- sr-users - kamailio.org

KEMI basic logging
by Eugene Prokopiev 30 Mar '18

30 Mar '18

Hi, Tried just to reject any request with some log message: $ cat kamailio/Dockerfile FROM centos:centos7 RUN yum update -y RUN yum install -y wget RUN wget -O /etc/yum.repos.d/kamailio.repo http://download.opensuse.org/repositories/home:/kamailio:/v5.1.x-rpms/CentO… RUN yum install -y kamailio kamailio-lua VOLUME /etc/kamailio ENTRYPOINT ["kamailio", "-DD", "-E"] $ cat kamailio/config/kamailio.cfg listen=udp:0.0.0.0:5060 loadmodule "tm.so" loadmodule "sl.so" loadmodule "xlog.so" loadmodule "app_lua.so" modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") cfgengine "lua" $ cat kamailio/config/kamailio.lua function ksr_request_route() KSR.log("===== request - from kamailio lua script\n") KSR.sl.send_reply(503, "Server not configured") end Result is: kamailio_1 | 12(18) CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 5 kamailio_1 | 0(1) ALERT: <core> [main.c:746]: handle_sigs(): child process 7 exited by a signal 11 kamailio_1 | 0(1) ALERT: <core> [main.c:749]: handle_sigs(): core was generated vms_kamailio_1 exited with code 1 But valid response '503 Server not configured' with commented KSR.log. So, how to log propertly? -- WBR, Eugene Prokopiev

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Re: [SR-Users] How to determine correct port to set in record-route header
by Pranathi Venkatayogi 30 Mar '18

30 Mar '18

Any help? From: Pranathi Venkatayogi Sent: Monday, January 23, 2017 2:35 PM To: 'Kamailio (SER) - Users Mailing List' <sr-users(a)lists.sip-router.org> Subject: How to determine correct port to set in record-route header Hi, I am using Kamailio behind NAT, unable to figure how to make it put “public ip” in Record-route header, I am manually inserting the hard-coded header myself as below. However now I am having trouble choosing the right port number in all scenarios. $var(dstPort) = 5061; if (dst_port==5060) { $var(dstPort) = 5060; } insert_hf("Record-Route: <sip:MY_PUBLICIP_ADDR:$var(dstPort);transport=tls;lr;nat=yes>\r\n", "Record-Route"); In one scenario I see the conflicting port numbers in “via” header vs “Record route”. The ACK is being set to port 5060 based on Record route header and is not being received by the callee. 2017-01-23 14:52:49.970233 [blink.exe 2780]: SENDING: Packet 11, +0:00:36.411867 10.0.27.108:58217 -(SIP over TLS)-> 172.31.211.31:5061 SIP/2.0 180 Ringing Via: SIP/2.0/TLS 63.149.103.72:5061;received=172.31.211.31;branch=z9hG4bK4799.7067bd2a48063748a4c353fa408eaefa.0 Via: SIP/2.0/UDP 10.0.16.52;rport=5060;branch=z9hG4bK4799.935a27c6ef5ad112225964cbe7c1be44.0;i=2 Via: SIP/2.0/TLS 10.11.200.12:51793;rport=51793;received=10.11.200.12;branch=z9hG4bKPj49f6aeca3f9b4155ab4c5304b544aa4d;alias Record-Route: <sip:63.149.103.72:5060;transport=tls;lr;nat=yes> Record-Route: <sip:10.0.16.52:5061;transport=tls;lr;nat=yes> Call-ID: 09ce10efa6a946bf9445ccc21857393e From: "cust1" <sip:cust1@devtranslation.sms-test.cyracom.com>;tag=95082caabecc42548c2fec5ccd29e5de To: <sip:spanish@translation.sms-test.cyracom.com>;tag=ec365cc8489b48a7bc5725d21b7d97a1 CSeq: 17422 INVITE Server: Blink 3.0.0 (Windows) Contact: <sip:20745891@10.0.27.108:58216;transport=tls> Allow: SUBSCRIBE, NOTIFY, PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, MESSAGE, REFER Content-Length: 0 Questions: 1. How can I let in-built record-route function automatically choose “public ip” so I can get rid of my manual insertion altogether. 2. If no option, what is the right way to choose the port on which packet is being received, so it is same as what is on “VIA”. 3. Any other pointers to improve the logic here?

