unlimited root soft data unlimited root soft fsize unlimited root soft memlock unlimited root soft nofile unlimited root soft rss unlimited root hard stack 240 root soft cpu unlimited root soft nproc unlimited root soft as unlimited root soft priority -11 root soft locks unlimited root soft sigpending unlimited root soft msgqueue unlimited root soft nice -11 root soft nofile 65536 root hard nofile 65536 I can't figure out why this is happening... Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

7 17

Kamailio as SBC for Microsoft Teams
by Henning Westerholt 29 Nov '19

29 Nov '19

Hello, I have created a "How-To" blog post on using Kamailio as "session border controller" for Microsoft Teams Direct Routing: https://skalatan.de/de/blog/kamailio-sbc-teams You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

7 10

Adding "q" value for the main branch
by Maxim M 28 Nov '19

28 Nov '19

Maxim Malygin <maxim.malygin(a)gmail.com> 17:08 (5 минут назад) кому: sr-users Hello, I need to form list of Contact headers with "q" parameters for "302 Moved Temporary". I do something like this: ... # $xavp(contacts) - contains list of contact URIs (uri) and Q values (q) $var(i) = 0; $var(num) = $cnt($xavp(contacts)); while($var(i) < $var(num)) { if($var(i) == 0) { # How to add q value to this branch? $ru = $xavp(contacts[$var(i)]=>uri); } else { append_branch("$xavp(contacts[$var(i)]=>uri)", "$xavp(contacts[$var(i)]=>q)"); } } send_reply("302", "Moved Temporarily"); ... Is it possible to set "q" value for the main branch? For additional branches I do it via append_branch("uri", q_value) but I didn't find any way to do it for the main branch pointed via $ru/$du. The pseudo-variable "$branch" also doesn't allow me to access the main branch. Thanks, Maxim

1 0

Get totally rid of H.264 SDP
by Kjeld Flarup 28 Nov '19

28 Nov '19

Hi I have a door telephone, with GSM fallback. Since telephone operators are going away from SS7, more and more traffic goes SIP. Thus we now have some numbers where we can see that doing an invite with H.264 causes them to fail. I have tried to remove coded 96 with sdp_remove_codecs_by_id, but it still leaves most of the video SDP in the invite. SDP received by kamailio Media Description, name and address (m): video 6100 RTP/AVP 96 Media Attribute (a): rtcp:6101 IN IP4 192.168.237.239 Media Attribute (a): rtpmap:96 H264/90000 Media Attribute (a): fmtp:96 profile-level-id=42000d;packetization-mode=1 Media Attribute (a): sendrecv After sdp_remove_codecs_by_id I send this. But without "96" Media Description, name and address (m): video 6534 RTP/AVP Media Attribute (a): sendrecv Media Attribute (a): rtcp:6535 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): candidate:r9KO0wNPMTGktklS 1 UDP 2130706431 192.168.237.15 6534 typ host Media Attribute (a): candidate:r9KO0wNPMTGktklS 2 UDP 2130706430 192.168.237.15 6535 typ host My problem is that the door telephone does not do an ACK when it gets the 200 OK My suspicion is that it does not understand the video response Media Description, name and address (m): video 0 RTP/AVP 0 Media Attribute (a): ice-ufrag:35ZngQdL Media Attribute (a): ice-pwd:DG6kUbcYwBFKNS6YzCG27wRlG4 Media Attribute (a): sendrecv Is there a way to remove the whole video part and not just the coded id? It is not an option to disable video in the door telephone, as we need the video. It is just in the fallback is must be removed. -- --------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

1 0

Kamailio does not detect message
by Adesh Pandey 28 Nov '19

28 Nov '19

Hi, I have below scenario: User registers on Kamailio server, calls to a number which is forwarded to asterisk through dispatcher, asterisk server answers and sends a text message to the caller using sendText but first message is delivered to client with message "501 Not Implemented" but in the subsequent calls asterisk shows that the text was delivered but kamailio did not detect the message. Guys, any help would be appreciated, I am using the below mentioned config. https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg Kamailio version: 5.2

1 0

Kamailio developers meeting - follow up remarks
by Daniel-Constantin Mierla 27 Nov '19

27 Nov '19

Hello, last week we had the 2nd annual Kamailio Developers Meeting hosted by Sipgate in Dusseldorf, Germany: * https://www.kamailio.org/w/developers-meeting/ 16 people were at the event in various roles. Giacomo Vacca published on his blog a good summary of what happened there: * https://www.giacomovacca.com/2019/11/my-notes-on-kamailio-developer-meeting… This year we had a lot of discussions, as well as work done on multiple planes, not only Kamailio code. So I am trying to list here some of the conclusions for future development, the technical aspects of the meeting, so everyone is aware and can provide feedback. 1) Effective work was done on: * kamailio code * kamailio rpm packaging * kamailio tools (kamctl) * kamailio release process * kamailio project keys (to be used to sign the packages) 2) Documentation 2.a) Wiki * it was somehow a rough consensus to move the wiki content to github, along with changing the format from dokuwiki markdown to the standard/github markdown. This should enable people to make pull requests so developers or community members can review and aprove new content. It also makes it easier to contribute using existing github account, now the kamailio.org/wiki is requiring to make a dedicated account, which many prefer not to do it. * the presentation can be done either by using mkdocs to generate html files hosted on kamailio.org or using the github provided wiki portal. 2.b) Docs for variables and transformations * there was a proposal to move them in the documentation the modules that export them, there are pros and cons, needs more discussions. Now they are in the wiki, so this probably has to resume after deciding on 2.a). 3) Kamailio Modules 3.a) replication (dmq) - several participants discussed about negotiations between nodes to take active role on some cases (e.g., active dialogs) 3.b) api integration - quite some interest in JSON-based API routing, concluding in extending rtjson to cover more use cases 3.c) security - have options to restrict the use of TLS1.3 or newer 4) Kamailio Releases * v5.3.2 was released during the event, allowing to document the process * work to automatize the process is planned, then eventually assing teams for takeing cares of releases from specific branches 5) Kamailio Testing * existing docker-based testing framework should be extended and integrated in CD/CI pipeline 6) Kamailio packages 6.a) rpms * rpm.kamailio.org has been prepared and is expected to take over the opensuse build service for building rpms and hosting them. Expected to provide support for hosting many kamailio versions in the same release series so one can do downgrade to older releases. Also, there is work in progress to provide nightly builds. 6.b) debs * work is planned to offer many kamailio versions in the same release series 7) Kamailio tools * kamctl/kamdbctl should be obsoleted in favor of kamcli, which offers a better framework for input validation and output formatting, as well as better portability, no longer depending on shell interpreter 8) Various discussions * kemi exports from C point of view and how to combine the documentation for modules and their kemi exports * how to make kamaiio friendlier in virtualized environments (ended up in the need of making the use of advertised address a bit more dynamic) * project organizatoric topics - to be approached separately * next events - Fosdem - someone should submit a proposal to present about Kamailio 9) Long term goals We speak here more or less about Kamailio 6.0 ... * change the behaviour of the native config interpreter to be consistent with the other programming languages in terms of handling the response code (change what is now: the evaluation of negative value to false and positive value to true and the hidden return 0 to exit) * make the pool of processes more generic, so they can handle traffic from more sockets (being sip traffic or something else) -- this should make better use of resources, as some sockets might be less busy that others I hope I covered the important topics, if I remember something else, I will reply on this thread. Or maybe other participants can contribute missing topics. Should anyone have comments or suggestions on the above topics, or new ones, let's discuss on sr-users because it impacts the long term use of Kamailio (sr-dev is cc-ed now for notification purposes). Overall, there were 2 very intensive days, however in a friendly and relaxed environment offered by Sipgate. Extremely useful discussions, not only about Kamailio, but about RTC ecosystem. At the evening social network event event, a couple of external people joined, some of them presenting very interesting use cases of Kamailio. Many thanks to all participants that allocated time and resources to come to Dusseldorf, as well as to the companies that covered expenses for participants. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 2

Kamailio xavp_contact - Segmentation Fault
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, Kamailio is crashing when i'm trying to set the parameter; modparam("tm|usrloc", "xavp_contact", "ulattrs") That's happening with Kamailio 5.3.1 and 5.2.5 too. Crash is happening when Im register 2 devices with the same extension. This is the core dump: [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib64/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f /usr/local/etc/ka'. Program terminated with signal 11, Segmentation fault. #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 288 core/xavp.c: No such file or directory. (gdb) bt #0 0x000000000057cca2 in xavp_get_internal (name=0x8, list=0x0, idx=0, prv=0x0) at core/xavp.c:288 #1 0x0000000000581869 in xavp_insert (xavp=0x0, idx=0, list=0x0) at core/xavp.c:752 #2 0x00007f05be0ee3a1 in ki_t_next_contacts (msg=0x7f05cb76c218) at t_serial.c:552 #3 0x00007f05be0f09e8 in t_next_contacts (msg=0x7f05cb76c218, key=0x0, value=0x0) at t_serial.c:756 #4 0x0000000000434d41 in do_action (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1067 #5 0x00000000004418e8 in run_actions (h=0x7ffe93c9cad0, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1572 #6 0x0000000000441fa9 in run_actions_safe (h=0x7ffe93c9de90, a=0x7f05cb642250, msg=0x7f05cb76c218) at core/action.c:1636 #7 0x000000000065734e in rval_get_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, i=0x7ffe93c9cf78, rv=0x7f05cb642570, cache=0x0) at core/rvalue.c:912 #8 0x000000000065b8fe in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9cf78, rve=0x7f05cb642568) at core/rvalue.c:1910 #9 0x000000000065bd51 in rval_expr_eval_int (h=0x7ffe93c9de90, msg=0x7f05cb76c218, res=0x7ffe93c9d42c, rve=0x7f05cb642c98) at core/rvalue.c:1918 #10 0x0000000000434807 in do_action (h=0x7ffe93c9de90, a=0x7f05cb645028, msg=0x7f05cb76c218) at core/action.c:1043 #11 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb637e48, msg=0x7f05cb76c218) at core/action.c:1572 #12 0x0000000000431767 in do_action (h=0x7ffe93c9de90, a=0x7f05cb583858, msg=0x7f05cb76c218) at core/action.c:691 #13 0x00000000004418e8 in run_actions (h=0x7ffe93c9de90, a=0x7f05cb574750, msg=0x7f05cb76c218) at core/action.c:1572 #14 0x0000000000442071 in run_top_route (a=0x7f05cb574750, msg=0x7f05cb76c218, c=0x0) at core/action.c:1657 #15 0x00000000005874a9 in receive_msg (buf=0xa6ec00 <buf.6868> "OPTIONS sip:10@192.168.0.231:5060 SIP/2.0\r\nVia: SIP/2.0/UDP 192.168.0.231;branch=z9hG4bKd8a9.1b3abe656532065414a2f35a8916008d.1\r\nVia: SIP/2.0/UDP 192.168.0.131:5060;received=192.168.0.131;branch=z9hG"..., len=694, rcv_info=0x7ffe93c9e4d0) at core/receive.c:341 #16 0x000000000047bf81 in udp_rcv_loop () at core/udp_server.c:541 #17 0x0000000000424f9d in main_loop () at main.c:1669 #18 0x000000000042c688 in main (argc=13, argv=0x7ffe93c9ea18) at main.c:2710 Thank you.

2 1

Kamailio Parrallel Forking
by Ilie Soltanici 27 Nov '19

27 Nov '19

Hello, I'm trying to use parallel forking if one extension is registered from multiple devices. (using WebSockets) This is the related code from kamailio.cfg: ------------------------------------------------------------------------------------------------------------------------------------ if (!t_load_contacts()) { xlogl("L_WARN", "Error loading contacts for $rU\n"); sl_send_reply("500", "Server Internal Error (Code:$cfg(line))"); exit; } else { xlogl("L_INFO", "Contacts loaded for $rU\n"); } if (!t_next_contacts()) { xlogl("L_INFO", "t_next_contacts - Only one contact found for $rU, calling\n"); } else { xlogl("L_INFO", "t_next_contacts - Multiple contacts found, parallel forking\n"); } route(RELAY); ------------------------------------------------------------------------------------------------------------------------------------ I noticed that on Kamailio 5.2.5 - this code is working fine, and Kamailio is sending an INVITE to both Devices.But, on Kamailio 5.3.1 - that's not working anymore. For some reason, kamailio is sending just one INVITE to one device with wrong URI. The URI is set to the second device instead of the correct one. Was something changed in the t_next_contacts function related to this? Thank you

1 0

Kamailio 5.2.3 segfaulting
by Daniel Greenwald 27 Nov '19

27 Nov '19

I'm seeing this in the log and kamailio is crashing every hour or so: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 33/var/log/kamailio/kamailio.log Any ideas? Should I emergency upgrade to 5.3.1?

2 12

No Audio using RTP Proxy
by Sujit Roy 27 Nov '19

27 Nov '19

Hello I am facing a problem as below. Please suggest for the work around. My call flow is like this. SIP Gateway-1 (IP x.179) -> SIP Gateway-2 ( IP x.177) -> Kamalio+RTPProxy So when the call arrives at Kamalio+RTPProxy, i m getting below in log. Nov 26 23:25:31 rtpproxy[18508]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 1b7c870763616c6c15fff410@ 192.168.100.177, tag 1aa18fc201a68168;1 requested, type strong Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_command_ul_handle: new session on IPv4 port 15920 created, tag 1aa18fc201a68168;1 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTP address with 192.168.100.177:27360 Nov 26 23:25:31 rtpproxy[18508]: INFO:1b7c870763616c6c15fff410@ 192.168.100.177:rtpp_stream_prefill_addr: pre-filling caller's RTCP address with 192.168.100.177:27361 But x.177 is working on signalling mode only ( Not routing Media ) . As a result, i m not getting any voice from IP x.179 What can be done to change the caller's RTP address to x.179 in RTPProxy ? Thanks -- Regards =================== Sujit Roy

2 1

Encode host ip in Contact header for semi-stateless setup
by Daniel Greenwald 27 Nov '19

27 Nov '19

At kamailio world 2017, Jan Lorenz of Pascom explained how they use kamailio in a semi-stateless manner and encode the asterisk's IP into the contact header on the way out so that it can be decoded and used to route properly on the way back without Record-Routing. I would like to create such a setup. I'm looking for pointers if there is a particular module/method that would be ideal to to use for this. Any other gotcha's to consider in such a setup. My plan is to keep the transaction on the same kamailio by not altering the VIA but allow the subsequent in-dialog transactions to potentially hit another kamailio by changing the contact to a shared SRV record (and encoding the freeswitch box's IP into it for later retrieval.) Thanks in advance. Daniel Greenwald

4 4

dialog module: Does dlg_manage() have to be called for all requests?
by George Diamantopoulos 27 Nov '19

27 Nov '19

Hello all, I've been reading through the dialog module docs again, and there seems to be a discrepancy in what's suggested in the docs. In the intro, it is stated that 'To create the dialog associated with an initial INVITE request, execute the function “dlg_manage()"'. Later on, in section "7.10" the following example code is provided: request_route { ... if(is_method("INVITE") && !has_totag()) { $dlg_ctx(timeout_route) = "DLGTIMEOUT"; $dlg_ctx(timeout_bye) = 1; } dlg_manage(); ... } In this example code, dlg_manage seems to be called for all requests, as it is not dependent on the conditional that only requests with no to-tag are to be handled. Previously, Daniel had suggested to me in person that one also run dlg_manage() for CANCELs, and in-dialog reINVITEs, BYEs and ACKs to mitigate some buggy corner cases. Is this still the case? What do you guys and gals do in production? Have you had to revise how you use dlg_manage in terms of calling it for initial INVITEs only vs all requests? Thanks! BR, George

5 6

Modify SIP Message Header and Message Body parameter
by Sujit Roy 26 Nov '19

26 Nov '19

Hello Please see below SIP Message SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 192.1681.101:5060;branch=z9hG4bK5a61798d7c850d39;rport=5060 From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567@ 192.1681.102> Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE *Server: kamailio (5.1.9 (x86_64/linux))* Content-Length: 0 *1. I want to Change the Server Parameter . How to do it ? * SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.1681.101:5060;rport=5060;branch=z9hG4bK5a61798d7c850d39 Record-Route: <sip: 192.1681.102;lr> From: <sip:summona@ 192.1681.101>;tag=51a0dc1e620de16d To: <sip:101880171234567 @ 192.1681.102>;tag=as728725b3 Call-ID: 6fd9b9d563616c6c15f2d7fa@ 192.1681.101 CSeq: 21701 INVITE Server: VOS3000 V2.1.4.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 600;refresher=uas *Contact: <sip:1018801751481417@104.216.39.66:5060 <http://sip:1018801751481417@104.216.39.66:5060>>* Content-Type: application/sdp Require: timer Content-Length: 289 v=0 o=root 912643161 912643161 IN IP4 192.1681.104 s=VOS3000 V2.1.4.0 c=IN IP4 192.1681.104 t=0 0 m=audio 10784 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:230 a=sendrecv *a=nortpproxy:yes* 2. I Want to hide the Contact Parameter and a= nortpproxy:yes ==> How to do it as well ? Thanks in advance. -- Regards =================== Sujit Roy

2 1

Beginner Help: Relay vs Forward
by Michael Iedema 26 Nov '19

26 Nov '19

Hi all, I am trying to place Kamailio between several SIP services which are differentiated by port and by the type of message they handle. To be concrete, here are the services: Originator: - reachable on port 5060 - originates all traffic to the services Service A: - reachable on port 5061 - handles REGISTER Service B: - reachable on port 5062 - handles INVITE Service C: - reachable on port 5063 - handles MESSAGE The services, originator and Kamailio instance may all be run on the same IP or may be split onto different addresses. Previously, the originator needed to know the address and port of each service. Now I want to make Kamailio responsible for that task so it can be done more intelligently. To start though I simply want to get traffic flowing between the services and this is where I’m struggling to climb the learning curve. In my test setup the originator and all services are on the same IP with only Kamailio running on a different IP. Address 1 Address 2 ======================== Originator <--> Kamailio:6060 | | Service A <-------> | Service B <-------> | Service C <-------> | My first attempt at the logic looks like this: # HOST DEFINITIONS #!substdef "/HOST_ORIGINATOR/192.168.86.110/" #!substdef "/HOST_REGISTER/192.168.86.110/“ #!substdef "/HOST_INVITE/192.168.86.110/" #!substdef "/HOST_MESSAGE/192.168.86.110/" # PORT DEFINITIONS #!substdef “/PORT_ORIGINATOR/5060/" #!substdef "/PORT_REGISTER/5061/“ #!substdef "/PORT_INVITE/5062/" #!substdef "/PORT_MESSAGE/5063/" # ROUTE LOGIC route { # ORIGINATOR sent us something if ( $(ct{nameaddr.uri}{uri.port}) == HOST_ORIGINATOR ) { xlog("L_INFO", "ORIGINATOR: sent $rm\n”); if (is_method("REGISTER")) { $du = "sip: HOST_REGISTER:PORT_REGISTER"; xlog("L_INFO", "ORIGINATOR -> REGISTER: Destination URI is now $du\n"); t_relay(); exit; } } … #### END SNIPPET #### My very newbie questions are: - Am I even on the right track here? I understand that t_relay() establishes a stateful dialog handler between the originator and register services. - Can I accomplish this bi-directional path statelessly with forward()? - Do I need to add logic to handle the messages from the REGISTER service back to ORIGINATOR? This config currently fails because of "no corresponding socket found” when the relay triggers. I believe this is because I’m running the Kamailio instance in a Docker container so my first step today is to set up a native install. I also looked at the dispatcher module which seems very close to what I want to do. Maybe that’s a more natural way to go? Anyway, apologies for the lengthy explanation. I hope it is at least clear what I intend to do. I will gladly research and do trial and error to solve this but need to know I’m going down roughly the right path. I’m not sure what I should even be googling even of the time :-) Thanks as always for your time and the great software! Regards, -Michael

