- sr-users - kamailio.org

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

08 Jan '20

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

3 11

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

08 Jan '20

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

2 1

WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

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

3 4

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

07 Jan '20

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

4 7

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

07 Jan '20

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

2 1

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

07 Jan '20

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

5 10

Siremis & freeswitch
by John Tuxies 07 Jan '20

07 Jan '20

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

4 11

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

06 Jan '20

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

1 0

SNI Query
by mahesh b 06 Jan '20

06 Jan '20

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

2 4

Installation issue
by Hassen Bencheikh 06 Jan '20

06 Jan '20

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

5 4

MSRP proxy
by Sergey Safarov 06 Jan '20

06 Jan '20

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

2 1

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

06 Jan '20

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

2 1

Tweaking TCP settings
by Dovid Bender 06 Jan '20

06 Jan '20

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

2 1

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

06 Jan '20

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

3 8

dlg_set_timeout error
by Pepelux 03 Jan '20

03 Jan '20

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

3 6

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

03 Jan '20

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

5 10

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

02 Jan '20

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

2 2

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

31 Dec '19

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

3 2

Calculate call duration
by Duarte Rocha 31 Dec '19

31 Dec '19

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

3 2

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

25 Dec '19

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

4 3

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

24 Dec '19

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

2 6

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

24 Dec '19

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

2 1

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

24 Dec '19

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

1 0

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

24 Dec '19

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

2 2

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

22 Dec '19

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

2 1

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

22 Dec '19

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

3 5

Need support
by Srikanth Magloori 21 Dec '19

21 Dec '19

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

4 3

Questions about Presence
by Володимир Іванець 20 Dec '19

20 Dec '19

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

2 2

Kamailio Product Support
by Simal Mathai 20 Dec '19

20 Dec '19

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

2 1

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

20 Dec '19

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

3 5

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

19 Dec '19

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

4 6

Kamailio SIP TLS Troubleshooting
by Sudhakar Parasher 19 Dec '19

19 Dec '19

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

4 10

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

19 Dec '19

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

2 2

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

19 Dec '19

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

2 1

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

18 Dec '19

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

6 11

kamcmd stuck
by Andrew Chen 17 Dec '19

17 Dec '19

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

2 2

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

16 Dec '19

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

3 6

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

16 Dec '19

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

2 4

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

16 Dec '19

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

2 1

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

15 Dec '19

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

2 1

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

15 Dec '19

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

1 0

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

14 Dec '19

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

3 3

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

13 Dec '19

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

4 16

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

12 Dec '19

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

3 2

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

12 Dec '19

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

1 0

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

12 Dec '19

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

2 4

xhttp_prom Metric Naming Issue
by ENGIN ESIN 12 Dec '19

12 Dec '19

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

2 1

Please Help
by Sharath 12 Dec '19

12 Dec '19

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

2 1

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

10 Dec '19

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

4 6

authcheck and fd variable
by Robert Lieback 10 Dec '19

10 Dec '19

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

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