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kamailio it does not start after reboot
by Atux Atux 30 Mar '18

30 Mar '18

i am installing Kamailio using apt in a debian 8 machine by following this guide http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour i managed to create a few users and through Jitsi to talk to each other. once i reboot the machine, then the problems start and Kamailio cannot start. here are a few details from my system. -/etc/apt/sources.list the part for Kamailio #Kamailio deb http://deb.kamailio.org/kamailio51 jessie main deb-src http://deb.kamailio.org/kamailio51 jessie main -Version info Distributor ID: Debian Description: Debian GNU/Linux 8.10 (jessie) Release: 8.10 Codename: jessie -root@debian:~# systemctl start kamailio Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. root@debian:~# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. failed! Any ideas, please?

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180 and 183 SIP messages
by Ali Taher 30 Mar '18

30 Mar '18

Hello everyone, I'm working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I'm routing it to B-Party. I'm facing an issue , when I'm getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not hearing the RBT and asking to send 180 rather than 183 . (the supplier is only sending 183 without 180). Is it possible using kamailio to replace this 183 message with 180 message , or even add 180 message while keeping the same 183 message ? Regards, Ali Taher

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kamailio & vmware
by Jean Cérien 29 Mar '18

29 Mar '18

Hello We are running a kamailio 5.0 on a VMWARE / VSPHERE 6.0 vm, with a couple of asterisk running on 2 physical hosts. Audio goes straight to the asterisk, no rtpengine / rtpproxy. I actually have no audio issues, but communication between the asterisk & kamailio for sip sometime fails - I get a few 408. I cant tell if this is network related or virtualisation related. Anyone has advice on kamailio on a VM, when it only handles sip ? Rgds J.

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Kamailio fails to start because of jsonrpl
by Dmitry 29 Mar '18

29 Mar '18

Kamailio fails to start and I see the following errors: kamailio[2115]: ERROR: <core> [core/pvapi.c:1085]: pv_parse_spec2(): bad tr in pvar name "jsonrpl" kamailio[2115]: ERROR: <core> [core/pvapi.c:1111]: pv_parse_spec2(): invalid parsing in [$(jsonrpl(body){kz.json, result.NRSETS})] at (4) kamailio[2115]: CRITICAL: <core> [core/cfg.y:3447]: yyerror_at(): parse error in config file /etc/kazoo/kamailio/dispatcher-role.cfg, line 243, colu...ult.NRSETS}) I installed kamailio-json-modules, Kamailio v5.1.2 installed from DEB packets from deb.kamailio.org What is the cause of the error?

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Kamailio to Asterisk Proxy with NAT
by Benjamin Marty 29 Mar '18

29 Mar '18

Hello, I have multiple Asterisk Servers in a private Network. They also have a Public IP via Destination NAT. Now I want to use a Kamailio Proxy in front of them. The Routing I want to do with the Kamailio Dispatcher module. My Question is now how the RTP Media Stream should/can flow. The clients are in different other networks. So P2P Media Stream isn't possible. Should I now run the RTP Stream Client - Asterisk or Client - Kamailio - Asterisk? Is there an Example for an such scenario? Thanks Benjamin

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Missing exit function on KEMI?
by Enrico Bandiera 29 Mar '18

29 Mar '18

Hello, I'm trying to use the KSR.x.exit() function in a KEMI python script but at runtime i get this error: KSR.x.exit() AttributeError: 'module' object has no attribute 'exit' It seems that the exit function is missing even though it is documented. My kamailio version is: kamailio 5.2.0-dev4 (x86_64/linux) (it's a nightly) Any idea on what I can do? Thanks, Enrico.