4 5

Re-use TCP connections on different transactions
by Joel Serrano 26 Nov '19

26 Nov '19

Hello, I'm setting up a Kamailio instance behind a TCP load balancer (with proxy protocol and NAT routing: meaning Kam stays in the flow all the time). I've managed to get working almost everything we need for our service, except for one thing, and that is for Kam to use existing connections for subsequent transactions: Following this example: [image: image.png] EXT & INT represent the external and internal interface of a LB between the UAC and Kamailio, using TLS on both legs and proxy protocol. Transaction 1: INVITE, 100, 180, 183, 200 OK UAC 1.1.1.1:1111 -> 2.2.2.2:443 (EXT) 3.3.3.3:3333 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 2: ACK UAC 1.1.1.1:1112 -> 2.2.2.2:443 (EXT) 4.4.4.4:4444 (INT) -> 7.7.7.7:5060 (Kamailio) Transaction 3: BYE Kam 7.7.7.7:5060 -> 3.3.3.3:3333 (INT) 2.2.2.2:443 (EXT) -> 1.1.1.1:1111 UAC My problem is with Transaction 3. In this case the BYE is originated by the callee, and Kam has to send it to the caller. As the TCP load balancer is between Kam and the UAC, Kam has to send it to the LB so then the LB can forward it back to the UAC. This works well for msgs that belong to the same transaction (INVITE, 100, 180, 183, 200 OK) but it fails when they don't belong to the same transaction. Thanks to the newly added $tcp(c_si) and $tcp(c_sp) pseudovars, I can save the internal IP:Port of the LB, so I can send stuff later to it, my problem is that Kam doesn't seem to allow this? On the original INVITE, I use the following to save where I have to reach the UAC: add_contact_alias("$tcp(c_si)", "$tcp(c_sp)", "tls"); Then, handle_ruri_alias() will take care of setting $du to the correct (internal LB) IP:Port so I can reach the UAC, this works. My problem is that Kamailio doesn't identify that there is a valid existing TLS connection still up (from the INVITE), and tries to create a new one (and this obviously doesn't gives all sorts of problems). So when I run handle_ruri_alias(), and $du is set to 3.3.3.3:3333 (from the example above), instead of using the existing connection, Kamailio tries to create a new one. I have a log statement right before with the result of tcp_conid_state(1) (the connid is 1 for this connection) and the $rc is 1 (Connection is OK), but when I tell Kamailio it has to use it I get this in the logs: DEBUG: {1 11726467 BYE gqR1qqNK8B} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} And then the problems begin... I have tried playing around with: tcp_reuse_port tcp_connection_match But no luck..! I also thought it could be a problem of the connection being created on one worker, and a different worker handling BYE transaction, so tested with children=1 and tcp_children=1, but still same problem. A more detailed log: In blue my log statement checking for the status of conid "1", in red Kam not being able to find it, although it exists (as validated by tcp_conid_state(), and even in netstat I can see the connection established). In this log, 35.191.0.66:60271 would be the equivalent of 3.3.3.3:3333 and 104.175.176.242:28157 would be 1.1.1.1:1111 from the example above. ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1657]: _tcpconn_find(): found connection by id: 1"} "message":" NOTICE: {1 11727734 BYE 5-LX4GdI9X} <script>: JOEL TEST New request - M=BYE TCP STATUS:1 ID=5-LX4GdI9X"} ... "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1144]: handle_ruri_alias(): setting dst_uri to <sip:35.191.0.66:60271;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} nathelper [nathelper.c:1166]: handle_ruri_alias(): rewriting r-uri to <sip:linphone@104.175.176.242:28157;transport=tls>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:1328]: t_newtran(): msg (0x7f3c884259d0) id=534\/18664 global id=532\/18664 T start=0xffffffffffffffff"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:497]: t_lookup_request(): start searching: hash=63496, isACK=0"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:455]: matching_3261(): RFC3261 transaction matching failed - via branch [z9hG4bK808f.eee2444f92a02cb33e1b7a21f20bc6bb.0]"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_lookup.c:675]: t_lookup_request(): no transaction found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tm [t_hooks.c:336]: run_reqin_callbacks_internal(): trans=0x7f3c83b8c598, callback type 1, id 0 entered"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/crypto\/md5utils.c:67]: MD5StringArray(): MD5 calculated: 3071029feb05962b26b53a9664a14210"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1095]: check_route_param(): route params checking against [;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes] (orig: [r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes])"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} rr [loose.c:1101]: check_route_param(): params are <;r2=on;lr=on;ftag=Eb~TbdfTA;did=cab.01e2;nat=yes>"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} siputils [checks.c:123]: has_totag(): totag found"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/msg_translator.c:161]: check_via_address(): (198.1.54.228, 198.1.54.228, 0)"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:2060]: tcp_send(): no open tcp connection found, opening new one"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 35.191.0.66"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1242]: tcpconn_new(): on port 60271, type 3"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} <core> [core\/tcp_main.c:1561]: tcpconn_add(): hashes: 1446:2350:0, 5"} "message":" DEBUG: {1 11727734 BYE 5-LX4GdI9X} tls [tls_server.c:199]: tls_complete_init(): completing tls connection initialization"} ... So time to seek help from the community, any suggestions/ideas/comments? Sorry if all this sounds confusing, I've tried my best to put in text the whole scenario in and "understandable" way... Is this even doable? Thanks, Joel.

3 10

How to enforce REGISTER with AKAv1 (IMS)
by Moshe A 26 Nov '19

26 Nov '19

Hello, How can we enforce REGISTER with AKAv1? Here our setup 1. kamailio version 4.3.4 for IMS 2. Boghe IMS client v1.0.58.550 3. scscf.cfg is configure with: #!define REG_AUTH_DEFAULT_ALG "AKAv1-MD5" 4. Set Authentication Schemes in Private User Identity - IMPI for user bob to: Digest-AKAv1 (3GPP) results: First REGISTER fails with 401 Unauthorized Second REGISTER 200 OK, but the REGISTER is MD5 and not AKAv1 Attached is tcpdump. By the way: in which DB should we find Authentication Schemes? can't find it in hss_db, kamailio and not in scscf. Thank you, Moshe.

1 0

call not going to end point
by Gaurav Bmotra 26 Nov '19

26 Nov '19

hi i m using kamailio 5.4 dev Docker image when i try to call form calientA to clientB i got (Temporarily Unavailable (code: 480)) i m using freeswitch as media server so the flow and structure should be like this ClientA---->Kamailio------>FS---->Kamailio>---->ClientB but issue is (what i m getting ) ClientA---->Kamailio------>FS---->Kamailio>-------->FS-------->kamailio--------->FS IT CREATE THE CALL LOOP b/w KAMAILIO AND FREESWITCH *MY CONFIG FILE IS ATACHED * in logs i m getting -------------------------------------------------------- Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 38902: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "18_4 Uc0,8,101 9857c00a-8947-1238-c6a7-12590b724429 3.82.96.70 22082 jKrKymKNg480p;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 9857c00a-8947-1238-c6a7-12590b724429, tag jKrKymKNg480p;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 35888: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "20_4 Uc0,8,101 98589959-8947-1238-c6a7-12590b724429 3.82.96.70 29846 m5a51aNvapN6D;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 98589959-8947-1238-c6a7-12590b724429, tag m5a51aNvapN6D;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 37706: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "13_4 Uc0,8,101 985962c1-8947-1238-c6a7-12590b724429 3.82.96.70 29432 pQXp50p3471BN;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985962c1-8947-1238-c6a7-12590b724429, tag pQXp50p3471BN;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 45078: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:get_command: received command "16_5 Uc0,8,101 985a3ce3-8947-1238-c6a7-12590b724429 3.82.96.70 25096 r9F88praZSeHc;1" Nov 24 10:25:36 ip-172-20-61-37 local0.info rtpproxy[6]: INFO:GLOBAL:rtpp_command_ul_handle: new IPv4/IPv4 session 985a3ce3-8947-1238-c6a7-12590b724429, tag r9F88praZSeHc;1 requested, type strong Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:create_twinlistener: can't bind to the IPv4 port 58376: Address not available (99) Nov 24 10:25:36 ip-172-20-61-37 local0.err rtpproxy[6]: ERR:GLOBAL:rtpp_command_ul_handle: can't create listener Nov 24 10:25:36 ip-172-20-61-37 local0.debug rtpproxy[6]: DBUG:GLOBAL:rtpc_doreply: sending reply "E72\n" -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- *here is my config file * #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.4 - 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 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # # *** To use RTPEngine (instead of RTPProxy) for nat traversal execute: # - define WITH_RTPENGINE # - install RTPEngine: https://github.com/sipwise/rtpengine # - start RTPEngine: # rtpengine --listen-ng=127.0.0.1:2223 ... # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:LOCAL_IP:5091 advertise PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/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" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" #!ifdef WITH_RTPENGINE loadmodule "rtpengine.so" #!else loadmodule "rtpproxy.so" #!endif #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- #-------------------dispatcher params------------- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/run/kamailio/kamailio_ctl") # ----- sanity params ----- modparam("sanity", "autodrop", 0) # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT #!ifdef WITH_RTPENGINE # ----- rtpengine params ----- modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") #!else # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") #!endif # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); $rcv_addr="sip:"+$si+":"+$sp; if(starts_with("$rcv_addr","$du")) { route(LOCATION); } else { route(RELAY); } exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Per SIP request initial checks route[REQINIT] { # no connect for sending replies set_reply_no_connect(); #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($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|sipvicious|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP request from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; #!ifdef WITH_RTPENGINE if(nat_uac_test("8")) { rtpengine_manage("replace-origin replace-session-connection"); } else { rtpengine_manage("trust-address replace-origin replace-session-connection"); } #!else if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } #!endif if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } if(isbflagset(FLB_NATB)) { # no connect message in a dialog involving NAT traversal if (is_request()) { if(has_totag()) { set_forward_no_connect(); } } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies reply_route { if(!sanity_check("17604", "6")) { xlog("Malformed SIP response from $si:$sp\n"); drop; } } # Manage incoming replies in transaction context 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 } -- *Regards:* Gaurav Kumar

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How to get the audio codec used for each established call.
by Abdoul Osséni 25 Nov '19

25 Nov '19

Hello all, I try to save in the CDR the audio codec used for each established call. Can you help me? Regards Abdoul.I

3 4

SDP Modification
by Sujit Roy 25 Nov '19

25 Nov '19

Hello How can i change c= and o= in SDP using RTPProxy ? Please advise Thanks -- Regards =================== Sujit Roy

2 1

Passing media thru RTP Proxy
by Sujit Roy 25 Nov '19

25 Nov '19

Hello I have installed and configured both Kamailio and RTP Proxy and calls are going. But i m unable to force media to go thru RTP Proxy. How can i enforce media to go thru RTP Proxy ? Thanks -- Regards =================== Sujit Roy

3 2

Kamailio stopped relaying SIP text message from asterisk
by Adesh Pandey 25 Nov '19

25 Nov '19

Hi, I have been working on kamailio with following scenario: 1. Webrtc SIP Users register to Kamailio 2. Users initiate outgoing calls to some other numbers 3. Kamailio dispatches outgoing calls to Asterisk 4. Asterisk answers the call and then send a text message to the caller using SendText, which was working fine a few days back but all of sudden it has stopped working 5. At Asterisk end it is showing that message was delivered successfully but when I check SIP user logs, there is no message event. Please review my config file at following URL: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4 Developers, I also want to know if there is any way to log *text message* in kamailio server.

3 3

Unable to connect to sip server via proxy
by Ashwini Kumar 24 Nov '19

24 Nov '19

Hi I have two network interfaces on my system in which one interface is connected to my office WAN/LAN and other is connected to SIP private network connected to PSTN. Now, I want to access sip server using the internal IP of our office LAN as proxy. *192.168.6.2 (Office LAN IP as proxy) -> 10.50.185.2 (SIP)* So on machine 192.168.6.2, I setup kamailio as SIP proxy server so that our office LAN machine can connect to SIP server directly via SIP proxy. I used config defined in - *https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamailio-minimal-proxy.cfg* <https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…> and I changed the local domain alias with SIP IP -* 10.50.185.2*. When I try to connect to SIP via proxy using *linphone* (UDP) I got below errors in* /var/log/syslog* and shows *connection in progress* every time. Nov 23 15:15:49 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:53 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:15:57 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:01 ubantudesk /usr/sbin/kamailio[25785]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Nov 23 15:16:05 ubantudesk /usr/sbin/kamailio[25784]: WARNING: {1 20 REGISTER ek82LguQ6z} sanity [sanity.c:585]: check_cl(): content length header missing in request Can you help me what I should do? -- *Thanks & Regards,* *Ashwini Kumar | Technology |* www.industrybuying.com | Ghitorni, Delhi - 110030 | India | È*+91-8802137782* | * ashwinikumar(a)industrybuying.com |

2 1

Avoid full table scan for dispatcher table on db_redis using ds_reload() from config script
by Joel Serrano 23 Nov '19

23 Nov '19

Hello, I'm trying out redis as db backend, and right now I only have dispatcher records there... Every so often, I do a ds_reload() from within Kam config script, and I see in logs: Nov 22 20:36:35 test COPS[25531]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query After enabling debug logs: Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1761]: db_redis_query(): querying prefix (table) 'dispatcher' Nov 22 21:15:35 test COPS[26661]: DEBUG: db_redis [redis_dbase.c:1811]: db_redis_query(): no columns given to build query keys, falling back to full table scan Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:119]: db_new_result(): allocate 56 bytes for result set at 0x7f0be4273898 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:156]: db_allocate_columns(): allocate 40 bytes for result names at 0x7f0be4273938 Nov 22 21:15:35 test COPS[26661]: DEBUG: <core> [db_res.c:167]: db_allocate_columns(): allocate 20 bytes for result types at 0x7f0be42739c8 Nov 22 21:15:35 test COPS[26661]: WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'dispatcher' while performing query Given the msg... "no columns given to build query keys, falling back to full table scan", I assume I'm missing a keys modparam.. but I don't know how to build it, and the docs (to me) seem confusing? https://kamailio.org/docs/modules/devel/modules/db_redis.html#db_redis.p.ke… Can anyone help me understand how to build the keys modparam for dispatcher table? Or if the reason for this is different, what do I need to do to avoid the full table scan and the warning in the log? Thanks, Joel.

1 1

Question about ims_ipsec_pcscf
by Woscek, Martin W. 22 Nov '19

22 Nov '19

Hi all, We are looking for information as to what the ims_ipsec_pcscf module was written for and tested against. What was the target UE/EPC/PCRF that Tsvetomir Dimitrov wrote the module towards? Has an emulated EPC been tested with this module? If so which one(s)? Thanks in advance, Martin

1 0

Cannot read an indexed xavp from Kemi-Lua
by Michel Pelletier 22 Nov '19

22 Nov '19

Hi, I am having trouble reading an indexed xavp from Kemi-Lua. The xavp contains 3 entries but I can only access the first one. The xavp is the one filled by the dispatcher module after a call to KSR.dispatcher.ds_select(group, 4) with the flags parameter set to 2. I can get the number of entries using KSR.pvx.xavp_child_get("ds_ctx", "cnt") which returns 3. I can also get the contents of the first element using, for example, KSR.pvx.xavp_child_get("ds_dst", "attrs"). But all my attempts at adding an index value using KSR.pvx.xavp_get() or KSR.pvx.xavp_child_get() return the value 'nil'; Below is the output of KSR.pvx.pv_xavp_print(): +++++ start XAVP list: (nil) ((nil)) (level=0) ----- end XAVP list: (nil) (level=0) +++++ start XAVP list: 0x7f94ce01f1f8 ((nil)) (level=0) *** (l:0 - 0x7f94ce01f1f8) XAVP name: ds_ctx XAVP id: 1762531321 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01f150> +++++ start XAVP list: 0x7f94ce01f150 (0x7f94ce01f218) (level=1) *** (l:1 - 0x7f94ce01f150) XAVP name: cnt XAVP id: 7279890 XAVP value type: 1 XAVP value (int): 3 *** (l:1 - 0x7f94ce01f150) end ----- end XAVP list: 0x7f94ce01f150 (level=1) *** (l:0 - 0x7f94ce01f1f8) end *** (l:0 - 0x7f94ce01eaf8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ea30> +++++ start XAVP list: 0x7f94ce01ea30 (0x7f94ce01eb18) (level=1) *** (l:1 - 0x7f94ce01ea30) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey3;maxload=200 *** (l:1 - 0x7f94ce01ea30) end *** (l:1 - 0x7f94ce01e988) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01e988) end *** (l:1 - 0x7f94ce01e8c8) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01e8c8) end ----- end XAVP list: 0x7f94ce01ea30 (level=1) *** (l:0 - 0x7f94ce01eaf8) end *** (l:0 - 0x7f94ce01edd0) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01ed08> +++++ start XAVP list: 0x7f94ce01ed08 (0x7f94ce01edf0) (level=1) *** (l:1 - 0x7f94ce01ed08) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey2;maxload=200 *** (l:1 - 0x7f94ce01ed08) end *** (l:1 - 0x7f94ce01ec60) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ec60) end *** (l:1 - 0x7f94ce01eba0) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01eba0) end ----- end XAVP list: 0x7f94ce01ed08 (level=1) *** (l:0 - 0x7f94ce01edd0) end *** (l:0 - 0x7f94ce01f0a8) XAVP name: ds_dst XAVP id: 1762531862 XAVP value type: 6 XAVP value: <xavp:0x7f94ce01efe0> +++++ start XAVP list: 0x7f94ce01efe0 (0x7f94ce01f0c8) (level=1) *** (l:1 - 0x7f94ce01efe0) XAVP name: attrs XAVP id: 1835799177 XAVP value type: 2 XAVP value (str): duid=newjersey1;maxload=200 *** (l:1 - 0x7f94ce01efe0) end *** (l:1 - 0x7f94ce01ef38) XAVP name: grp XAVP id: 7056653 XAVP value type: 1 XAVP value (int): 1 *** (l:1 - 0x7f94ce01ef38) end *** (l:1 - 0x7f94ce01ee78) XAVP name: uri XAVP id: 8122237 XAVP value type: 2 XAVP value (str): sip:123.123.123.123:5060 *** (l:1 - 0x7f94ce01ee78) end ----- end XAVP list: 0x7f94ce01efe0 (level=1) *** (l:0 - 0x7f94ce01f0a8) end ----- end XAVP list: 0x7f94ce01f1f8 (level=0) Regards, Michel Pelletier