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Kamailio NAT AWS EC2
by Benjamin Marty 29 Mar '18

29 Mar '18

Hi, I am new to Kamailio and coming a bit from the Asterisk side. My goal is to set up a bunch of users on a Kamailio Instance on AWS EC2 and get them able to call each other and talk to each other over the Internet without P2P media stream. Is rtpproxy the tool that enables that? As far my knowledge is right Kamailio isn't able to do any sort of RTP handling. Regards, Benjamin

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kamailio with media server
by Atux Atux 28 Mar '18

28 Mar '18

Hi. New to the area of Kamailio. i am did install in debian kamailio with rtpproxy and i have created 3 users 1000-10002 (one for each jitsi user) they talk nice between them. I have followed this tutorial http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour and in less than 5 minutes i had my accounts registered. i would like to have a media server so the users could hears announcements and stuff. At the end i would like to have kamailio as a test lab PBX where i could connect my SIP trunk providers and my users to route calls. Is there any guide on how to setup a media server and the services, please?

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is t_flush_flags() really needed?
by Juha Heinanen 28 Mar '18

28 Mar '18

While testing xflags, i noticed that a regular flag that I set AFTER calling t_newtrans() stays set in onreply_route even when I do not call t_flush_flags(). In a route block: if (!t_newtran()) { ERROR(Failed to create new transaction for $rm <$ru>); exit; }; if (!isflagset(19)) INFO(******** flag 19 is NOT set after t_newtran()); setflag(19); and in onreply_route: if (isflagset(19)) INFO(********* flag 19 is set); else INFO(********* flag 19 is NOT set); Syslog gets: Mar 25 11:24:41 trout /usr/bin/sip-proxy[21362]: INFO: ******** flag 19 is NOT set after t_newtran() Mar 25 11:24:41 trout /usr/bin/sip-proxy[21433]: INFO: ********* flag 19 is set So what is the point of t_flush_flags()? -- Juha

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Terminate dialog from another dialog
by Joel Serrano 27 Mar '18

27 Mar '18

Hi guys, I'm trying to set kamailio to not allow simultaneous calls from the same user, but if a second call comes in, the behavior desired is to process that second call and terminate the previous (already active) one. Say you have call A that creates dialog A. Then you have call B come in, creates dialog B. Can I save some info from dialog (call) A in a htable for example, then check that htable for the needed info from dialog (call) B and terminate dialog (call) A if it exists? I think using dlg_get() and dlg_bye() would do the trick but not 100% sure how they work.. The example in the docs seems exactly the case: dlg_get --> "Search and set current dialog based on Call-ID, From-Tag and To-Tag parameters" if(dlg_get("abcdef", "123", "456")) { dlg_bye("all"); } After the dlg_get() and dlg_bye() what happens to call B? Does the route continue running or does it end? Will it automatically switch back to the current dialog or do you have to manually switch back running dlg_get() again? (Sorry if I'm not explaining myself correctly, let me know if it sounds confusing) Does anyone have experience with these two functions? Thanks, Joel.

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kamailio 5.0.6 try to using dialog module - dont starts dialog counting
by Karsten Horsmann 27 Mar '18

27 Mar '18

Hello, i try to get dialog-infos to see how many calls are running on my kamailio. But i can not figure out what steps are necessary to get this running. loading the module and the dependencies - check. set and dlg_flag via module param - check. MUST i use a DB?! Or what is missing? kamctl stats | grep dialog "dialog:active_dialogs = 0", "dialog:early_dialogs = 0", "dialog:expired_dialogs = 0", "dialog:failed_dialogs = 0", "dialog:processed_dialogs = 0", the dialog snippets (grep out ouf my config): #!define DLG_FLAG 8 loadmodule "dialog.so" modparam("dialog", "dlg_flag", DLG_FLAG) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "rr_param", "did") modparam("dialog", "profiles_with_value", "active_calls; inbound_calls; outbound_calls") modparam("dialog", "send_bye", 1) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_extra_hdrs", "Hint: inactivity timeout\r\n") # Added to requests generated locally by the module (e.g. BYE) modparam("dialog", "track_cseq_updates", 1) -- Kind Regards *Karsten Horsmann*

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To tags in 407 challenges
by Alex Balashov 27 Mar '18

27 Mar '18

Hi, This may be more a SIP question than a Kamailio question, but I am trying to figure out why 407 challenges from Kamailio contain To tags. I suppose I've not run across it before, and I can't find an RFC-based rationale. A negative reply to an invite transaction without an intermediate early dialog-forming reply (e.g. 183 with To tag) does not form a dialog, nor takes place within a context in which a dialog was created. So what gives? Insights appreciated! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Bug in evrexec module?
by Skip Morse 27 Mar '18

27 Mar '18

Hi all, anyone try the evrexec module yet? When I try, Kamailio fails to start and I'm getting an error that it can't find the event route I'm specifying in the modparam, but I started with trying the example from the docs. I think it might be a bug https://www.kamailio.org/docs/modules/5.1.x/modules/evrexec.html Here's what I tried: modparam("evrexec", "exec", "name=evrexec:timer;wait=1000;workers=1;") event_route[evrexec:timer] { $var(x) = 0; while(1) { xlog("$$var(x) is $var(x)\n"); $var(x) = $var(x) + 1; sleep("600"); } } Here's the log: 0(1765) ERROR: evrexec [evrexec_mod.c:232]: evrexec_param(): event route not found: evrexec:timer 0(1765) CRITICAL: <core> [core/cfg.y:3450]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 136, column 70: Can't set module parameter ERROR: bad config file (1 errors) Version info: kamailio -V version: kamailio 5.1.2 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 06:41:34 Mar 9 2018 with gcc 4.8.5 Thoughts? Thanks, -Skip

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Removing a cipher
by Wilkins, Steve 27 Mar '18

27 Mar '18

Hello all, I am using Kamailio 5 and am trying to remove the Diffie-Hellman ciphers but have had no luck. I have researched it and have not seen any articles yet where anyone has had any luck. I need to be able to connect from webrtc port 443. Has anyone had any luck with the cipher-list? Thank you, -Steve

1 0

Dispatcher Routing Failure
by Daniel W. Graham 27 Mar '18

27 Mar '18

What constitutes a 'failed request' in dispatcher module, in which a gateway is marked inactive? >From documentation - ds_ping_interval With this parameter you can define the interval for sending a request to a gateway marked as inactive upon a failed request routing to it. ds_probing_threshold If you want to set a gateway into inactive mode, there can be a specific number of failed requests until it will change from "active" to "inactive". Thanks, -dan

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Regular Expression Syntax
by Duarte Rocha 27 Mar '18

27 Mar '18

Greetings, I'm having some trouble with the syntax used in the {re.subst,expression} method. What i want to do is add a prefix to a capture group. I can make it work like that /^[0-9]+/prefix$tU/ but this solution is far from perfect since i may not need the entire $tU. I would like to use it like this /^[0-9]+/prefix\1/ in order to not limit my regex to Kamailio only. When i do it like this the regex result is only "prefix". What's the correct syntax to add a prefix using the capture group ? Thanks in advance

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topos_redis and session-timer timeouts
by Daniel Tryba 27 Mar '18