1 0

kamailio call loop issue B/W kamailio and freeswitch
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi im using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. ibut the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS -- here is my config file data --- #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi thank you for reply i have check but same issue ,,, is the config file data plz check if you can help thank you ------------------------------------ #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_ACCDB #!define WITH_NAT # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.org> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes for each UDP socket * - value inherited by tcp_children and sctp_children when not set explicitely */ children=8 /* uncomment the next line to disable TCP (default on) */ # disable_tcp=yes /* number of SIP routing processes for all TCP/TLS sockets */ # tcp_children=8 /* uncomment the next line to disable the auto discovery of local aliases * based on reverse DNS on IPs (default on) */ # auto_aliases=no /* add local domain aliases */ # alias="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to * bind on a specific interface/port/proto (default bind on all available) */ # listen=udp:10.0.0.10:5060 listen=udp:172.20.LOCAL_IP:5091 advertise 3.83.PUBLIC_IP:5091 /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 /* upper limit for TCP connections (it includes the TLS connections) */ tcp_max_connections=2048 #!ifdef WITH_TLS enable_tls=yes /* upper limit for TLS connections */ tls_max_connections=2048 #!endif ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" loadmodule "dispatcher.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "xavp_dst", "_dsdst_") modparam("dispatcher", "xavp_ctx", "_dsctx_") # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains #route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN #route(PSTN); route(DISPATCH); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif if($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(is_method("OPTIONS") && uri==myself && $rU==$null) { sl_send_reply("200","Keepalive"); exit; } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") { if(sd_lookup("speed_dial")) { route(SIPOUT); } } #!endif #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) { route(SIPOUT); } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } route(RELAY); exit; } # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { route(TOVOICEMAIL); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif # if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; } route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { if(ds_next_dst()) { xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # IP authorization and user authentication route[AUTH] { #!ifdef WITH_AUTH #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=myself) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==myself) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif } ------------------------------------------------------------------------- On Fri, Nov 22, 2019 at 4:57 PM Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > hi thank you for reply > i have check but same issue > i have attached my config file ... can you check if there is any issue > > thank you > > On Fri, Nov 22, 2019 at 4:41 PM David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > >> Hello, >> >> You’re only pasting the dispatcher route. >> >> You need to follow the script logic. An invite comes in, how does it know >> the call is coming from the client and has to be sent to freeSWITCH? >> Conversely, how does it know it’s coming from freeswitch? >> >> Probably using the permissions module with >> >> https://kamailio.org/docs/modules/3.1.x/modules_k/permissions.html#id2528655 >> >> You can check if the $si is the freeswitch, then send to clientB >> >> On Fri, 22 Nov 2019 at 11:40, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> there is something wrong in your config file, failing to detect the >>> traffic from freeswitch. Load debugger module and enable cfgtrace logging, >>> then see what actions are executed in the configuration file, maybe you can >>> spot what goes wrong there. >>> >>> Otherwise, to get more help from the mailing lists, the full config is >>> needed. >>> >>> Cheers, >>> Daniel >>> On 22.11.19 09:51, Gaurav Bmotra wrote: >>> >>> hi >>> i m using kamailio 5.3 as register server and using dispatcher i m >>> dendign the calls to freeswitch .. >>> >>> but the issue is then freeswitch send back call to kamailio after >>> processing media ... it should go back to client but it again send call to >>> freeswitch ,, which create the loop and hangup the call with (Temporarily >>> Unavailable (code: 480)) >>> >>> >>> *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 >>> >>> *but I am getting this * >>> >>> client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS >>> >>> >>> *Routing Logic *------------------------------- >>> if(!ds_is_from_list()) { >>> >>> route(DISPATCH); >>> } >>> >>> -------- >>> # Dispatch requests >>> >>> >>> >>> route[DISPATCH] { >>> >>> >>> >>> # round robin dispatching on gateways group '1' >>> >>> >>> >>> if(!ds_select_dst("1", "4")) { >>> >>> >>> send_reply("404", "No destination"); >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: >>> $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> # Try next destionations in failure route >>> >>> >>> >>> failure_route[RTF_DISPATCH] { >>> >>> >>> >>> xnotice("RTF_DISPATCH: $rm $rU [$ci]"); >>> >>> >>> >>> if (t_is_canceled()) { >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> # next DST - only for 500 or local timeout >>> >>> >>> if (t_check_status("(^5)") >>> >>> >>> or (t_branch_timeout() and !t_branch_replied())) >>> { >>> >>> xnotice("RTF_DISPATCH: WARNING $du is broken and marked >>> as inactive"); >>> >>> >>> # ds_mark_dst("ip"); >>> >>> >>> >>> if(ds_next_dst()) { >>> >>> >>> >>> #xdbg("--- SCRIPT: retrying to <$ru> via <$du> >>> (attrs: $xavp(_dsdst_=>attrs))\n"); >>> >>> >>> t_on_failure("RTF_DISPATCH"); >>> >>> >>> >>> route(RELAY); >>> >>> >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> >>> } >>> >>> >>> -- >>> thank you >>> >>> >>> >>> >>> >>> >>> >>> *Regards:* >>> Gaurav Kumar >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda >>> Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work(a)gmail.com >> phone: +34669448337 >> > > > -- > > > > > > > > > *Regards:* > Gaurav Kumar > > -- *Regards:* Gaurav Kumar

1 0

disparcher making loop with call
by Gaurav Bmotra 22 Nov '19

22 Nov '19

hi i m using kamailio 5.3 as register server and using dispatcher i m dendign the calls to freeswitch .. but the issue is then freeswitch send back call to kamailio after processing media ... it should go back to client but it again send call to freeswitch ,, which create the loop and hangup the call with (Temporarily Unavailable (code: 480)) *what i want *client1------>Kamailio------>FS-----Kamailio----->Client2 *but I am getting this * client1------>Kamailio------>FS-----Kamailio----->FS----->kamailio----->FS *Routing Logic*------------------------------- if(!ds_is_from_list()) { route(DISPATCH); } -------- # Dispatch requests route[DISPATCH] { # round robin dispatching on gateways group '1' if(!ds_select_dst("1", "4")) { send_reply("404", "No destination"); exit; } xdbg("--- SCRIPT: going to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destionations in failure route failure_route[RTF_DISPATCH] { xnotice("RTF_DISPATCH: $rm $rU [$ci]"); if (t_is_canceled()) { exit; } # next DST - only for 500 or local timeout if (t_check_status("(^5)") or (t_branch_timeout() and !t_branch_replied())) { xnotice("RTF_DISPATCH: WARNING $du is broken and marked as inactive"); # ds_mark_dst("ip"); if(ds_next_dst()) { #xdbg("--- SCRIPT: retrying to <$ru> via <$du> (attrs: $xavp(_dsdst_=>attrs))\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } -- thank you *Regards:* Gaurav Kumar

3 2

How to retrieve local socket that will be used to send a message from the branch route
by Patrick Wakano 22 Nov '19

22 Nov '19

Hello list, Hope you are all doing well! I am trying to figure out a way to retrieve the local socket the t_relay() has decided to use in a mhomed=1 env. Is there a way to do that? I couldn't find any variable other than the $snd ones that maybe have this info, but I am not using onsend_route anyway..... I am after this value because this is what I can use to figure out if I need to engage RTPEngine with parameters "external, internal", "external, external", "internal, external" or "internal, internal". In my case, my server is in between two networks, and calls can traverse or can be routed back to the same interface depending on what was decide by the LCR module. Signalling is fine, but engaging the RTPEngine must be done with the correct parameters to get the correct IPs in the SDP, other else I get one way audio in the call..... Any help is much appreciated! Thank you! Kind regards, Patrick Wakano

4 7

Adding SIP Gateway Endpoint in Kamailio
by Sujit Roy 21 Nov '19

21 Nov '19

Hello Here is my call flow scenario. SIP Gateway (A) Kamailio (K) Asterisk (AST) SIP Gateway (B) Now i want to send calls from A -> B by using Asterisk as media server. Kamailio shall be used to authenticate A and allow A to send calls to B. What are the configurations i need to make in Kamailio and Asterisk ? Thanks in advance. -- Regards =================== Sujit Roy

3 6

How to access different uris in a header
by Duarte Rocha 21 Nov '19

21 Nov '19

Greetings, In order to identify Redirecting Number and Original Called Number on a given request i look for the top( Redirecting Number) and bottom (Original Called) Diversion headers using hdr(Diversion)[0] and hdr(Diversion)[-1]. However, i have a client that sends me all the Diversions URIS in the same header. Does kamailio has a similar tool for accessing the different URIS in the header? Best Regards,

2 1

CentOS Repo for Kamailio 5.3 - Strange Python23 Dependency?
by Zach Nielsen 21 Nov '19

21 Nov '19

Hi all, Working with some of our Centos-based Kamailio boxes lately has produced a strange result when using the provided yum repos (download.opensuse.org). When trying to yum update the system or install some additional modules, we're getting feedback from yum indicating that it's expecting /usr/bin/python23 --> Finished Dependency Resolution Error: Package: kamailio-5.3.1-3.1.x86_64 (home_kamailio_v5.3.x-rpms) Requires: /usr/bin/python23 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest Tried my best to do my due diligence here to make sure there was nothing up with our local env that was causing this, but this issue persists across multiple systems, including a new (no frills) one that we just build out. Can I get some confirmation if this is a "me" issue, or if maybe there's an issue with what made it into the repo? Also, just a general question... what's the typical way of you folks deploying Kamailio from a sysadmin perspective? Do you generally compile from source, or do any of you use the provided repositories to manage it? Thanks! Zach

5 6

kamailio Dispatcher error
by Gaurav Bmotra 20 Nov '19

20 Nov '19

hi i want to use dispatcher Module and following is the config.. --------------------------------------- loadmodule "dispatcher.so" -------------- modparam("dispatcher", "db_url", "mysql://kamailio:kamailiorw@localhost /kamailio") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "ds_hash_size", 2) -------------------------------------------------------------- but when i hit $ kamailio out put: ERROR: bad config file (4 errors) ------------------------------------------------------ logs : kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled; vendor preset: enabled) Active: failed (Result: exit-code) since Wed 2019-11-20 15:16:34 IST; 16s ago Process: 25747 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255/EXCEPTION) Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Service RestartSec=100ms expired, scheduling restart. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Scheduled restart job, restart counter is at 5. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Stopped Kamailio (OpenSER) - the Open Source SIP Server. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Start request repeated too quickly. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: kamailio.service: Failed with result 'exit-code'. Nov 20 15:16:34 gaurav-Latitude-E7240 systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. ------------------------------------------------------------- plz help me -- Thank you *Regards:* Gaurav Kumar

4 6

Dynamic routing and REGISTER
by Igor Olhovskiy 20 Nov '19

20 Nov '19

Hi! I’m trying to get `drouting` module working for REGISTER. But seems it’s not passing one of internal checks, cause in REGISTER R-URI does not contains username and I’m getting errors drouting [drouting.c:567]: build_ruri(): stripping 0 makes username <> null drouting [drouting.c:970]: do_routing(): failed to build ruri So, as I got, this module is mainly for calls (means INVITE) routing, rather than to use it like one more dispatcher, or it’s also a way to handle this part here? Thanks, Igor

3 5

Query regarding Kamailio Sip Server for IMS supporting VOLTE
by Shubhendu Das 20 Nov '19

20 Nov '19

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

1 0

How to intercept an INVITE after (ts_append)
by Oleg Podguyko 20 Nov '19

20 Nov '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using rtpgngine module I proxy media stream via rtpengine(rtpengine_manage()). I change SDP. SiP2SIP calls work fine. Now I would like to have possibility to "wake up" sip-clients(at the android and ios phones) which are usually not on-line via push notification. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { ts_store(); # if it a new transuction ,it will be store for some time. } } switch ($var(rc)) { case 1: route(RELAY); break; case -1: break; case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu"); # if transuction for this sip-client is exist. The invite will be send. exit; } If the client is not registered at the kamailio during the call. Then such calls are stored through the "ts_store" function for some time. And if during this time the client send register, kamailio will be sent INVITE to it. But I have the problem. Such calls after "wake up"go through a special branch route that I don't know how to intercept. As a result, I can't change SDP(using rtpengite_manage commands) and rtp stream do not proxy via rtpengine… and call failed. /usr/sbin/kamailio[9957]: exec: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=428 a=26 n=ts_append /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tsilo [ts_append.c:72]: ts_append(): transaction 36343:92182831 found for 777777777, going to append branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:255]: lookup_helper(): contact for [777777777] found by address /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} registrar [lookup.c:456]: lookup_helper(): instance is /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:77]: t_append_branches(): transaction 36343:92182831 in status 0 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:99]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (1) outgoing branches /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:77777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/msg_translator.c:2933]: create_via_hf(): id added: <;i=1>, rcv proto=2 /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:777777777@95.165.146.181:63618;transport=TCP;rinstance=8b1865b95767b7ba] with ruid [uloc-5dd2faa0-26e5-1] /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} tm [t_append_branches.c:163]: t_append_branches(): Call uvAzTGsEV9N2ReDjmWoo8A..: 0 (0) outgoing branches after clear_branches() /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (9957 (22)), reusing fd /usr/sbin/kamailio[9957]: DEBUG: {1 2 REGISTER JxPcA9D1hIJ8PmkcnOvI7g..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... -- Oleg Podguyko

2 2

TLS v 1.3 failed call
by Andrew Chen 19 Nov '19

19 Nov '19

Hi guys, I was wondering if someone can help decipher what these few lines mean? Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from 10.94.98.18:51698 using TLSv1.3 TLS_AES_256_GCM_SHA384 256 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: 206.81.191.45:443 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: tls [tls_server.c:1189]: tls_read_f(): Reading on a renegotiation of connection (n:1652) (0) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1527]: tcp_read_req(): EOF Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0x56367f8b8f80, 12, -1, 0x10) fd_no=2 called Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1680]: release_tcpconn(): releasing con 0x7fcfc20c57b0, state -1, fd=12, id=23 ([10.94.98.18]:51698 -> [10.94.98.18]:443) Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21501]: DEBUG: <core> [core/tcp_read.c:1684]: release_tcpconn(): extra_data 0x7fcfc2089f88 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: <core> [core/tcp_main.c:3320]: handle_tcp_child(): reader response= 7fcfc20c57b0, -1 from 16 Nov 19 20:12:50 sjomainkama51 /usr/sbin/kamailio[21517]: DEBUG: tls [tls_server.c:683]: tls_h_close(): Closing SSL connection 0x7fcfc2089f88 Bottom line is kamailio closed the connection and I have method setting to this: root@sjomainkama51:/etc/kamailio # grep method tls.cfg method = TLSv1.1+ Supposedly this should work? -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

2 2

Fwd: Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Gentlemen, Thanks for your feedback. Yes, I've doubled the RAM last night (between my posts) and I am monitoring the behavior. It *feels* that it is definitely working better. I haven't enabled memjoin yet - are there any repercussions that should be expected for enabling memjoin=yes? Thanks for your guidance. Tom

3 2

Websocket immediately disconnecting
by Tom D 19 Nov '19

19 Nov '19

Hi David, Thanks for your reply. After increasing the verbosity of the debugging, I found the following entries. So am I understanding the log correctly: the websocket module is trying to get some free memory (from Kamilio's general pool?). It doesn't receive any yet proceeds with parsing the response (I get this same parsing error even when it's working) and ultimately, it doesn't register the websocket? I appreciate your (or anyone's) thoughts. Regards, Tom Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:737]: sr_ssl_ctx_info_callback(): SSL handshake started Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:749]: sr_ssl_ctx_info_callback(): SSL handshake done Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_domain.c:753]: sr_ssl_ctx_info_callback(): SSL disable renegotiation Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:431]: tls_accept(): tls_accept: new connection from HI.DD.E.N:63356 using TLSv1/SSLv3 AES256-GCM-SHA384 256 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:434]: tls_accept(): tls_accept: local socket: HI.DD.E.N:8089 Nov 19 02:44:44 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: tls [tls_server.c:445]: tls_accept(): tls_accept: client did not present a certificate Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: CRITICAL: <script>: 857:HTTP Request Received Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f8b8cdcd000, 65728); Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f8b8cdcd000, 65728) called from websocket: ws_conn.c: wsconn_add(197), module: websocket; Free fragment not found! Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_conn.c:199]: wsconn_add(): allocating shared memory Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line(): parse_first_line: bad message (offset: 22) Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TLS HI.DD.E.N:63356#015#012Access-Control-Allow-Origin: https://www.doubango.org#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: taS0s3Z7t4tOhklSbJDLGmSSYsY=#015#012Server: kamailio (5.2.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> Nov 19 02:44:45 MY-HOSTNAME /usr/sbin/kamailio[14928]: ERROR: websocket [ws_frame.c:583]: ws_frame_receive(): WebSocket connection not found

3 3

Asterisk text messages not delivered to SIP client
by Adesh Pandey 19 Nov '19

19 Nov '19

Hi Guys, I have been using Kamailio as dispatcher but I have been facing a weird situation that when asterisk SendText command says that message was delivered successfully but webrtc sip client did not receive the message. My deployment has Kamailio as registrar where sip users are registered and when they make any outgoing call, the call is proxied by RTPEngine and then dispatched to Asterisk boxes. Asterisk boxes send some text to the client using SendText but the message is not received by the SIP client. My config as follows: https://gist.github.com/adeshpandey/2c338db7f5d992267f7415de0325e0b4

1 0

Kamailio meeting Istanbul and Cairo
by Henning Westerholt 19 Nov '19

19 Nov '19

Hello, I will be in about 2 weeks in Istanbul and in Cairo a few days later. I am thinking of organizing a Kamailio/SIP/VoIP social networking event (perhaps a dinner or a pub meeting). Possible dates are the 19th- 20th November in Istanbul and the 23th - 24th November in Cairo. Are people from the community interested in joining such event? If yes, write me your preferred date -- based on feedback in the next days, it will be decided whether it is going to happen and when. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

2 2

Re: [SR-Users] call disconnected in 45 min
by Liu, Charles 18 Nov '19

18 Nov '19

Hi All, I have install kamailio IMS using the example config files for P/S/I-cscf with version 5.2... call randomly disconnected between 15 to 45 minutes. Wireshark log show scscf sent BYE to the caller and callee. Any idea what maybe the problem? Scscf debug output ......... ............ 20(25) INFO: ims_registrar_scscf [cxdx_avp.c:138]: cxdx_get_avp(): cxdx_get_experimental_result_code: Failed finding avp 20(25) INFO: ims_registrar_scscf [cxdx_sar.c:84]: create_return_code(): created AVP successfully : [saa_return_code] - [1] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1923]: print_all_dlgs(): ********************13(18) DEBUG: ims_dialog [dlg_handlers.c:1924]: print_all_dlgs(): printing 4096 dialogs 13(18) DEBUG: ims_dialog [dlg_handlers.c:1862]: internal_print_all_dlg(): Trying to get lock for printing 13(18) DEBUG: ims_dialog [dlg_handlers.c:1868]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1869]: internal_print_all_dlg(): Dialog h_entry:h_id = [3396 : 2798] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1870]: internal_print_all_dlg(): Dialog age: 2749 13(18) DEBUG: ims_dialog [dlg_handlers.c:1871]: internal_print_all_dlg(): Dialog call-id: ATokIPEjouNpuwSo6MpHUw.. 13(18) DEBUG: ims_dialog [dlg_handlers.c:1872]: internal_print_all_dlg(): Dialog state: 4 13(18) DEBUG: ims_dialog [dlg_handlers.c:1873]: internal_print_all_dlg(): Dialog ref counter: 2 13(18) DEBUG: ims_dialog [dlg_handlers.c:1874]: internal_print_all_dlg(): Dialog did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1875]: internal_print_all_dlg(): Dialog from_tag: 8975d611 13(18) DEBUG: ims_dialog [dlg_handlers.c:1876]: internal_print_all_dlg(): Dialog from_uri: sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1877]: internal_print_all_dlg(): Dialog caller contact: sip:bob@192.168.164.23:39180;transport=UDP 13(18) DEBUG: ims_dialog [dlg_handlers.c:1878]: internal_print_all_dlg(): Dialog first request cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1879]: internal_print_all_dlg(): Dialog caller route set: <sip:mo@10.1.1.1;lr=on;ftag=8975d611;rm=8> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1880]: internal_print_all_dlg(): Dialog lifetime: 43200 13(18) DEBUG: ims_dialog [dlg_handlers.c:1881]: internal_print_all_dlg(): Dialog bind_address: udp:172.26.1.60:6060 13(18) DEBUG: ims_dialog [dlg_handlers.c:1882]: internal_print_all_dlg(): Dialog-creating transaction pointer [0x7f8bb2a51858] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1888]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1889]: internal_print_all_dlg(): Dialog out h_entry:h_id = [3296 : 0] 13(18) DEBUG: ims_dialog [dlg_handlers.c:1890]: internal_print_all_dlg(): Dialog out did: 13(18) DEBUG: ims_dialog [dlg_handlers.c:1891]: internal_print_all_dlg(): Dialog out to_tag: c9eb5e55 13(18) DEBUG: ims_dialog [dlg_handlers.c:1892]: internal_print_all_dlg(): Dialog out caller cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1893]: internal_print_all_dlg(): Dialog out callee cseq: 1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1894]: internal_print_all_dlg(): Dialog out callee contact: sip:alice@192.168.164.84:49153;alias=192.168.164.84~49153~1 13(18) DEBUG: ims_dialog [dlg_handlers.c:1895]: internal_print_all_dlg(): Dialog out callee route set: <sip:mt@172.26.1.60:6060;lr;ftag=8975d611;did=44d.eea>,<sip:mt@10.1.1.1;lr;ftag=8975d611;rm=7> 13(18) DEBUG: ims_dialog [dlg_handlers.c:1896]: internal_print_all_dlg(): Dialog out branch: z9hG4bKb7d5.6f11b9b34a8a493a3a1b7949831e2f47.0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1897]: internal_print_all_dlg(): Dialog out state (deleted): 0 13(18) DEBUG: ims_dialog [dlg_handlers.c:1899]: internal_print_all_dlg(): ----------13(18) DEBUG: ims_dialog [dlg_handlers.c:1903]: internal_print_all_dlg(): Releasing lock for dlgout 13(18) DEBUG: ims_dialog [dlg_handlers.c:1906]: internal_print_all_dlg(): ----------------------------13(18) DEBUG: ims_dialog [dlg_handlers.c:1934]: print_all_dlgs(): ********************13(18) DEBUG: ims_auth [authorize.c:187]: reg_await_timer(): Looking for expired/useless at 44469727 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 121 <alice(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:alice@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44469993 0x7f8bb2a5c988 13(18) DEBUG: ims_auth [authorize.c:192]: reg_await_timer(): Slot 914 <bob(a)ims.mnc001.mcc001.3gppnetwork.org<mailto:bob@ims.mnc001.mcc001.3gppnetwork.org>> 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 3 Exp 44470022 0x7f8bb2a54e88 13(18) DEBUG: ims_auth [authorize.c:197]: reg_await_timer(): .. AV 1 - 2 Exp 44469974 0x7f8bb2a506a8 13(18) DEBUG: ims_auth [authorize.c:203]: reg_await_timer(): ... dropping av 1 - 2 13(18) DEBUG: ims_auth [authorize.c:232]: reg_await_timer(): [DONE] Looking for expired/useless at 44469727 13(18) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 13(18) DEBUG: ims_dialog [dlg_hash.c:1066]: unref_dlg(): unref dlg 0x7f8bb2a551f8 with 1 -> 2 13(18) DEBUG: ims_dialog [dlg_req_within.c:379]: dlg_terminate(): terminating confirmed dialog 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to caller 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 3 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to caller 13(18) DEBUG: ims_dialog [dlg_req_within.c:272]: send_bye(): sending BYE to callee 13(18) DEBUG: ims_dialog [dlg_hash.c:1047]: ref_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 4 13(18) DEBUG: ims_dialog [dlg_req_within.c:289]: send_bye(): BYE sent to callee 1(6) ERROR: <script>: BYE (sip:bob@ims.mnc001.mcc001.3gppnetwork.org;transport=UDP (172.26.1.60:6060) to sip:alice@ims.mnc001.mcc001.3gppnetwork.org, ATokIPEjouNpuwSo6MpHUw..) 1(6) DEBUG: ims_dialog [dlg_handlers.c:844]: dlg_onroute(): route param is '44d.eea' (len=7) 1(6) DEBUG: ims_dialog [dlg_hash.c:887]: lookup_dlg(): ref dlg 0x7f8bb2a551f8 with 1 -> 5 1(6) DEBUG: ims_dialog [dlg_hash.c:889]: lookup_dlg(): dialog id=2798 found on entry 3396 1(6) DEBUG: ims_dialog [dlg_hash.h:459]: match_dialog(): searching dlg_outs 1(6) DEBUG: ims_dialog [dlg_hash.h:496]: match_dialog(): p: 0x7f8bb2a68c48 ft[8975d611] tt [c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_hash.h:520]: match_dialog(): dout: tt[c9eb5e55] 1(6) DEBUG: ims_dialog [dlg_profile.c:419]: set_current_dialog(): setting current dialog [3396:2798] 1(6) DEBUG: ims_dialog [dlg_hash.c:1293]: next_state_dlg(): dialog 0x7f8bb2a551f8 changed from state 4 to state 6, due event 7 CONFIDENTIALITY NOTICE: This email and any attachments are for the sole use of the intended recipient and may contain material that is proprietary, confidential, privileged or otherwise legally protected or restricted under applicable government laws. Any review, disclosure, distributing or other use without expressed permission of the sender is strictly prohibited. If you are not the intended recipient, please contact the sender and delete all copies without reading, printing, or saving.

1 0

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I figured this finally out and all works for me very well. In usrloc dbmode i set value of 3 (its saving immediately changes into DB). In request route for REGISTER method i set t_on_reply("MANAGE_REPLY_REGISTER" and for REGISTER with expires set to 0 i set t_on_reply("MANAGE_REPLY_UNREGISTER") (used pseudovariable $expires(max)). I stored Contact header in $avp(contact) for unREGISTER request for further use when calling unregister(). Then in onreply_route for MANAGE_REPLY_REGISTER i call save lcoation and in MANAGE_REPLY_UNREGISTER for reply code 200 i call unregister function with the stored contact (i had to remove < > and display name using re.subst). So far it looks to work perfect. pon., 18 lis 2019 o 13:38 Voip support <voipexpert0(a)gmail.com> napisał(a): > > Thanks again for all replies... I need to try it out - did not have time > for it yet. > > I think i will go for the onreply_route answer and try to save location in > this place. > > pon., 18 lis 2019 o 06:23 Joel Serrano <joel(a)textplus.com> napisał(a): > >> Yes sir! Works perfectly! >> >> To give you a quick example: >> >> 1) Add in the appropriate place for the REGISTER request: >> t_on_reply("MANAGE_REG_REPLY"); >> 2) Create a new onreply route: >> >> # Manage registration incoming replies >> onreply_route[MANAGE_REG_REPLY] { >> if(status=~"2[0-9][0-9]") { >> if (save("location", "0x04")) { >> xlog("L_NOTICE", "[usrloc] saved user location\n"); >> } >> } >> } >> >> >> Note1: In my setup, I needed to strictly simulate asterisk (chan_sip) >> registrations, so only one AoR was allowed, thus the flags 0x04, if you are >> using PJSIP you might not need it. >> Note2: With this example code, a "unregister" (REGISTER with expires=0) >> will not remove the AoR from Kamailio, to achieve that I check >> independently if the header Expires exists and is 0, or if the contact >> param Expires exists and is 0, and in that case I use the unregister() >> function instead of the save(). >> >> Hope it helps! >> Joel. >> >> >> >> On Sun, Nov 17, 2019 at 5:42 PM Ovidiu Sas <osas(a)voipembedded.com> wrote: >> >>> Did it work calling save() on the reply route? >>> >>> -ovidiu >>> >>> On Thu, Nov 14, 2019 at 6:20 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> > >>> > I am using kamailio 5.3. >>> > >>> > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could >>> work. >>> > >>> > David Villasmil , i know i can use SQLOPS but i was thinking of easier >>> method of doing this. >>> > By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> > Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> > I need to be able to check what was the real source IP of client >>> REGISTER. >>> > >>> > Regards, >>> > Tomasz >>> > >>> > >>> > >>> > pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> >>> napisał(a): >>> >> >>> >> I am using kamailio 5.3. >>> >> >>> >> Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it >>> could work. >>> >> >>> >> David Villasmil , i know i can use SQLOPS but i was thinking of >>> easier method of doing this. >>> >> By successful REGISTER i mean save only REGISTER attempt that >>> authenticated correctly at SIP server and in 200 OK a binding was returned. >>> >> Same if the binding is later removed i would like remove it from >>> database. I need this because when i run REGISTER over kamailio to >>> destination SIP server i see all register from same IP. >>> >> I need to be able to check what was the real source IP of client >>> REGISTER. >>> >> >>> >> Regards, >>> >> Tomasz >>> >> >>> >> >>> >> >>> >> >>> >> czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> >>> napisał(a): >>> >>> >>> >>> Try to use save() when you process the 200 ok reply for REGISTER in >>> >>> the reply route. >>> >>> >>> >>> Regards, >>> >>> Ovidiu Sas >>> >>> >>> >>> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >>> wrote: >>> >>> > >>> >>> > I setup kamailio to forward REGISTER packets using dispatcher to >>> two servers main and backup. >>> >>> > >>> >>> > The routing logic is quite simple i use the dispatcher probing >>> mechanism to detect if my main dispatcher set id (one server) is active or >>> inactive once it detect a failure all packets are sent to secondary SIP >>> server to achieve High Availability. >>> >>> > >>> >>> > I am looking for module or a way to save into kamailio database >>> client information such as IP address, register username (from FROM >>> header), digest username (authorization username), call-id for successful >>> registration. >>> >>> > >>> >>> > I tried to use registrar module but once i used function >>> save("location") the client was registered on kamailio and packets was not >>> forwarded to main/secondary SIP server (like dispatcher stopped working, >>> the destination did not override). >>> >>> > >>> >>> > REGISTER is sent to SIP URI of kamailio. >>> >>> > >>> >>> > Please let me know if you have any idea how to make this data >>> update in database (insert an entry on fresh successfull REGISTER, delete >>> on expired or removed binding). >>> >>> > >>> >>> > Best regards, >>> >>> > Tomasz >>> >>> > >>> >>> > _______________________________________________ >>> >>> > Kamailio (SER) - Users Mailing List >>> >>> > sr-users(a)lists.kamailio.org >>> >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> >>> >>> >>> >>> -- >>> >>> VoIP Embedded, Inc. >>> >>> http://www.voipembedded.com >>> >>> >>> >>> _______________________________________________ >>> >>> Kamailio (SER) - Users Mailing List >>> >>> sr-users(a)lists.kamailio.org >>> >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> > >>> > _______________________________________________ >>> > Kamailio (SER) - Users Mailing List >>> > sr-users(a)lists.kamailio.org >>> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.com >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users(a)lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >>

1 0

Websocket immediately disconnecting
by Tom D 18 Nov '19

18 Nov '19

Hi Everyone, We're running Kamailio 5.2.5 with XHTTP, Websocket and TLS extensions - for the purpose of being able to provide WebRTC capability. The settings for the HTTP request are quite default as set out below. The SIP client we're using is SIP5ML in different web browsers. After a successful start of Kamailio, it is possible to use SIP5ML and its features - i.e. it REGISTER's properly, calls can be made, etc. After about a few hours of running, Kamailio refuses to interact and abruptly closes the TCP connection. The only solution (discovered so far) is to restart Kamailio, which is not feasible on a server that is serving production needs. (On our development server, where the load is only a fraction - this problem does not occur). When the problems does start to occur, in Wireshark I see: 1) Normal TLS negotiation and certificate exchange; followed by 2) SIP5ML initiating the initial HTTP GET 3) Kamailio responds to the upgrade with a 101/Switch Protocols. (4- SIP5ML outputs a SIP REGISTER on a defunct socket as its was closed) Immediately after the 101/Switching Protocol message is received, Kamailio closes down the TCP connection with a FIN message. Nothing in the message or Kamailio log files help to identify what module is responsible for this. Would anyone have any advise where to look further? BTW, I don't see the TCP connection being Reset in the exchanges - so any advise will be appreciated. Tom event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != 8089) { xlogl("L_CRIT", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden - 0x24", "", ""); exit; } xlogl("L_CRIT", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { # Validate Host - make sure the client is using the correct # alias for WebSockets if ($hdr(Host) == $null || !is_myself("sip:" + $hdr(Host))) { xlogl("L_CRIT", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden - 0x25", "", ""); exit; } # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information about the # successful connection exit; } } xhttp_reply("404", "Resource Not Found - 0x27", "", ""); } event_route[websocket:closed] { xlogl("L_CRIT", "WebSocket connection from $si:$sp has closed\n"); }

2 1

Kamailio stop responding to AWS R53 health check
by Andrew Chen 18 Nov '19

18 Nov '19

Hi all, R53 reported healthcheck failure on tcp port 5060. Looking at the system, all the ports that kamailio listens on can't be monitored remotely. To test this theory, I can telnet successfully to consul port. I ran tcpdump and see no responses returned back to R53 monitoring hosts: tcpdump: verbose output suppressed, use -v or -vv for full protocol decode listening on ens5, link-type EN10MB (Ethernet), capture size 262144 bytes 19:11:49.368378 IP ec2-54-241-32-75.us-west-1.compute.amazonaws.com.10072 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 1252984048, win 29200, options [mss 1460,sackOK,TS val 3770272726 ecr 0,nop,wscale 7], length 0 19:11:49.558877 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780607162 ecr 0,nop,wscale 7], length 0 19:11:50.572870 IP ec2-107-23-255-43.compute-1.amazonaws.com.60794 > sjoprodkama51.fuzemeeting.com.sip: Flags [S], seq 3484910496, win 29200, options [mss 1460,sackOK,TS val 1780608175 ecr 0,nop,wscale 7], length 0 Fixed the issue after restarting kamailio process. Given this system is support production traffic, we have minimized the logging on it. The events that we thought may lead to this is as follows: Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20310]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda69958 r: 0x7f33fda699d8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20309]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdd829e8 r: 0x7f33fdd82a68 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20311]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20312]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbd5d78 r: 0x7f33fdbd5df8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda47c98 r: 0x7f33fda47d18 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaec8f0 r: 0x7f33fdaec970 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: BUG: tls [tls_server.c:1303]: tls_read_f(): IO error (0) Success Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc7f490 r: 0x7f33fdc7f510 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20317]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20314]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdbb40b8 r: 0x7f33fdbb4138 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20313]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda01dc0 r: 0x7f33fda01e40 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd960380 r: 0x7f33fd960400 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:14209102:SSL routines:tls_early_post_process_client_hello:unsupported protocol Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdacac30 r: 0x7f33fdacacb0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20308]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdaa8f70 r: 0x7f33fdaa8ff0 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20307]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd93e6c0 r: 0x7f33fd93e740 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20305]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb70738 r: 0x7f33fdb707b8 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20315]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdb822d0 r: 0x7f33fdb82350 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20316]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fda25fd8 r: 0x7f33fda26058 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20306]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fdc196f8 r: 0x7f33fdc19778 (-1) Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1408F119:SSL routines:ssl3_get_record:decryption failed or bad record mac Nov 13 18:28:05 sjoprodkama51 /usr/sbin/kamailio[20304]: ERROR: <core> [core/tcp_read.c:1505]: tcp_read_req(): ERROR: tcp_read_req: error reading - c: 0x7f33fd9c27a8 r: 0x7f33fd9c2828 (-1) Any thoughts on what would cause this issue? Thanks -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

4 14

Re: [SR-Users] Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

> Oops. No, it’s not a bug, it’s just that the URI isn’t the only value present in $ct. > > Try $(ct{nameaddr.uri}{uri.host}) Yup, that fixed it. Thanks again! I have another question about how I’m applying these to some routing logic but I’ll start a new thread for that in a bit.

1 0

http_async_client and HTTP2
by Igor Olhovskiy 18 Nov '19

18 Nov '19

HI! Seems to be, http_async client is not compatible with http2 servers. I'm having like kamailio_1 | 15(20) INFO: http_async_client [http_multi.c:238]: debug_cb(): [cURL] Connection #0 to host relay-api.pbxservice.com left intact kamailio_1 | 15(20) ERROR: <core> [core/parser/parse_fline.c:264]: parse_first_line(): parse_first_line: bad message (offset: 11) kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR: parse_msg: message=<HTTP/2 404 kamailio_1 | date: Thu, 14 Nov 2019 18:31:34 GMT .... kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:337]: parse_headers(): bad header field [(null)] kamailio_1 | 15(20) ERROR: <core> [core/parser/msg_parser.c:1274]: get_body(): failed to parse to end of headers But plain http_client worked good (not async for sure) kamailio 5.3.0 -- Best regards, Igor

3 3

Problem with URI transformations
by Duarte Rocha 18 Nov '19

18 Nov '19

Greetings, I'm trying to parse some URI info on Kamailio but i'm having some troubles the URI transformations. Here i have two examples of URIS from which i'm trying to extract username(with user parameters included) and URI Parameters : - sip:123456789;test=param@1.1.1.1 - sip:123456789;test=param@1.1.1.1;user=phone In order to test this i made the following code : $var(URI) = "sip:123456789;test=param@1.1.1.1"; xerr("DEBUG 1 : URI $var(URI)"); $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 1 : User = $var(User)"); xerr("DEBUG 1 : Parameters = $var(Parameters)"); $var(URI) = "sip:123456789;test=param@1.1.1.1;user=phone"; xerr("DEBUG 2 : URI $var(URI)"); ; $var(User) = $(var(URI){uri.user}); $var(Parameters) = $(var(URI){uri.params}); xerr("DEBUG 2 : User = $var(User)"); xerr("DEBUG 2 : Parameters = $var(Parameters)"); The result is the following : DEBUG 1 : URI sip:123456789;test=param@1.1.1.1 DEBUG 1 : User = 123456789;test=param DEBUG 1 : Parameters = DEBUG 2 : URI sip:123456789;test=param@1.1.1.1;user=phone DEBUG 2 : User = 123456789 DEBUG 2 : Parameters = user=phone If the URI has parameters, the User parameters won't be included in the User part. Is this a bug or is it working as intended? I'm running Kamailio 5.2.0 . Thank you for your time. Best Regards, Duarte Rocha

3 6

Parsing Contact Port Field
by Michael Iedema 18 Nov '19

18 Nov '19

Hello everyone, I have a potentially silly beginners question: how can I parse the contact port field and act on it in my routing logic? I know that the $ct variable contains the entire contact header and I can print it in an xlog() call. However, I’d like to do something like the following in my routing logic: . . . SIP CONTENT Contact: <sip:someaccount@192.168.86.104:5062>;expires=1800 . . . CONFIG LOGIC #!define PORT_SERVICE1 5061 #!define PORT_SERVICE2 5062 route { if ( src_port == PORT_SERVICE1 ) { xlog(“L_INFO”, “Received $rm from SERVICE1\n”); } else if ( src_port == PORT_SERVICE2 ) { xlog(“L_INFO”, “Received $rm from SERVICE2\n”); } } I realize that src_port is not the correct value to compare against. I want to compare against the originating contact’s port value. How can I extract the port field from $ct? Many thanks in advance and apologies for the beginners question. I’ve googled for what I think I’m trying to do without any real results on functions or tokenizers, etc. I’m using Kamailio 5.3.0. Regards, -Michael