27 Mar '18

I was testing topos with a mysql backend and all was fine. But since the purpose of this machine doesn't require a complete mysqlserver I tried the redis backend. With redis the calls will drop after some timeout (somewhere < 450 seconds (session timer set to 900)). Config: # topos modparam("ndb_redis", "server", "name=srv2;addr=127.0.0.1;port=6379;db=2") modparam("topos", "storage", "redis") modparam("topos_redis", "serverid", "srv2") #modparam("topos", "storage", "db") #modparam("topos", "db_url", DBURL) modparam("topos", "mask_callid", 0) modparam("topos", "sanity_checks", 0) modparam("topos", "branch_expire", 180) modparam("topos", "dialog_expire", 10800) modparam("topos", "clean_interval", 60) The call will end on the first session-time firing with a "404 Not here" Debug contains: Mar 27 12:20:56 kam kamailio[11649]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:20:56 kam kamailio[11649]: DEBUG: ndb_redis [redis_client.c:878]: redisc_exec_argv(): rsrv->ctxRedis = 0x55a7c8da8500 Mar 27 12:20:56 kam kamailio[11649]: DEBUG: topos_redis [topos_redis_storage.c:980]: tps_redis_load_dialog(): hmap with key [d:z:atpsh-5aba19b2-2d7f-2] not found This records was inserted/updated: Mar 27 12:17:10 kam kamailio[11647]: DEBUG: topos_redis [topos_redis_storage.c:253]: tps_redis_insert_dialog(): inserted dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 26 Mar 27 12:17:10 kam kamailio[11647]: DEBUG: topos_redis [topos_redis_storage.c:282]: tps_redis_insert_dialog(): expire set on dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 3 Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:1293]: tps_redis_update_dialog(): updated dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11648]: DEBUG: topos_redis [topos_redis_storage.c:1293]: tps_redis_update_dialog(): updated dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:962]: tps_redis_load_dialog(): loading dialog record for [d:z:atpsh-5aba19b2-2d7f-2] Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:1381]: tps_redis_end_dialog(): updated on end the dialog record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 6 Mar 27 12:17:11 kam kamailio[11650]: DEBUG: topos_redis [topos_redis_storage.c:1412]: tps_redis_end_dialog(): expire set on branch record for [d:z:atpsh-5aba19b2-2d7f-2] with argc 3 It looks like the last update/expire sets the timeout to 180s, so the not found is reasonably expected. But why is it that these records aren't deleted in the mysql backend? 0000 00 00 03 04 00 06 00 00 00 00 00 00 00 00 08 00 ................ 0010 45 00 00 6d 0a 52 40 00 40 06 32 37 7f 00 00 01 E..m.R@.@.27.... 0020 7f 00 00 01 b1 0c 18 eb c4 68 f0 3b 93 17 20 b0 .........h.;.. . 0030 80 18 03 d6 fe 61 00 00 01 01 08 0a 00 08 50 67 .....a........Pg 0040 00 08 50 67 2a 33 0d 0a 24 36 0d 0a 45 58 50 49 ..Pg*3..$6..EXPI 0050 52 45 0d 0a 24 32 35 0d 0a 64 3a 7a 3a 61 74 70 RE..$25..d:z:atp 0060 73 68 2d 35 61 62 61 31 39 62 32 2d 32 64 37 66 sh-5aba19b2-2d7f 0070 2d 32 0d 0a 24 33 0d 0a 31 38 30 0d 0a -2..$3..180..

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Kamailio Call monitoring
by Tomi Hakkarainen 27 Mar '18

27 Mar '18

Hi you all, I have set up Kamailio and RTPProxy in front of Asterisk. Currently successfully able to route calls from two differrent SIP-Trunk providers terminated to Kamailio and RTPProxy. I would like to get a view of calls currently running through Kamailio? I should be able to see some details about the call like the calling called numbers and the time call has been active. I’m not sure if Siremis would provide such view. Also looked Homer and also not sure it has something like that. Still those provide much more like for troubleshooting. If anyone has idea how that would be possible to accomplish or know that there already is such implementation I would be interested. Tomi

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Kamailio send 2 Invite request for Client behind NAT
by amit Kumar 27 Mar '18