3 3

Re: [SR-Users] Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 18 Nov '19

18 Nov '19

I am using kamailio 5.3. Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could work. David Villasmil , i know i can use SQLOPS but i was thinking of easier method of doing this. By successful REGISTER i mean save only REGISTER attempt that authenticated correctly at SIP server and in 200 OK a binding was returned. Same if the binding is later removed i would like remove it from database. I need this because when i run REGISTER over kamailio to destination SIP server i see all register from same IP. I need to be able to check what was the real source IP of client REGISTER. Regards, Tomasz pt., 15 lis 2019 o 00:16 Mike from ApoloCore <mike(a)apolocore.com> napisał(a): > I am using kamailio 5.3. > > Ovidiu Sas, i will try this tomorrow today is bit late. Maybe it could > work. > > David Villasmil , i know i can use SQLOPS but i was thinking of easier > method of doing this. > By successful REGISTER i mean save only REGISTER attempt that > authenticated correctly at SIP server and in 200 OK a binding was returned. > Same if the binding is later removed i would like remove it from database. > I need this because when i run REGISTER over kamailio to destination SIP > server i see all register from same IP. > I need to be able to check what was the real source IP of client REGISTER. > > Regards, > Tomasz > > > czw., 14 lis 2019 o 23:33 Ovidiu Sas <osas(a)voipembedded.com> napisał(a): > >> Try to use save() when you process the 200 ok reply for REGISTER in >> the reply route. >> >> Regards, >> Ovidiu Sas >> >> On Thu, Nov 14, 2019 at 3:40 PM Voip support <voipexpert0(a)gmail.com> >> wrote: >> > >> > I setup kamailio to forward REGISTER packets using dispatcher to two >> servers main and backup. >> > >> > The routing logic is quite simple i use the dispatcher probing >> mechanism to detect if my main dispatcher set id (one server) is active or >> inactive once it detect a failure all packets are sent to secondary SIP >> server to achieve High Availability. >> > >> > I am looking for module or a way to save into kamailio database client >> information such as IP address, register username (from FROM header), >> digest username (authorization username), call-id for successful >> registration. >> > >> > I tried to use registrar module but once i used function >> save("location") the client was registered on kamailio and packets was not >> forwarded to main/secondary SIP server (like dispatcher stopped working, >> the destination did not override). >> > >> > REGISTER is sent to SIP URI of kamailio. >> > >> > Please let me know if you have any idea how to make this data update in >> database (insert an entry on fresh successfull REGISTER, delete on expired >> or removed binding). >> > >> > Best regards, >> > Tomasz >> > >> > _______________________________________________ >> > Kamailio (SER) - Users Mailing List >> > sr-users(a)lists.kamailio.org >> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >

3 2

Saving REGISTER status in kamailio database while REGSITER packets are forwarded.
by Voip support 15 Nov '19

15 Nov '19

I setup kamailio to forward REGISTER packets using dispatcher to two servers main and backup. The routing logic is quite simple i use the dispatcher probing mechanism to detect if my main dispatcher set id (one server) is active or inactive once it detect a failure all packets are sent to secondary SIP server to achieve High Availability. I am looking for module or a way to save into kamailio database client information such as IP address, register username (from FROM header), digest username (authorization username), call-id for successful registration. I tried to use registrar module but once i used function save("location") the client was registered on kamailio and packets was not forwarded to main/secondary SIP server (like dispatcher stopped working, the destination did not override). REGISTER is sent to SIP URI of kamailio. Please let me know if you have any idea how to make this data update in database (insert an entry on fresh successfull REGISTER, delete on expired or removed binding). Best regards, Tomasz

5 5

dispatcher, pvar_hash and multiple kamailio instances
by Barry Flanagan 15 Nov '19

15 Nov '19

Hi, I have multiple Kamailio instances using dispatcher using alg “7” - hash over the content of PVs string, both using the same dispatcher list. Is it reasonable to expect both Kamailio, given the same hash_pvar, to select the same host from the same set? Looking at the source I do not see any reason why not, but I am getting different results from the two Kamailio servers. The only difference between them is different minor version numbers - 4.3.4 and 4.3.7 I want to ensure that calls with the same hash_pvar value go to the same backend no matter which Kamailio processes the request. Thanks! -Barry Flanagan

3 2

Re: [SR-Users] Question on Database Version Table
by JR Richardson 15 Nov '19

15 Nov '19

Yep version_table Set the name of the table holding the table version. Useful if the proxy is sharing a database within a project and during upgrades. Default value is “version”. Example of usage: version_table="version44" That's what I was looking for, perfect! Yes, I added accountcode and notes to my table for internal use, was just making notes on columns I need on the new version upgrade. Thanks. JR On Tue, Jun 18, 2019 at 2:17 PM Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > > Hello, > > like Alex said, you can have two version tables, one for each versions > of kamailio you play with and use in config the core parameter to set > the name of that table. > > I am actually writing to say that address table never had accountcode > and notes columns. Likely they were added in your deployment for other > purposes or either the default column names were changed. The default > column names are listed at: > > - > https://kamailio.org/docs/db-tables/kamailio-db-5.2.x.html#gen-db-address > > These were the same for rather long time, iirc the recent version number > changed mainly due to with modifications done to the size, not due to > new columns. > > Cheers, > Daniel > > On 18.06.19 20:50, JR Richardson wrote: > > Hey All, > > > > I'm doing some upgrades and noticed database table version numbers > > have increased. > > > > database table adjustments: > > | address | production ver 3 | upgrade 5.2 ver 6 | need to add > > accountcode and notes column > > | dispatcher | production ver 3 | upgrade 5.2 ver 4 | need to add attr column > > | trusted | production ver 5 | upgrade 5.2 ver 6 | need to add > > ruri_pattern and priority column > > > > I run mysql master/slave so I'm concerned about the master database > > not having the correct version numbers or can I just update the > > version number in the table when I do the upgrade of the kamailio > > versions? > > > > Thanks. > > > > JR

4 6

Processes blocking - troubleshooting slow actions
by Julien Chavanton 15 Nov '19

15 Nov '19

I am troubleshooting intermittent latency on a server, several processes will report slow action at the same time every now and then, something odd it that some of the actions are most likely not using any mutexes, I can not imagine how they can end up blocking unless the process is actually suspended by the OS, however the CPU usage and load average is quite low on this server. Is there any reason why assigning to a PKG var could be delayed 3 seconds. 2019-11-13T21:58:04.018631614Z 23(42) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:115] took too long [2579486 us] $var(reply) = 0; // I wonder how this can endup blocking ? 2019-11-13T21:58:04.018748710Z 42(61) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [corefunc (16)] cfg [/etc/kamailio/kamailio.cfg:221] took too long [2222010 us] } // Samething here, simply exiting a scope 2019-11-13T21:58:04.015675095Z 11(30) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay_cancel (24)] cfg [/etc/kamailio/kamailio.cfg:86] took too long [2588818 us] } if (!t_relay_cancel()) { // Indeed a t_relay_cancel 2019-11-13T21:58:04.018647988Z 26(45) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [t_relay (24)] cfg [/etc/kamailio/kamailio.cfg:289] took too long [2739366 us] t_relay(); // Indeed a t_relay 2019-11-13T21:58:04.018651243Z 15(34) WARNING: <core> [core/action.c:1586]: run_actions(): alert - action [sl_send_reply (26)] cfg [/etc/kamailio/kamailio.cfg:112] took too long [2587812 us] sl_send_reply("100", "Trying"); // Indeed a sl_send_reply Not sure which mailing list is best for this question Regards Julien

2 4

kamailio 5.2.4 unable to load db_mysql.so
by Rupesh Kumar 15 Nov '19

15 Nov '19

I am trying to compile and load kamailio 5.2.4 on centos 7.5. I am getting below errors whenever I start kamailio . I have the latest openssl and openssl-devel packages installed on my centos . Not sure what other dependencies . Not starting Kamailio SIP Server: invalid configuration fil[FAILED] [FAILED]0(23235) ERROR: <core> [core/sr_module.c:498]: load_module(): could not open module </usr/local/lib64/kamailio/modules/db_mysql.so>: /usr/local/lib64/kamailio/modules/db_mysql.so: undefined symbol: SSL_get_error 0(23235) CRITICAL: <core> [core/cfg.y:3508]: yyerror_at(): parse error in config file //usr/local/etc/kamailio/kamailio.cfg, line 229, column 12-24: failed to load module ERROR: bad config file (1 errors) 0(23235) INFO: <core[FAILED]sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized [FAILED]

1 1

Next online devel meeting - irc/video conference - Thursday, Nov 14, 15:00UTC
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, on Thursday and Friday this week over a dozen of Kamailio developers will be in Dusseldorf for the annual coding meeting, so we thought of doing an online meeting via some video conferencing bridge, with IRC as a backup chatting channel, to allow also other community members to join remotely and discuss the development plans for short/medium term. It is more or less what we used to do during the past IRC devel meetings, but to make the discussions go faster and shorter the meeting, so people in Dusseldorf spend more time coding, this time we will try video conferencing. Very likely we will use Jitsi Meet, so anyone that wants to join should be prepared with a WebRTC capable web browser (all major ones support it) -- link and details how to join will be provided short before the meeting starts, right now the plan is to do it on Thursday, Nov 14, at 15:00UTC/London (16:00Berlin-Paris, 10:00New York). If someone wants to address questions but doesn't have the possibility to join the video conf bridge, then join the IRC #kamailio channel on freenode.net and post your question there during the meeting. The IRC channel will be monitored and used for text chatting whenever needed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

2 3

Kamailio v5.3.1 Released
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, Kamailio SIP Server v5.3.1 stable release is out. This is a maintenance release of the latest stable branch, 5.3, that includes fixes since the release of v5.3.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.3.x. Deployments running previous v5.3.x versions are strongly recommended to be upgraded to v5.3.1. For more details about version 5.3.1 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/11/kamailio-v5-3-1-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 0

Releasing Kamailio v5.3.1
by Daniel-Constantin Mierla 14 Nov '19

14 Nov '19

Hello, the plan right now is to release v5.3.1 during the next Kamailio Developers Meeting, if we succeed to coordinate on site, therefore expect it to happen next week during Nov 14-15. Besides just doing this specific release, to goal is to try to build a more automatic process for releases, document how is done, etc... updates about the results will follow after the event. Anyhow, for now the usual the pre-release notice: if anyone is aware of some issue that is not yet reported, do it asap in order to give it a chance to be analyzed and fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

1 1

event_route[core:msg-received] native vs kemi/lua
by GG GG 14 Nov '19

14 Nov '19

Hi! I just made a simple test performance between native script and kemi/lua with the event_route[core:msg-received] callback. # lua script children=1 cpu.process=100% function ksr_core_received() exit; end # native script children=1 cpu.process=70% event_route[core:msg-received] { exit; } # tested with udp 5060, 1000 Mbits/sec iperf -c 127.0.0.1 -p 5060 -u -b1000m I don't know if it's normal and what we should expect from kemi, but I would like to share this with you. Maybe someone could confirm the performance difference. Have a good day !

2 8

kemi framework design
by GG GG 12 Nov '19

12 Nov '19

Hi, looking at the documentation and the code, I couldn't answer myself. - does kamailio use one lua interpreter per fork ? - about the safety, I just made a bad lua module in c, to force a segmentation fault in the module. I would expect kamailio to protect itself from the crash, but kamailio restarted.

2 1

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

3 2

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Change SDP Session Name
by Alvaro Araos 11 Nov '19

11 Nov '19

Hi everyone, Exists some way to change the SDP Session Name value (s=)??? Saludos, Alvaro Araos

1 0

Kamailio in cloud environments
by Alejandro Recarey 11 Nov '19

11 Nov '19

Hi, I am trying to get kamailio working cloud provider, autoscaling them behind a TCP load balancer. Is there a way for kamailio to discover its Public IP at startup? Or do people generally program their own startup scripts that modify the kamailio config file before starting it? Is there any gotchas you have hit while configuring this setup? Thank you! Alex

8 13

uac_replace_from(uri) alters display name
by Sergiu Pojoga 10 Nov '19

10 Nov '19

Hi ppl, Need to replace origination carrier's domain with my own, so I do: branch_route[MANAGE_PSTN_IN_BRANCH] { uac_replace_from("sip:$fU@mydomain.net"); } The dialog-forming INVITE is relayed properly with ONLY the domain part being replaced. However, subsequent messages such as ACK and BYE have the From DISPLAY NAME (_uac_fdp) stripped off. Not that it affects real functionality in any known way (CLID is still good for callee), but curious if this is intentional or not, as *uac_replace_from(uri)* clearly states that: "*Replace in FROM header the URI part without altering the display name*" # kamcmd dlg.list variables: { { _uac_fdpnew: } { _uac_fdp: "JOHN DOE" } { _uac_funew: sip:18661112222@mydomain.net } { _uac_fu: sip:18661112222@carrierdomain.net } Thanks!

2 4

OPTIONS request not authenticating
by Gustavo A. M. 10 Nov '19

10 Nov '19

Greetings, I'm trying to configure kamailio to get OPTIONS response, but it does not working properly. After a successful REGISTER, I send OPTIONS first time, get Proxy- Authenticate header to calculate Proxy-Authorization, and resend OPTIONS with Proxy-Authenticate header, but server follows answering with a 407 response, apparently ignoring Proxy-Authenticate. Note that same password is used in REGISTER request, and first time REGISTER response with a 401, not a 407 response. I think messages sent from client are right, and problem is on the server configuration. Follow messages and kamailio.cfg. Thanks in advance, Gustavo Adolfo Moennich OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 0 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 0 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.467d Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905;received=192.168.122.12 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1wV3HNJXdfUDchNlEUkT6kHCtqBAn" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 OPTIONS sip:6000@kamailio.memorea.com SIP/2.0 Call-ID: A6DB9DCAA18049E784034FB6F5F95680 Contact: <sip:6000@kamailio.memorea.com> Content-Length: 0 CSeq: 1 OPTIONS Expires: 20 From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC Max-Forwards: 70 Proxy-Authorization: Digest uri=sip:6000@kamailio.memorea.com,nonce="Xc c1wV3HNJXdfUDchNlEUkT6kHCtqBAn",realm="kamailio.memorea.com",response=" 5bcd184eea53284f3ec548c2412ce36a",username=6000 To: "Bob" <sip:6000@kamailio.memorea.com> User-Agent: Memorea 1.0 (SIP 2.0) Build 1.1.1.0, Windows x64 Build 6.1.7601 Via: SIP/2.0/UDP 192.168.122.21:5060;rport=58905 SIP/2.0 407 Proxy Authentication Required Call-ID: A6DB9DCAA18049E784034FB6F5F95680 CSeq: 1 OPTIONS From: "Bob" <sip:6000@kamailio.memorea.com>;tag=0B3D2156FF9E46DAAD7FC43 B62C9E2FC To: "Bob" <sip:6000@kamailio.memorea.com>;tag=9dd61ff61e802d8e2bef5f146 21ef3c2.32de Via: SIP/2.0/UDP 192.168.122.21:5060;received=192.168.122.12;rport=58905 Proxy-Authenticate: Digest realm="kamailio.memorea.com", nonce="Xcc1xl3HNJqRobJFuZggPSGvNdtp+GnP" Server: kamailio (5.2.1 (x86_64/linux)) Content-Length: 0 kamailio.cfg content's: #!KAMAILIO #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB # # Kamailio (OpenSER) SIP Server v5.2 - default configuration script # - web: https://www.kamailio.org # - git: https://github.com/kamailio/kamailio # # Direct your questions about this file to: <sr-users(a)lists.kamailio.or g> # # Refer to the Core CookBook at https://www.kamailio.org/wiki/ # for an explanation of possible statements, functions and parameters. # # Note: the comments can be: # - lines starting with #, but not the pre-processor directives, # which start with #!, like #!define, #!ifdef, #!endif, #!else, #!trydef, # #!subst, #!substdef, ... # - lines starting with // # - blocks enclosed in between /* */ # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # - option for NAT SIP OPTIONS keepalives: WITH_NATSIPPING # # *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To block 401 and 407 authentication replies execute: # - define WITH_BLOCK401407 # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif ####### Include Local Config If Exists ######### import_file "kamailio-local.cfg" ####### Defined Values ######### # *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!ifndef DBURL #!define DBURL "mysql://kamailio:temp@database.memorea.com/kamailio" #!endif #!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif # - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 ####### Global Parameters ######### ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 log_prefix="{$mt $hdr(CSeq) $ci} " /* number of SIP routing processes */ 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:10.0.0.10:5060 #!ifdef WITH_TLS enable_tls=yes #!endif /* life time of TCP connection when there is no traffic * - a bit higher than registration expires to cope with UA behind NAT */ tcp_connection_lifetime=3605 ####### Custom Parameters ######### /* These parameters can be modified runtime via RPC interface * - see the documentation of 'cfg_rpc' module. * * Format: group.id = value 'desc' description * Access: $sel(cfg_get.group.id) or @cfg_get.group.id */ #!ifdef WITH_PSTN /* PSTN GW Routing * * - pstn.gw_ip: valid IP or hostname as string value, example: * pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" * * - by default is empty to avoid misrouting */ pstn.gw_ip = "" desc "PSTN GW Address" pstn.gw_port = "" desc "PSTN GW Port" #!endif #!ifdef WITH_VOICEMAIL /* VoiceMail Routing on offline, busy or no answer * * - by default Voicemail server IP is empty to avoid misrouting */ voicemail.srv_ip = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif ####### Modules Section ######## /* set paths to location of modules */ # mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" #!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "corex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "acc.so" loadmodule "counters.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif #!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif #!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif #!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif #!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif #!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif #!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif #!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "pretty_format", 1) /* set the path to RPC fifo control file */ # modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") /* set the path to RPC unix socket control file */ # modparam("jsonrpcs", "dgram_socket", "/var/run/kamailio/kamailio_rpc.sock") # ----- ctl params ----- /* set the path to RPC unix socket control file */ # modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) # ----- rr params ----- # set next param to 1 to add value to ;lr param (helps with some UAs) modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ # modparam("registrar", "max_contacts", 10) /* max value for expires of registrations */ modparam("registrar", "max_expires", 3600) /* set it to 1 to enable GRUU */ modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. * if you enable this parameter, be sure the enable "append_fromtag" * in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speeddial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) /* register callback to match myself condition with domains list */ modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4) # ----- htable params ----- /* ip ban htable with autoexpire after 5 minutes */ modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif #!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif #!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_level_name", "exec") #!endif ####### Routing Logic ######## /* Main SIP request routing logic * - processing of any incoming SIP request starts with this route * - note: this is the same as route { ... } */ request_route { # per request initial checks route(REQINIT); # NAT detection route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); } # Wrapper for relaying requests route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood detection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=$si) { if($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($ua =~ "friendly-scanner|sipcli|VaxSIPUserAgent") { # silent drop for scanners - uncomment next line if want to reply # sl_send_reply("200", "OK"); exit; } if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if (uri==$ru) { if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) { options_reply(); } } if(!sanity_check("17895", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessly route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == $ru) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction ... ignore and discard exit; } } sl_send_reply("404","Not here"); exit; } # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); #!ifdef WITH_NATSIPPING # do SIP NAT pinging setbflag(FLB_NATSIPPING); #!endif } if (!save("location")) { sl_reply_error(); } exit; } # User location service route[LOCATION] { #!ifdef WITH_SPEEDDIAL # search for short dialing - 2-digit extension if($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==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } if (is_method("OPTIONS") || from_uri==$fu) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("OPTIONS|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!=$fu && uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!else # authentication not enabled - do not relay at all to foreign networks if(uri!=$ru) { sl_send_reply("403","Not relaying"); exit; } #!endif return; } # Caller NAT detection route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) { set_contact_alias(); } } setflag(FLT_NATS); } #!endif return; } # RTPProxy control and signaling updates for NAT traversal route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if(nat_uac_test("8")) { rtpproxy_manage("co"); } else { rtpproxy_manage("cor"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } #!endif return; } # URI update for dialog requests route[DLGURI] { #!ifdef WITH_NAT if(!isdsturiset()) { handle_ruri_alias(); } #!endif return; } # Routing to foreign domains route[SIPOUT] { if (uri==$ru) return; append_hf("P-hint: outbound\r\n"); route(RELAY); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")) return; # only local users allowed to call if(from_uri!=$fu) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + if (starts_with("$rU", "00")) { strip(2); prefix("+"); } if (strempty($sel(cfg_get.pstn.gw_port))) { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); } else { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } # XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif # Routing to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; } # Manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # Manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") { route(NATMANAGE); } } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) exit; #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_BLOCK401407 # block call redirect based on 401, 407 replies. if (t_check_status("401|407")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { $du = $null; route(TOVOICEMAIL); exit; } #!endif }

2 1

dispatch REGISTER over TLS
by sthustfo 10 Nov '19

10 Nov '19

I have a basic setup where kamailio receives SIP over websocket (no WSS) and forwards to SIP server over TLS. I have enabled TLS in kamailio.cfg and added dispatcher node as sips:SIP_SERVER:5061 and transport=tls. +----+-------+------------------------+-------+----------+---------------+----------------+ | id | setid | destination | flags | priority | attrs | description | +----+-------+------------------------+-------+----------+---------------+----------------+ | 4 | 1 | sips:10.0.0.100:5061 | 0 | 0 | transport=tls | SIP SERVER | +----+-------+------------------------+-------+----------+---------------+----------------+ Now when REGISTER is received over websocket, kamailio is responding with error code 500 and phrase "500 I'm terribly sorry, server error occurred (7/SL)". And on the console I see the following error messages. 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/md5utils.c:67]: MD5StringArray(): MD5 calculated: f1ecf7bcb659b07fe81e332e100044e5 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [ut.h:315]: uri2dst2(): no corresponding socket found for "10.0.0.100" af 2 (tls: 10.0.0.100:5061) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:467]: prepare_new_uac(): can't fwd to af 2, proto 3 (no corresponding listening socket) 12(33858) ERROR: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -7 (-7) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} tm [t_funcs.c:352]: t_relay_to(): -7 error reply generation delayed 12(33858) exec: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=587 a=24 n=sl_reply_error 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} <core> [core/msg_translator.c:162]: check_via_address(): (10.0.0.14, hsvmphm3ps12.invalid, 0) 12(33858) DEBUG: {1 2521 REGISTER o9q7ujqgin33qgp00abijv} websocket [ws_conn.c:452]: wsconn_get(): wsconn_get for id [3] *tls.cfg contents* [client:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /home/test/kamailio/internal.key certificate = /home/test/kamailio/internal.crt ca_list = /home/test/kamailio/ca_list.pem Any reason why this error is seen? Any inputs appreciated. Thanks.

4 20

Re: [SR-Users] _w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I added $du=$null; before route(RELAY) as you suggested but I get the same results: Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: transaction canceled Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 8 12:40:22 spock /usr/sbin/kamailio[21826]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) Regards, Konstantinos > On Fri Nov 8 10:24:06 CET 2019 Daniel Tryba wrote: > >> On Thu, Nov 07, 2019 at 04:53:43PM +0200, Konstantinos Merentitis wrote: >> >> I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: >> > You appear to do the same thing I do, except setting $du to null. If the user is REGISTERed $du will be set to their endpoint, changing $ru will have no effect on routing. So add $du=$null; in the TOVOICEMAIL route before route(RELAY) and see what happens. >

3 4

_w_t_relay_to(): t_forward_noack failed when forwarding call
by Konstantinos Merentitis 08 Nov '19

08 Nov '19

I would like to forward all incoming calls that are not answered from kamailio, to my asterisk voicemail server. - I have not setup any Realtime Integration. I am using the "Redirected Dialed Number Information Service" header and calling "554(a)192.168.1.134" which is my voicemail server. Forwarding succeeds when the subscriber is not registered to kamailio: Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============User sip:6297@192.168.1.131 not found here, going to voicemail============= Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:45 spock /usr/sbin/kamailio[17606]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== I tried to modify the failure route in order to also forward not answered (failed?) calls to asterisk mailbox but i get the following error and the call is never forwarded: Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: transaction canceled Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: #012 RDNIS = sip:297@192.168.1.134 Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: <script>: =============forwarding to voicemail1, RDNIS: sip:297@192.168.1.134 =========== Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: tm [tm.c:1679]: _w_t_relay_to(): t_forward_noack failed Nov 7 16:19:25 spock /usr/sbin/kamailio[17609]: ERROR: sl [sl_funcs.c:392]: sl_reply_error(): stateless error reply used: transaction canceled (487/SL) failure and voicemail routes are slightly modified from the default config: 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")) { xlog("=============User $ru not found here, going to voicemail=============\n"); $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; } 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")) { $var(RDNIS)=$di; xlog("\n RDNIS = $var(RDNIS) \n"); if($avp(oexten)==$null) return; # $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail1, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } else { if($rU==$null) return; # $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip + ";transport=udp") # + ":" + $sel(cfg_get.voicemail.srv_port); xlog("=============forwarding to voicemail2, RDNIS: $di =========== "); $ru = "sip:554@192.168.1.134:5060"; } route(RELAY); exit; #!endif return; } # Manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { #modified here xlog("transaction canceled\n"); route(TOVOICEMAIL); #exit; #/modified } #!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; xlog("============= Route failure, forwarding to voicemail ============="); route(TOVOICEMAIL); exit; } #!endif }

2 1

How to read some data inside the INVITE ..
by Laura 07 Nov '19

07 Nov '19

Dears.. how is possibile to read that data 9052562462354464 inside the first INVITE message === Example === U 2019/11/07 14:46:59.320936 192.168.1.1:5060 -> 192.168.1.242:5060 INVITE sip:9052562462354464@192.168.1.242 SIP/2.0. Record-Route: <sip:192.168.1.1;lr;ftag=105854496;did=24e.e751;vst=AAAAAAAAAwQAAAAAAAAAAAAAAAAADQIAAwEDAAAHDwAGBAg1MDYw;nat=yes>. Max-Forwards: 66. Via: SIP/2.0/UDP 51.254.158.40:5060;branch=z9hG4bK501c.87a8e402c8f1cf93192ddd8e6744da57.1. From: "38617774738" <sip:38617774738@10.1.1.1>;tag=105854496. To: <sip:562462354464@192.168.1.242>. Call-ID: 1573138018-1602381967(a)10.1.1.182. CSeq: 20215490 INVITE ..... ETC ETC... Thanks to all

4 4

Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

3 5

multihomed kamailio and sip-reinvites SRTP to RTP
by Karsten Horsmann 07 Nov '19

07 Nov '19

Hello Mailinglist, i try to figure out, how to solve RE-INVITES and SRTP: i have an kamailio route names route[MEDIAPROXY] that i used to controll rtpengine in an private and public ip setup. This works fine for me now. It use the permission module and the adress table to figure out $var(mediagroup) = allow_address_group($Ri, 0); received from internal or external ip. The idea is from SBC-OS chamless stolen, so thanks for that. So fare so good. I adapted SRTP also to this. But on RE-INVITES it stops handling SRTP correctly to the encryption upstream. And i am sure, its my fault. But i have no clue how to detect that correctly. Any hints where i can read more about an correct handling on this? Here my main-config file / kamailio 5.3.x and rtpengine. https://gist.github.com/khorsmann/02bd98a628a190caee68fc3d876a3448 -- Mit freundlichen Grüßen *Karsten Horsmann*

2 2

Re: [SR-Users] Group module
by Daniel Tryba 07 Nov '19

07 Nov '19

On Wed, Nov 06, 2019 at 07:15:42PM +0100, Igor Olhovskiy wrote: > > Hm... Maybe there is other module to achieve such functions? > Best if it would be with cache :) > But if no - regex also fine. But if it works it works! I don't think there is a specific module to do this but it is easy to implement yourself. What I don't like personally about your solution is the use of the permission tables, you need atleast 1 per user (unless users have the same ip). Personally I do this by direct database queries in usr_preferences on INVITEs and REGISTERs. You could use any database (depending on how you store the ACL) and cache it in a htable. My solution is mysql specific (inet_aton) and forces users to have atleast 1 CIDR notated subnet: if(!avp_db_query("select value from usr_preferences where username='$au' and attribute='acl' and inet_aton(substring_index(value,'/',1))&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)=inet_aton('$si')&(1 << 32) - 1 & ~((1 << (32 - substring_index(value,'/',-1))) - 1)")) { t_reply("404","Go away"); exit; }

3 2

MongoDB findone(). Is projection supported?
by Lars Olsson 06 Nov '19

06 Nov '19

Hi, I am using NDB_MONGODB to ask some questions to a database. My database structure is pretty heavy, so I like to return as little data as required to Kamailio. I have tried the following MongoDB syntax for projection to include only given (my_field) column in the response. mongodb_find_one("srv_name", "database", "collection", "{ \"user_id\" : \"$var(user_id)\" }, { \"my_field\" : 1}", "mgr1") I always get full DB entry back. Is this supported? The same string works OK in the vanilla mongo client. How can I limit the result? Thanks Lars

2 3

Group module
by Igor Olhovskiy 06 Nov '19

06 Nov '19

Hi! I'm trying to solve quite simple task and want to solve it as simple as possible. Idea is to have restriction to allow register certain username/domain only from certain subnets. So, I decided to use group and permissions modules. But stuck on group one. Code is fairly simple modparam("group", "db_url", DBURL) modparam("group", "use_domain", 1) ... if (get_user_group("From", "$avp(user_groups)")) { xlog("$fu from $si:$sp is matched group $avp(user_groups)\n"); if (allow_source_address("$avp(user_groups)")) { ...allow furter } } else { xlog("$fu from $si:$sp is not matched any groups\n"); } grp table is looks like kamctl db exec 'select * from grp' +----+-------------+-------------------+-----+---------------------+ | id | username | domain | grp | last_modified | +----+-------------+-------------------+-----+---------------------+ | 1 | 61388140395 | sip.localhost.net | 5 | 2000-01-01 00:00:01 | +----+-------------+-------------------+-----+---------------------+ But actual line of debug in logs saying this kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=720 a=26 n=get_user_group kamailio_1 | 10(18) DEBUG: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} group [re_group.c:188]: get_user_group(): getting groups for <sip:61388140395@sip.localhost.net> kamailio_1 | 10(18) exec: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} *** cfgtrace:request_route=[PERMISSIONS] c=[/etc/kamailio/kamailio.cfg] l=727 a=25 n=xlog kamailio_1 | 10(18) ERROR: {1 1 REGISTER 4oQTNaaVlEWQC2VMK6YAKg..} <script>: sip:61388140395@sip.localhost.net;transport=UDP from 172.21.0.1:56787 is not matched any groups What am I missing here? Or module is too old? Thanks!

1 0

RFC: default config with jsonrpc over http instead of xmlrpc
by Daniel-Constantin Mierla 06 Nov '19

06 Nov '19

Hello, I think it was discussed during past IRC meetings or at some events across the world, with jsonrpc being these days more popular than xmlrpc, specially due to easier integration with web based apps, I am considering to replace the example of using xmlrpc over http(s) with jsonrpc in the default configuration file. It is not enabled by default for xmlrpc (requiring to define WITH_XMLRPC), so it will stay also as a define option for jsonrpc. The jsonrpcs module is already loaded for kamctl, it will require the xhttp module to be added. As for those looking for xmlrpc in the future, there is another example kamailio cfg which has the option with it: misc/examples/pkg/kamailio-oob.cfg Any other opinions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - April 27-29, 2020, in Berlin -- www.kamailioworld.com

4 3

$jsonrpl(body) empty or malfunctioned during higher traffic
by Michal Škuta 05 Nov '19

05 Nov '19

Hello, Kamailio community, we are implementing evapi communication to our Kamailio server. We have implemented and tested it on our testing server. Everything was working OK. Now we are testing it on our production servers with a lot more calls and we have some problems here. Every n seconds we are requesting from Kamailio list of all dialogs. And this list of dialogs (from jsonrpc *dlg.list*) is sent to evapi connection. Here is a snippet of our *Kamailio code*: route[CGR_DLG_LIST] { xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Requested dialog list"); # CGRateS connection is still alive, set check to true $sht(cgrconn=>check) = 1; if $sht(cgrconn=>cgr) == $null { sl_send_reply("503","Charging controller unreachable"); exit; } jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); xlog("L_INFO", "XLOG: $ci [CGR_DLG_LIST] Test1: $jsonrpl(body)"); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); } During a higher number of calls Kamailio is generating ERROR messages from evapi module. Here is the syslog: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: INFO: <script>: Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [ Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_dispatch.c:764]: _evapi_relay(): cannot serialize event Oct 30 12:02:15 kam-prod /opt/ipgate/kamailio/sbin/kamailio[19512]: ERROR: evapi [evapi_mod.c:265]: w_evapi_relay(): failed to relay event: *We are expecting something like this when nobody is calling* Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Requested dialog list Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: INFO: <script>: XLOG: 123 [CGR_DLG_LIST] Test1: { #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 } Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:740]: _evapi_relay(): relaying event data [{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }}] (96) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:778]: _evapi_relay(): sending [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) Oct 30 14:12:22 kam-test /opt/ipgate/kamailio/sbin/kamailio[32090]: DEBUG: evapi [evapi_dispatch.c:789]: _evapi_relay(): dispatching [0x169ff674] [96:{"event":"CGR_DLG_LIST_REPLY", #011#011#011"jsonrpl_body":{ #011"jsonrpc":#011"2.0", #011"result":#011[], #011"id":#0111 }},] (100) We enabled debug messages for module evapi and are trying to understand where is the problem. To make this work to this state we needed to increase the *pv_buffer* to really big size (2097152). It looks like $jsonrpl(body) is not ready and everything concated with it is not working. Any help or suggestion is welcomed. Sincerely Michal

3 6

Module acc - How to get sip_reason/code on log_write_cdr() messages
by David VILLAUME 05 Nov '19

05 Nov '19

Hello, I'm trying to get the reason in my cdrs in log_write_cdr() messages while I see It in acc_log_request() messages. Is there a way to retrieve it in log_write_cdr() messages ? Bellow is my config : modparam("acc", "log_level", 1) modparam("acc", "log_facility", "LOG_LOCAL3") modparam("acc", "cdr_log_enable", 1) modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 2) modparam("acc", "log_missed_flag", 3) modparam("acc", "report_cancels", 1) modparam("acc", "cdr_on_failed", 1) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_facility", "LOG_LOCAL3") modparam("acc", "cdr_expired_dlg_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdr_extra", "call_id=$ci;src_ip=$si;fU=$fU;")" David VILLAUME

2 2

Re: [SR-Users] driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 04 Nov '19

04 Nov '19

Hello Henning, is there an planed release date for this 5.3.x version? Unfortunately without this change i cant deploy it without changing all my mysql user credentials to be sure there is no "kamailio DBURL separator" within the password field. Cheers Karsten [...] -- Mit freundlichen Grüßen *Karsten Horsmann*

3 2

kamailio - failover setup
by Youssef Boujraf 04 Nov '19

04 Nov '19

Dear All, I have seen in the new kamailio release 5.3.0, kamailio support for haproxy protocol – both the v1 (human-readable) and v2 (binary) versions of the protocol are supported ( https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ ) but it seems to be for Database, webserver failover not for kamailio itself. Haproxy is ONLY supporting TCP but not UDP. My question is : I am looking for failover service of kamailio servers looks like haproxy but udp & tcp. - one domain name : sip.secure.com pointing my public ip address. - two kamailio server pointing same database content (user, whitelist, ...) no asterisk or pbx. Kamailio is used Only for signaling (users, whitelist & rtp) Monitoring service : identify who is the kamailio server online, offline and the ability to force all the SIP/RTP/MANAGEMENT connection from Kamailio Srv1 to Kamailio Srv2 and vice-versa ? Thanks for your suggestions. -- Youssef

4 4

How to use sdp_remove_media() ?
by Denys Pozniak 04 Nov '19

04 Nov '19

Hello! I'm trying to remove m=video stream from incoming SIP/SDP request in th way below: modparam("rtpengine", "read_sdp_pv", "$avp(read_sdp_pv)") modparam("rtpengine", "write_sdp_pv", "$avp(write_sdp_pv)") ... sdp_remove_media("video"); msg_apply_changes(); sdp_get("$avp(read_sdp_pv)"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| rb=/$rb/ \n"); xlog("L_WARN","[$cfg(route)] *m=video removed* ci| avp(read_sdp_pv)=/$avp(read_sdp_pv)/ \n"); rtpengine_manage("$avp(rtpengine_offer)"); set_body("$avp(write_sdp_pv)", "application/sdp"); But no luck, xlog shows m=video in $rb and $avp(rtp_sdp_read) as well. -- BR, Denys Pozniak

3 8

Phonenum module
by Igor Olhovskiy 01 Nov '19

01 Nov '19

Hi! I'm playing around libphonenum to get validation accross e.164 numbers. So, idea is simple if (phonenum_match($fU, "src_num")) { if ($phn(src_num=>valid) != 1) { send_reply("400", "Invalid e.164 format"); xlog("L_ALERT","ALERT: FU: $fU is not e.164 number $phn(src_num=>error)\n"); exit; } } But on every (actually valid e.164 number like 380442459966 or 61388141397) I got Parsing number failed in error result. Tried both phonenum_match($fU, "src_num") and phonenum_match("$fU", "src_num") Actually phonenum_match("380442459966", "src_num") also not working Debian 10 Buster version: kamailio 5.2.5 (x86_64/linux) libphonenumber7 is already the newest version (7.1.0-5+b4).

2 6

kamailio on kubernetes
by Gaurav Bmotra 01 Nov '19

01 Nov '19

hello everyone i m working on a project kamailio on kubernetes for high availability and auto scalability but as i m able to run it in side the cluster with using kubenetes feature but having issue to access the service form outside the cluster i m using the ingress in kubernetse to point the service its flow like that domain.com:80 ------ingress-controler--------kamailio-service:80:5060------- POD(kamailio)5060 ISSUE IS I M NOT GETING CONNECTION BITWEEN DOMAIN.COM TO POD(kamailio) is there any one ho can help me in this thank you -- *Regards:* Gaurav Kumar

2 1

Routing Blocks - Missing in latest core wiki
by Joel Serrano 31 Oct '19

31 Oct '19

Hi, I was looking at the documentation and noticed that "Routing Blocks" section is missing in latest releases... http://www.kamailio.org/wiki/cookbooks/5.2.x/core#routing_blocks is OK But: http://www.kamailio.org/wiki/cookbooks/5.3.x/core#routing_blocks http://www.kamailio.org/wiki/cookbooks/devel/core#routing_blocks Don't exist, have they been moved elsewhere? Thanks! Joel.

3 8

Command Line SIP client
by Gaurav Bmotra 31 Oct '19

31 Oct '19

hi i m looking for Command Line SIP client , where i can register user form CLI for testing thank you -- *Regards:* Gaurav Kumar

9 11

How to intercept an INVITE after waking up (TSILO module) to change SDP
by Oleg Podguyko 31 Oct '19

31 Oct '19

Hello everybody! Kamailio works as a Registrar. Additionally, rtpengine is deployed next to it. Using the module rtpgngine I change SDP. Webrtc <-> SIP calls work fine. RTPengine proxies media. It is necessary to understand that when the call comes, I determine what Protocol the call came wss/tcp/udp, and then through the command "rtpengine_manage" change SDP accordingly. Now I have a new task. Wake up mobile customers through push notification. This client may be as WebRTC and also regular SIP. Trying to use TSILO module. # User location service route[LOCATION] { lookup("location"); $var(rc) = $rc; if (is_method("INVITE")) { if (t_newtran()) { route(SEND_PUSH); xlog("L_INFO", "tsilo store\n"); ts_store(); } } switch ($var(rc)) { case 1: xlog("L_INFO", "LOCATION, Subscriber found, RC: $var(rc)\n"); route(RELAY); break; case -1: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); exit; break; case -3: xlog("L_INFO", "LOCATION, Subscriber doesn't found, RC: $var(rc)\n"); send_reply("404", "Not Found"); exit; case -2: xlog("L_INFO", "LOCATION, Method Not Allowed, RC: $var(rc)\n"); send_reply("405", "Method Not Allowed"); exit; } } If a call occurs when the called client is registered, then we go to the route(RELAY), which creates a branch in which I change the SDP. Everything works. route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE")) { if(!t_is_set("branch_route")) t_on_branch("MANAGE_BRANCH"); } if (!t_relay()) { sl_reply_error(); } exit; } And already in MANAGE BRANCH the necessary SDP is put. If the called party is not registered. In the beginning everything goes well. I find this caller through a separate server. Send push notification. The client successfully registers on kamailio. # Handle SIP registrations route[REGISTRAR] { if (!is_method("REGISTER")) return; if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); setbflag(FLB_NATSIPPING); } if(isflagset(FLT_SRC_WSS)) { setbflag(FLB_DST_WS); } if (!save("location")) { sl_reply_error(); } ts_append("location", "$tu") exit; } Next I better show a piece of the log: окт 31 14:55:08 /usr/sbin/kamailio[17488]: exec: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} *** cfgtrace:request_route=[REGISTRAR] c=[/etc/kamailio/kamailio.cfg] l=429 a=26 n=ts_append окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tsilo [ts_append.c:72]: ts_append(): transaction 8800:338205132 found for 1234567890, going to append branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_lookup.c:1629]: t_lookup_ident_filter(): transaction found окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:255]: lookup_helper(): contact for [1234567890] found by address окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} registrar [lookup.c:456]: lookup_helper(): instance is окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:77]: t_append_branches(): transaction 8800:338205132 in status 0 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:99]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (1) outgoing branches окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:126]: t_append_branches(): Current uri sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/msg_translator.c:162]: check_via_address(): (50.7.93.27, uas7i396ceb5.invalid, 0) окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:149]: t_append_branches(): added branch [sip:1234567890@10.10.10.1:58027;transport=TCP;rinstance=727a64a3a59c508f] with ruid [uloc-5dbacb09-4450-7] окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} tm [t_append_branches.c:163]: t_append_branches(): Call uefn7rvufi112jj5lt91: 0 (0) outgoing branches after clear_branches() окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2238]: tcpconn_send_put(): send from reader (17488 (21)), reusing fd окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2473]: tcpconn_do_send(): sending... окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2507]: tcpconn_do_send(): after real write: c= 0x7fb39dde4c68 n=3131 fd=12 окт 31 14:55:08 /usr/sbin/kamailio[17488]: DEBUG: {1 2 REGISTER cY6rPubTvJ3ZbyxOrrvGHw..} <core> [core/tcp_main.c:2508]: tcpconn_do_send(): buf= Thus, the transaction wakes up and the initial INVITE is sent to the new registered client without any SDP change! Which eventually leads to 415 error. Now the question itself. Tell me! As me at this moment intercept this INVITE, to it has passed through in advance prepared path for changing SDP. Thank you. -- Oleg Podguyko

1 0

kamailio - failover setup
by Ilie Soltanici 31 Oct '19

31 Oct '19

We are using keepalived with virtual IP Address for Kamailio, dispatcher module for Asterisk, 2 MySQL Databases master-master replicated and 2 Servers with ProxySQL. So far so good.

2 3

[abalashov@evaristesys.com: binrpc reply sizes]
by Alex Balashov 31 Oct '19

31 Oct '19

Hi, What is the relationship between the `ctl` modparams `binrpc_buffer_size`, `binrpc_max_body_size`, and `binrpc_struct_max_body_size`? More especially, how interdependent are they? Is there a set of circumstances in which one would wish for the structure buffer to be smaller than the reply body buffer? What about the internal buffer? Is it used only for some kind of intermediate formatting, or does it also hold the entire reply in some aspect? The reasons for asking are predictable: RPC command 'dlg.list' throws out "reply too big" on some occasions, and rather than simply increasing all of these drastically, I would like to learn more about which ones it makes sense to increase, and in what proportion with respect to each other. I also think this idea should be more clearly developed in the documentation. 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 4

setbflag and isbflagset usage.
by Володимир Іванець 30 Oct '19

30 Oct '19

Hello! I'm using branch flags but not sure if the behavior is correct. There is a logic in branch route that decides if branch flag must be set and does it with this command: *setbflag(FLB_FLAG, $T_branch_idx)*. Then I'm checking if the flag is set with *if* statement later in the configuration file for requests and in onreply route for replies: *if (isbflagset(FLB_FLAG, $T_branch_idx))*. The thing is that *if* statement always returns true. The only way I can fix this behavior is to use *resetbflag* at the beginning of branch route: *resetbflag(FLB_SIPTOPUB, $T_branch_idx)*. Thank you! Regards, Volodya Ivanets.

1 0

strange conditional "if ($au!=$tU)" with previous exit
by PICCORO McKAY Lenz 30 Oct '19

30 Oct '19

i have this in asterisk integration how to, and i noted the "exit" before the "if($au!=$tU)" .. i dont understan the conditional and the exit there! please can someon xplain me that!? # authenticate the REGISTER requests (uncomment to enable auth) #!ifdef WITH_ASTERISK if (!www_authorize("$td", "sipusers")) #!else if (!www_authorize("$td", "subscriber")) #!endif { www_challenge("$td", "0"); exit; } if ($au!=$tU) { sl_send_reply("403","Forbidden auth ID"); exit; } i investigate at the kamailio cgf documentation and there's no clear topic related! http://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#tu_-_to_uri Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 3

Re: [SR-Users] [sr-dev] Kamailio as SBC for Microsoft Teams
by Henning Westerholt 30 Oct '19

30 Oct '19

Hello Rhys, I already replied to you yesterday, maybe it got into Spam on your side. :-) Quote below: > thanks. About the presence integration question - according to my research currently the only possibility would be to use the > existing Skype APIs and then use this (per interop-mode) also in MS Teams. > > Microsoft announced the availability of a API to set the presence status also for Teams, this is now scheduled for Q1 2020. There > are other limitations which also needs to be solved from MS and they are also scheduled for Q1 2020. > Feel free to contact me directly if you are interested in this topics. Cheers, Henning Am 29.10.19 um 08:51 schrieb Rhys Hanrahan: Hi Henning, This is a great article, thanks for sharing. I'm wondering if you've experimented with the possibility of integration of presence/dialog information with teams as well? Pretty much every integration guide I've seen on MS's UC solutions, including the previous ones (Lync, Communicator, SFB ) always seem limited to integrating calling via trunks. To me, the integration is not complete unless you can integrate presence things like: * If a user is on the phone then their status should be "busy" on Teams. When off call, set back to Available. * If a user is on a call in teams, our PBX should consider their PBX extension as "in use". Not sure it's possible, because I haven't seen anything of the sort being done yet, but I'm curious if this is something you looked into at all? Thanks. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 10 307 Pitt St, Sydney NSW 2000 [http://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg] ________________________________ From: sr-users <sr-users-bounces(a)lists.kamailio.org><mailto:sr-users-bounces@lists.kamailio.org> on behalf of Henning Westerholt <hw(a)skalatan.de><mailto:hw@skalatan.de> Sent: Tuesday, 29 October 2019 3:21 AM To: Kamailio (SER) - Users Mailing List Cc: Kamailio (SER) - Development Mailing List Subject: [SR-Users] Kamailio as SBC for Microsoft Teams Hello, I have created a “How-To” blog post on using Kamailio as “session border controller” for Microsoft Teams Direct Routing: https://skalatan.de/en/blog/kamailio-sbc-teams<https://skalatan.de/de/blog/kamailio-sbc-teams> You can this way use all the existing possibilities that Kamailio provide to interact with MS Teams as well. Best regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services _______________________________________________ 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 -- Kamailio Merchandising - https://skalatan.de/merchandising/ Kamailio services - https://skalatan.de/services Henning Westerholt - https://skalatan.de/blog/

2 2

Fwd: [CFP] FOSDEM 2020, RTC devroom, speakers, volunteers neeeded
by fosdem-rtc-admin＠freertc.org 28 Oct '19

28 Oct '19

FOSDEM - Real Time Communications devroom CfP ============================================= Overview -------- [FOSDEM](https://fosdem.org) is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2020 takes place 1-2 February 2020 in Brussels, Belgium. This document contains information about: - Real-Time Communications developer room (devroom) and lounge - speaking opportunities - volunteering in the devroom and lounge - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities) - the Planet aggregation sites for RTC blogs **NEW:** Save yourself entering Free-RTC events and CFP deadlines into your calendar and task list, follow our iCalendar feed: https://freertc.org/events.ics Call for participation - Real Time Communications (RTC) ------------------------------------------------------- The Real-Time devroom and Real-Time lounge are about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. **We are looking for speakers for the devroom and volunteers and participants for the tables in the Real-Time lounge.** The devroom is only on Sunday, 2nd of February 2020. The lounge will be present for both days. To discuss the devroom and lounge, please join the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). ### Speaking opportunities Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications devroom: deadline 23:59 UTC on 15th of December. Please use the [Pentabarf](https://penta.fosdem.org/submission/FOSDEM20/) system to submit a talk proposal for the devroom. On the "General" tab, please look for the "Track" option and choose "Real Time Communications devroom". Other devrooms and lightning talks: some speakers may find their topic is in the scope of more than one devroom. It is encouraged to apply to more than one devroom and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. Here you can find the [full list of devrooms](https://www.fosdem.org/2020/schedule/tracks/) and here you can apply for a [lightning talk](https://fosdem.org/submit). ### First-time speaking? FOSDEM devrooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the devroom administrators personally if you would like to ask any questions about it. ### Submission guidelines The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - The purpose of your talk - Any other talk applications (devrooms, lightning talks, main track) - Availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and devroom themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the devroom administrators based on the number of received proposals. As the two previous devrooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ----------------- To make the devroom and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - Organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 1 February - Participation in the Real-Time lounge - Circulating this Call for Participation to other mailing lists Social events and dinners ------------------------- The traditional FOSDEM beer night occurs on Friday, 31st of January. On Saturday night, there are usually dinners associated with each of the devrooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss) for further details about the Saturday night dinner options and how you can register for a seat. Related events around FOSDEM ---------------------------- As per usual, the [XMPP Summit](https://wiki.xmpp.org/web/Conferences/Summit_24) is happening ahead of FOSDEM. This time it will take place on the 30th and 31st of January also in Brussels. All details are available on the [summit website](https://wiki.xmpp.org/web/Conferences/Summit_24). Spread the word and discuss --------------------------- If you know of any mailing lists where this CfP would be relevant, please forward this document. If this devroom excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: - All projects https://planet.freertc.org planet(a)freertc.org - XMPP https://planet.jabber.org ralphm(a)ik.nu - SIP https://planet.sip5060.net planet(a)sip5060.net - (Español) https://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ------- For any private queries, contact us directly using the address **fosdem-rtc-admin(a)freertc.org** and for any other queries please ask on the [Free-RTC mailing list](http://lists.freertc.org/mailman/listinfo/discuss). The devroom administration team: - Saúl Ibarra Corretgé <s(a)saghul.net> - Ralph Meijer <ralphm(a)ik.nu> - Daniel-Constantin Mierla <miconda(a)gmail.com> - Daniel Pocock <daniel(a)pocock.pro> - Guus der Kinderen <guus.der.kinderen(a)gmail.com>

1 0

Rewrite contact address on 200 OK?
by Oded Arbel 28 Oct '19

28 Oct '19

Hi, we are having a problem with a network change use case. It goes on something like this: 1. Kamailio receives INVITE from caller. 2. Kamailio sends INVITE to callee MUA. 3. MUA response with 100, then 183. 4. MUA then loses network connectivity and re-establishes access with another IP address. 5. MUA sends 200 OK from the new IP address, but with `Contact` still set as the old address (which I think it should, because you aren't allowed to change contact in the middle of dialog?) 6. Kamailio forwards 200 OK to caller and receives ACK. 7. Kamailio sends ACK to old IP address of MUA. The last step is obviously a problem as the MUA will not be able to see the ACK. I'm trying to see how to resolve this issue. I can capture the "$si" and "$sp" of the new address in the reply route (I also tried to get the MUA to send OPTIONS immediately after reestablishing connectivity, and capture "$si"/"$sp" there, which also works), but I can't figure out how to update the destination of the ACK. I've tried running `fix_nated_contact()` as well as trying to write into various pseudo variables - in either the OPTIONS request handler or the 200 reply handler - which either errors or does nothing. Any suggestions? -- Oded Arbel

2 4

Kamailio to capture mirrored traffic
by Igor Olhovskiy 28 Oct '19

28 Oct '19

Hi! I'm trying to get Kamailio working as a traffic capture on a same machine with other PBX software installed. Actually, traffic is mirrored with iptables -A PREROUTING -t mangle -i eth0 -p udp --dport 5060 -j TEE --gateway 127.0.0.2 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) iptables -t nat -A PREROUTING -d 127.0.0.2 -p udp --dport 5060 -j DNAT --to 127.0.0.1:5062 (https://link.getmailspring.com/link/AB5F9D36-533D-4A52-ADE3-FB76B813163C@ge…) Kamailio request route is super simple request_route { xlog("L_ALERT", "[SIP-PACKET] Got packet [F=$fu R=$ru D=$du M=$rm IP=($si:$sp $Ri:$Rp) ID=$ci]\n"); drop; } I was trying to get Kamailio just listen on interface 127.0.0.1:5062, but no luck listen=udp:127.0.0.1:5062 Next was to use sipcapture module with following parameters loadmodule "sipcapture.so" modparam("sipcapture", "db_url", "text:///tmp/") modparam("sipcapture", "raw_socket_listen", "127.0.0.1:5060-5062") modparam("sipcapture", "raw_interface", "lo") modparam("sipcapture", "promiscious_on", 1) Also no luck. Means Kamailio can't see packets, but I see em with wireshark on lo interface. What is best way to get it working? Or I'm missing something? Thanks!

5 7

Cancel / OK handling
by Oz Mortimer 26 Oct '19

26 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call/force BYE if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

No audio after hold and unhold event
by arish haque 25 Oct '19

25 Oct '19

Hi all, No Audio flows between the two parties when one side resumes the call after putting on hold. Configuration - WebRTC <--> Kamailio <--> Asterisk Below is the warning I'm getting in asterisk console WARNING[32388]: chan_sip.c:10425 process_sdp: Declining non-primary audio stream: audio 10596 UDP/TLS/RTP/SAVPF 107 103 104 9 0 8 106 105 13 110 112 113 101 Is this a codec issue or something else. Thanks, Arish

1 0

Kamailio 3.3.7
by Brandon Armstead 24 Oct '19

24 Oct '19

Has anyone seen this error while compiling 3.3.7 raw_sock.c:431: undefined reference to `udpv4_vhdr_sum' system: Linux 4.9.0-11-amd64 #1 SMP Debian 4.9.189-3 (2019-09-02) x86_64 GNU/Linux lsmod: Module Size Used by bonding 147456 0 intel_rapl 20480 0 x86_pkg_temp_thermal 16384 0 intel_powerclamp 16384 0 coretemp 16384 0 kvm_intel 200704 0 kvm 598016 1 kvm_intel irqbypass 16384 1 kvm ttm 98304 0 drm_kms_helper 155648 0 crct10dif_pclmul 16384 0 crc32_pclmul 16384 0 joydev 20480 0 evdev 24576 3 ghash_clmulni_intel 16384 0 drm 360448 2 ttm,drm_kms_helper i2c_algo_bit 16384 0 iTCO_wdt 16384 0 shpchp 36864 0 ipmi_si 57344 0 iTCO_vendor_support 16384 1 iTCO_wdt sg 32768 0 pcspkr 16384 0 mei_me 36864 0 ipmi_msghandler 49152 1 ipmi_si mei 102400 1 mei_me video 40960 0 button 16384 0 acpi_power_meter 20480 0 acpi_pad 24576 0 ib_iser 49152 0 rdma_cm 53248 1 ib_iser iw_cm 49152 1 rdma_cm ib_cm 45056 1 rdma_cm ib_core 208896 4 ib_iser,ib_cm,rdma_cm,iw_cm configfs 40960 2 rdma_cm iscsi_tcp 20480 0 libiscsi_tcp 24576 1 iscsi_tcp libiscsi 53248 3 ib_iser,libiscsi_tcp,iscsi_tcp scsi_transport_iscsi 102400 4 ib_iser,libiscsi,iscsi_tcp ip_tables 24576 0 x_tables 36864 1 ip_tables autofs4 40960 2 ext4 589824 2 crc16 16384 1 ext4 jbd2 106496 1 ext4 crc32c_generic 16384 0 fscrypto 28672 1 ext4 ecb 16384 0 mbcache 16384 3 ext4 sd_mod 49152 4 hid_generic 16384 0 usbhid 53248 0 hid 122880 2 hid_generic,usbhid crc32c_intel 24576 2 aesni_intel 167936 0 aes_x86_64 20480 1 aesni_intel glue_helper 16384 1 aesni_intel lrw 16384 1 aesni_intel gf128mul 16384 1 lrw ablk_helper 16384 1 aesni_intel cryptd 24576 3 ablk_helper,ghash_clmulni_intel,aesni_intel ahci 40960 3 xhci_pci 16384 0 libahci 32768 1 ahci ixgbe 294912 0 xhci_hcd 188416 1 xhci_pci i2c_i801 24576 0 i2c_smbus 16384 1 i2c_i801 dca 16384 1 ixgbe ptp 20480 1 ixgbe libata 249856 2 ahci,libahci pps_core 16384 1 ptp mdio 16384 1 ixgbe usbcore 253952 3 usbhid,xhci_pci,xhci_hcd scsi_mod 225280 7 ib_iser,sd_mod,libiscsi,libata,scsi_transport_iscsi,iscsi_tcp,sg usb_common 16384 1 usbcore fan 16384 0 thermal 20480 0 I believe I've crossed this issue once before and it was a simple apt-get package to include compile options: make include_modules="mi_fifo db_mysql tm dialog sl tmx kex rr maxfwd textops textopsx nathelper acc avpops xlog uac alias_db siputils utils usrloc registrar pv regex htable sqlops sdpops topoh ctl db_mysql mysql" prefix="/usr/local" Thanks - Brandon

3 3

environmental variables in kamailio
by Gaurav Bmotra 24 Oct '19

24 Oct '19

hi i using kamailio 5.2 is it possible to use environmental variables value in kamailio.cfg like i want to use local ip and public ip in config file listen=udp:$(local_ip):5060 advertise $(public_ip):5060 if iti is possible then how i can use this thank you -- *Regards:* Gaurav Kumar

4 3

driver error: Can't connect to MySQL server with kamailio 5.3.x and mysql-config-groups
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hi List, i updated one of my testing kamailios from 5.2.x to latest 5.3.x. After restarting the process, i got an mysql error that the server "kamailio" is not found. I use the config-feature of my.cnf config-group names within the kamailio.cfg I can start my kamailio 5.3.x installation with "normal username:password@host" definitions for DBURL. Did anyone have the same issue? 0(19625) ERROR: db_mysql [km_my_con.c:139]: db_mysql_new_connection(): driver error: Can't connect to MySQL server on 'kamailio' (4) 0(19625) ERROR: <core> [db.c:319]: db_do_init2(): could not add connection to the pool 0(19625) ERROR: dialplan [dp_db.c:179]: dp_connect_db(): unable to connect to the database 0(19625) ERROR: dialplan [dialplan.c:191]: mod_init(): could not initialize data 0(19625) ERROR: <core> [core/sr_module.c:850]: init_mod(): Error while initializing module dialplan (/usr/lib64/kamailio/modules/dialplan.so) ERROR: error while initializing modules from my kamailio.cfg i defined the DBURL like this: # Use /etc/my.cnf.d/kamailio.cnf to store credentials # within a mysql-db-group #!define DBURL "mysql://[kamailio]/kamailio" and special modules can use it like this (or via DBURL) modparam("sqlops", "sqlcon", "cb=>mysql://[kamailio]/kamailio") cat /etc/my.cnf.d/kamailio.cnf # MySQL configuration group for kamailio [kamailio] user=kamailio password="kamailiorw" host=172.20.100.22 rpm -qa | grep kam kamailio-dialplan-5.3.0-13.1.x86_64 kamailio-websocket-5.3.0-13.1.x86_64 kamailio-jansson-5.3.0-13.1.x86_64 kamailio-mysql-5.3.0-13.1.x86_64 kamailio-tls-5.3.0-13.1.x86_64 kamailio-ims-5.3.0-13.1.x86_64 kamailio-statsd-5.3.0-13.1.x86_64 kamailio-5.3.0-13.1.x86_64 kamailio-redis-5.3.0-13.1.x86_64 cat /etc/redhat-release CentOS Linux release 7.6.1810 (Core) -- Cheers *Karsten Horsmann*

3 7

MS Teams Direct routing with kamailio
by Karsten Horsmann 23 Oct '19

23 Oct '19

Hello List, i search for the right place in kamailio to manipulate the dispatcher OPTIONS packets to add an Contact header. For uac generated traffic i can use the event_route[tm:local-request] from the tm module. But how can i manipulate kamailio *self* generated OPTION pings from the dispatcher? OPTIONS request for MS Teams must contain the Contact header of such a format: Contact: <sip:SBC_FQDN:5061;transport=tls> I follow this guide (for opensips) and try to adapt that step by step to kamailio: https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ CentOS 7 / kamailio 5.2.x (and maybe soon 5.3.x) / rtpengine -- Cheers *Karsten Horsmann*

2 2

Unable to use BLF
by Cehes (Free) 23 Oct '19

23 Oct '19

Hi all, This is many days we are trying many things and reading many threads :) We also re installed a simple brand new kamailio to be sure everything is ok. In fact, with sngrep we can see "subscribe" arriving on kamailio but on syslog we have a critical error : Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) exec: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} *** cfgtrace:request_route=[PRESENCE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=770 a=24 n=handle_subscribe Oct 23 10:08:42 GTO-Temp01 kamailio[2004]: 20(2027) CRITICAL: {1 29 SUBSCRIBE 3135373137353930373733323337-joash9r9lnre} <core> [core/mem/q_malloc.c:494]: qm_free(): *BUG: bad pointer 0xa (out of memory block!) *called from presence: *subscribe.c*: handle_subscribe(1040) - ignoring We tried to start kamailio with many more memory (x4) and this is the same. We tried with only "presenceXXX" plugins, and also with "puaXXX" plugins below is our config file : https://gist.github.com/Dyriolon/e894b9f2a67f9dfb7117421717ad6178 Any idea of what is wrong ? Thanks!

1 0

Call Pickup and Call Parking
by Zhan Bazarov 22 Oct '19

22 Oct '19

Hello, What is the Best way to implement call parking and call pickup in setup Kamailio(active-active) and dispatcher with a few asterisk round-robin? Could someone share some best practices, please? Thank you!

1 0

mysql create user and grant privileges
by Daniel-Constantin Mierla 22 Oct '19

22 Oct '19

Hello, the mysql server version 8 is no longer supporting creating the user and granting privileges in the same statement, they have to be separate. I checked the mariadb specs and they seem to support also the separate creation of user and granting privileges in two statements. I pushed the changes in kamdbctl to master branch via commit: - https://github.com/kamailio/kamailio/commit/c0879add55bef77703f2259f224b4a1… The reason I am writing here is to see if anyone is aware of some (old) versions of mariadb or mysql servers that do not support the creation and grant in two steps. If yes, then we can add a config option in kamctlrc to do one or the other variants. But for now I did it only with the new style, as it seems to be supported everywhere by latest versions, to avoid unnecessary option to keep also the old style. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

1 0

[Dialog] Removing entries from dialog memory
by Julien Klingenmeyer 21 Oct '19

21 Oct '19

Hi, I am wondering if there is any way to remove an entry from the active dialog list? I mean, assuming that Kamailio did not get any BYE related to a call for some reason and that I want that call to be “killed” from dialog memory before reaching the lifetime timeout. What I already tried with no success: sending a request to Kamailio catched by a dedicated route of the script then: * using the dlg_get function (with the target call as parameter) then send_bye function (then dropping the BYE request generated by the dialog module in the onsend_route with the drop function): no success. Dialog module correctly generates BYE requests but dialog related to the call is kept in memory until its lifetime timeout (and BYE requests are not dropped in the onsend_route either). * using the dlg_get function then dlg_set_timeout to set a very low timeout value for that particular call. No successs: Trying to update a bogus dlg as critical error. Could some RPC commands or whatever help me to accomplish what I try to do? Thanks Julien

4 9

pv_buffer_size limit
by Sergiu Pojoga 20 Oct '19

20 Oct '19

Hi ppl, I'm testing this external app (cgrates, you all know it) that talks to Kamailio via evapi. Every *n* minutes the app sends an evapi request to Kamailio which returns back the list of all dialogs like so: jsonrpc_exec('{"jsonrpc":"2.0","id":1, "method":"dlg.list","params":[]}'); evapi_relay("{\"event\":\"CGR_DLG_LIST_REPLY\", \"jsonrpl_body\":$jsonrpl(body)}"); The problem is that the size of *$jsonrpl(body)* PV is limited by Kamailio's *pv_buffer_size* setting. My math shows that a single dialog is about 1KiB of json data. *Problem*: Kamailio fails to start with a *pv_buffer_size *higher than 117 KiB (119808 bytes), which in turn rends the system's limit to about 117 simultaneous calls... pretty sad. Below find the error when trying to start Kamailio with higher values: 0(17080) ERROR: <core> [core/mem/q_malloc.c:291]: qm_find_free(): qm_find_free(0x7f02275c5010, 120832); Free fragment not found! 0(17080) ERROR: <core> [core/mem/q_malloc.c:425]: qm_malloc(): qm_malloc(0x7f02275c5010, 120832) called from core: core/pvapi.c: pv_init_buffer(2069), module: core; Free fragment not found! 0(17080) ERROR: <core> [core/pvapi.c:2072]: pv_init_buffer(): cannot init PV print buffer slot[39] Is there a workaround for this? Thanks, --Sergiu

2 1

Get value from header and check value type
by Karsten Horsmann 18 Oct '19

18 Oct '19

Hi List, I use Kamailio 5.2.x and I want to fetch an value from an X Header and do some checks like "is not empty" and "should be an integer". AFAIK per default all values are strings with the hdr function. What are the best way for some validations within Kamailio script language? if(is_present_hf("X-someheader")) { $avp(routing_target) = $(hdr(X-someheader){s.int}); if(!ds_list_exists("$avp(routing_target)")) Cheers Karsten Horsmann

2 2

How to get the dispatcher value dynamically
by Narayan P 18 Oct '19

18 Oct '19

Hello, We are using kamailio for load balancing as stateless proxy as mentioned below. {$du = "sip:192.168.130.XXX;transport=udp"; forward();} We have set $du to the correct media server value in the dispatcher module.Can you please tell me how to get the correct dispatcher value(Media server IP) and print this into log file. I am using $ru and $du to print,but it is not giving the correct one. for example: Suppose for the first call ; $du = "sip:192.168.130.XXX:5060;transport=udp"; $du = "sip:192.168.130.XXX:5062;transport=udp"; and so on. How to fetch the correct dispatcher value dynamically. Any response will be highly appreciated. Thanks, Rudra

3 2

need different Contact with the UAC modul.
by "Hans-Jürgen Brand" 18 Oct '19

18 Oct '19

3 4

Cancel / OK handling
by Oz Mortimer 18 Oct '19

18 Oct '19

Hi, I have an issue where a client is sending cancel after the call has been answered but before the client received the 200OK, I know they should then respond with a BYE based on the 200 final result, but they don’t (Meaning the call / billing stays up).. Is there a way I can timeout a call if ACK is not received to the 200 OK? I know technically this is the clients issue but they can’t / won’t fix Thanks is advance Oz.

1 0

Changing $fU or $fn in branch_route
by Jurijs Ivolga 18 Oct '19

18 Oct '19

Hi! I have small problem. When I assign $fU or $fn in request-route and then one more time I make new assignment in branch_route. Instead of rewriting it is just concatenate needed value at the end. lets assume $avp(cli)=1234567 and $avp(cliplus)=+1234567. So if in request_route I do $fU=$avp(cli); and then in branch_route $fU=$avp(cliplus);. $fU becomes one string: "1234567+1234567". Is it a bug or I missing something? Here is config snippet: request_route { ... route(MY); ... } route[MY] { ... $avp(cli) = "1234567" $avp(cliplus) = "+"+$avp(cli); $fU=$avp(cli); # here $fU=1234567 $fn=$avp(cli); # here $fn=1234567 route(LCR_ROUTE); exit; } route[LCR_ROUTE] { if(!is_method("INVITE")) return; if (!load_gws("1",$rU,$avp(cli))) { send_reply("503", "Error loading gateways"); exit; } $var(i)=0; while($(avp(lcr_gw_uri)[$var(i)])!= $null){ xlog("L_INFO", "loaded gw_uri_avp[$var(i)]=$(avp(lcr_gw_uri)[$var(i)]) \n"); $var(i) = $var(i)+1; } if (!next_gw()) { send_reply("503", "No available gateways"); exit; } xlog("L_INFO", "request-uri $ru \n"); xlog("L_INFO", "$avp(lcr_gw_uri), $avp(lcr_id)\n"); xlog("L_INFO", "flag $avp(lcr_flag) \n"); t_set_fr(0, 4000); t_on_failure("RTF_LCR_ROUTE"); t_on_branch("BRANCH_CUST"); route(RELAY); exit; } branch_route[BRANCH_CUST] { if ( $avp(lcr_flag) == 4) { $fU=$avp(cliplus); #here $fU=1234567+1234567 $fn=$avp(cliplus); #here $fn=1234567+1234567 } } Jurijs

2 5

do MOD_<module name> keywords exist?
by Juha Heinanen 18 Oct '19

18 Oct '19

I have used in config file these kind of constructs: #!ifdef MOD_<module name> ... #!endif Is the MOD_xxx keyword automatically set when module xxx is loaded? Perhaps I have invented the whole concept in a dream? -- Juha

2 4

Failed AKAv1-MD5 authentication with Boghe
by Bao, Yin 18 Oct '19

18 Oct '19

I have a Boghe (v2.0.153.836) working fine with Kamailio with MD5 algorithm however fails to authenticate if I configure S-CSCF to use AKAv1-MD5. We have the same configuration on private user identity in FHoSS on both the succ MD5 run and failed AKAv1-MD5 run. I wonder if anyone has faced this issue before. Here are some details on message exchanges: * S-CSCF included the challenge in 401 Unauthorized – Challenging the UE: WWW-Authenticate: Digest realm="example.example.org", nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=", algorithm=AKAv1-MD5, ck="629d6d9a6a6befa509b1a9bb17a9c2a3", ik="32fcd14a102279de2e38200a2257efcb", qop="auth,auth-int"\r\n * The same header was seen when 401 Unauth travelled through I-CSCF and P-CSCF (shouldn’t P-CSCF strips out the ck and ik fields before sending to UE?) * UE responded with another register with response: Authorization: Digest username="bob(a)example.example.org",realm="example.example.org",nonce="rzGnTBGPw3hE+mDPHrZ1PAAAAAAApAAAmfftHqYUuUA=",uri="sip:example.example.org",response="6445f9df2b785eab3fa461849880b48d",algorithm=AKAv1-MD5,cnonce="34baf441e99a23e0fcb2bc001355c4bf",qop=auth-int,nc=00000001\r\n * S-CSCF failed the calculation and sent back 403 Authentication Failed I tried AKAv2-MD5 setting as default algorithm on S-CSCF and the result is the same as above. It seems that only MD5 works on my setup. What are the different configs in HSS (or other places) AKAv1-MD5 need? I checked the AMF/OP on both HSS and Boghe and they match. Boghe does not have SQN setting but I set HSS to start with 0. Any pointers are appreciated! Yin

2 1

IMC through HTTP
by Péter Barabás 18 Oct '19

18 Oct '19

Hi, I want to implement conference messaging functionality into a client. IMC module seems perfect. In our scenario, the rooms and members are managed in a different server bacause of other user management functionalities and the kamailio only uses the rooms to broadcast messages to members. Is it possible anyway to create rooms and join members from a separate server like through http protocol with XHTTP module? The idea is: 1. Create a room on server A with adding members. Server A store the room data. 2. Server A sends http message to kamailio with proper format, like POST .../createRoom with jsonBody { name: „room1” } 3. Kamailio receives the http request and process it with creating a room ’room1’. 4. Server A sends members of room like POST .../addMembers with json body { room: „room1”, members: [ username: „user1”, username: „user2” ] } 5. Kamailio receives the https request and process it with joining memers into room All commands should have the belonging http request. Is it possible? Regards, Peter

2 2

MySQL vs MariaDB support
by Sergey Safarov 18 Oct '19

18 Oct '19

At present time we create RPM packages using MySQL libs. May be need switch to MariaDB libs. What you think about it?

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio.We try to find any high-level flag to disable it. but could not able to find. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

2 1

Need to suppress "200 cancelling" in kamailio
by amit Kumar 18 Oct '19

18 Oct '19

Hi All, As per our requirement, when A party send a cancel to B party we do not want "200 cancelling" message from Kamailio. We try to find any high-level flag to disable it. but could not able to find it. Can you please help us on how to achieve this. A Party Kamailio B Party ---------------->INVITE------------------------> <---------180 Ringing<------------------------ ------------>CANCEL--------------------------> <---------200 canceling [we want to suppress this message and do not want kamialio to inform A party about progress status of cancel] <------------OK----------------------------------- Thanks, Amit

1 0

Kamailio v5.3.0 Released - new major version is out
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Kamailio v5.3.0 is out – it comes with 6 new modules and a considerable set of improvements touching more than 100 existing modules. You can read detailed release notes at: * https://www.kamailio.org/w/kamailio-v5-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. v5.3.0 brings more flexibility and optimizations for KEMI interpreters, enhancements to load balancer, dialog tracking, uac remote registration and tls with libssl 1.1.x implementations, new variables, transformations and plenty of other new features. Enjoy Kamailio v5.3.0! Thank you for flying Kamailio! Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda

2 1

Releasing Kamailio v5.3.0
by Daniel-Constantin Mierla 17 Oct '19

17 Oct '19

Hello, if there are no known major issues that need longer time to take care of them, in my opinion we could consider releasing the next major version 5.3.0 next week, for me Thursday, October 10 seems to work the best, still allowing a bit more than one week for testing. Any other suggestions? Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 4

catching tls errors
by Juha Heinanen 17 Oct '19

17 Oct '19

Sometimes I see in syslog errors like this: Oct 15 16:44:57 salmon /usr/bin/sip-proxy[2064]: ERROR: tls [tls_util.h:42]: tls_err_ret(): TLS accept:error:1417C086:SSL routines:tls_process_client_certificate:certificate verify failed Is it possible somehow to catch the error in config file, for example, to figure out from which IP address the connection attempt came from? -- Juha

3 6

Kamailio weighted dispatch problems
by Jasen Hall 17 Oct '19

17 Oct '19

I am trying to change our request dispatching from round-robin to a weight-based method (ds_select_dst() algorithm 9 vs. algorithm 4) and am having a hard time getting an asymmetrical dispatch pattern enabled. In testing, I'm dispatching to two Ringswitch servers and trying to send three times the traffic to one instance. My destinations list file previously set both endpoints in the same group and simply provided the SIP URI for each. My new list file adds the attribute `weight` to each endpoint and looks like this: >1 sip:10.0.0.1:5060 weight=3 >1 sip:10.0.0.2:5060 weight=1 However, when I throw a couple thousand calls at Kamailio, I always end up with an almost perfect split of calls to both ringswitches. Regardless of the weights I choose, I never see variation in the dispatched load. Does anyone have experience with this issue and have some guidance? Thanks in advance. -- Jasen Hall | Invoca *Site Reliability Engineer* www.invoca.com [image: See the list] <https://invoca.sigstr.net/uc/5cba111d3527a200469b9afd>

3 4

IMS Kamailio Diameter Charging Server
by Jack R 14 Oct '19

14 Oct '19

Hi All, We have installed Kamailio as IMS setup pcscf icscf scscf - 3 instances running on Ubuntu.. Referring to the link below, it mentioned about Diameter based Kamailio Charging Server. https://www.kamailio.org/events/2016-KamailioWorld/Day2/15-Carsten.Bock-Dia… Where to find it? How to install it? to make it work in Kamailio IMS set up as per below articles: https://www.kamailio.org/docs/modules/stable/modules/ims_charging.html https://kamailio.org/docs/modules/devel/modules/ims_ocs.html Regards, Jack.

2 2

Kamailio v5.2.5 Released
by Daniel-Constantin Mierla 10 Oct '19

10 Oct '19

Hello, Kamailio SIP Server v5.2.5 stable release is out. This is a maintenance release of the latest stable branch, 5.2, that includes fixes since the release of v5.2.4. 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 v5.2.5. For more details about version 5.2.5 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/10/kamailio-v5-2-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Oct 21-23, 2019, Berlin, Germany -- https://asipto.com/u/kat

3 3

SSL Offloading with AWS ELB or Google GSLB and Kamailio
by Joel Serrano 10 Oct '19

10 Oct '19

Hi everyone, I was giving a try to setup Kamailio with a Cloud TCP load balancer in front, taking advantage of the newly added proxy protocol compatibility and my initial tests went very well. Flow: client -> (tcp) -> load balancer -> (tcp) -> Kamailio TCP socket I then did another quick test and enabled TLS, also with good results: Flow: client -> (tls) -> load balancer -> (tls) -> Kamailio TLS socket So far so good, proxy protocol works as expected. I wanted to go one step further and see if I could somehow offload SSL operations at the load balancer level, and leave kamailio handling plain tcp. Flow: client -> (tls) -> load balancer -> (tcp) -> Kamailio TCP socket This partially worked, and before I start digging into what I have to do to get it completely working, I'd like to know if anyone already has a similar setup, or even if Kamailio is able to handle such a scenario, the reason I'm asking is because of the headers, etc. In this last scenario, I receive in a TCP socket, a request with TLS headers all over the place.. INVITE sip:14a84f2016944eb0854ef0e9b71bfa10@app.mydomain.com:60655 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.16:60717;branch=z9hG4bK.KmUpamn5P;rport From: ... To: ... CSeq: 21 INVITE Call-ID: -j1QSnam9o Max-Forwards: 70 Route: <sip:sbc-test2.mydomain.com:443;lr> <http://sbc-test2.sbx.gii.me:443;lr>/> Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 436 Contact: <sip:linphone@A.B.C.D:60717;transport=tls>;+sip.instance="<urn:uuid:fabcb441-a348-49a7-948d-72448d6840eb>" I then forward this request via UDP to subsequent proxies for further processing, on the replies, my payload information back to the client should be TLS, although sent via a TCP socket.. Is this something that will not work by design? Is there any hack I can take advantage of? The goal would be for Kamailio to handle TLS headers via TCP socket, as the client expects TLS information, but the actual traffic should go in plan TCP, and the load balancer will take care of re-encrypting before replying to the client. Any ideas/suggestions/comments? I hope this email is understandable, I find it complicated to detail the exact problem, feel free to ask any questions if you don't understand anything. Thanks, Joel.

3 5

PUBLISH ERROR
by Social Boh 10 Oct '19

10 Oct '19

Hello, Kamailio 5.2.5. I'm testing presence on my Kamailio using different Clients (Jitsi, Bria5, Linphone). when they send the PUBLISH to Kamailio, all receive the same answer: 400 Invalid Request On the LOG file: ERROR: {1 21 PUBLISH O8-BtqLnzw} presence [publish.c:421]: ki_handle_publish_uri(): No E-Tag and no body found I'm pretty sure same configuration work with 5.2.3 version. Any hint? Regards -- --- I'm SoCIaL, MayBe

2 4

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