27 Mar '18

Hi All, Kamailio send 2 Invite request for Client behind NAT. due to this reason for one request,Client respond with 486 Busy here. we only want to send 1 INVITE request. Following are the AOR for client at kamailio.if field(ip port) in CONTACT and RECEIVED differ then kamailio send 2 INVITE request. Please advice if there is any way to send only ONE INVITE*.* AOR:: 123 Contact:: sip:123@*192.168.230.124:45672 <http://192.168.230.124:45672>* Q=1 Expires:: 894 Callid:: 003A8EC4-12351(a)192.168.230.124 Cseq:: 2 User-agent:: SIPPER for 3CX Phone Received:: sip:*204.192.206.208 <http://204.192.206.208>: 45672 * State:: CS_NEW Flags:: 0 Cflag:: 192 Socket:: udp:172.12.120.88:5060 Methods:: 4294967295 Ruid:: uloc-5aba112d-7fce-5 Reg-Id:: 0 Last-Keepalive:: 1522144583 Last-Modified:: 1522144583 thanks, amit

1 0

Kamailio NAT AWS EC2
by Ilie Soltanici 27 Mar '18

27 Mar '18

Hi, First of all, check if SIP Traffic is going through your SIP Server, not directly between Endpoints. To do this check if Record-route parameter is set in configuration file: record_route(); https://www.kamailio.org/docs/modules/5.1.x/modules/rr.html Also, be sure that in SDP body you have External IP of your Kamailio/RTPproxy. If Kamailio is behind NAT you may need to use fix_nated_sdp function from nathelper module: https://www.kamailio.org/docs/modules/5.1.x/modules/nathelper.html#nathelpe… Regards,

1 0

NOTIFY message without XML part
by denislav.tsonev＠musala.com 27 Mar '18

27 Mar '18

Hello, I use kamailio 5.1.1 with default configuration for PRESENCE: Route: #!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 and modparams: #!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 I use and GRANSTREAM phones. When I make subscribe from soft sip client (Micro sip) to the grandstreams, all NOTIFY messages are without XML. Subscriptions between two soft sip clients like Zoiper and Micro SIP works perfectly. Is this related to GRANDSTREAM phones - configuration or so... or is necessary to modify kamailio configuration and if Yes, please provide some hitns how to change kamailio config? Thanks in advance. Best regards. Denislav Raychev Tsonev | Integration and Infrastructure Engineer | Musala Soft JSC www.musala.com | t: +359 2 969 58 21 | m: +359 878 270 965 | f: +359 2 969 58 22 World Trade Center, 36 Dragan Tsankov blvd., Sofia 1057, Bulgaria

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Strange auth_challenge behaivor on 407
by Yuriy Gorlichenko 27 Mar '18

27 Mar '18

Hi I'm using version: kamailio 5.1.2 I have strange issue with auth_challenge kamialio works as NAT handler. When Register comes kamailio handles nat and save original ip:port at the received field in location table I found that when local port in contact is different than external port reply 407 on INVITE comes to externalip (from received) : internalport(from contact) And never reaches destination for exmaple Register comes from 192.168.1.10:50555 Kamailio saves at the Received 12.1.1.12:1030 but when invite comes form 12.1.1.12:1030 kamailio sends reply to 12.1.1.12:50555 i don't think that this is an issue in kamailio source code but i can find where i can configure it

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How should i configure kamailio connection freepbx?
by 星昊通技术 27 Mar '18

27 Mar '18

hello all: I install kamailio 5.1 and siremis 5.1, I want to configure kamailio connection freepbx, how should i configure kamailio vai siremis GUI? Thanks. ------------------ Best regards! alex Hiastar 深圳星昊通科技 Sangoma in APAC, 潮流网络IP话机，IPPBX Asterisk cards, SBC, NetBorder VOIP Gateway, Transcoding, 潮流网络IP话机 QQ：2933996052 website:www.hiastar.com|www.freepbx.cn, contact number: 13574109204 关注微信公众帐号：Asterisk-CN

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Help on setting UP SIP PRoxy
by Mohammed Ayub 26 Mar '18

26 Mar '18

HI, Myself Ayub from Phonology, Bangalore, India. We are trying to set up an SIP proxy server using Kamailio with Siremis as web interface. Have installed and configured kamailio and siremis but need help in config if sip domain , users and etc... Please revert. Regards Ayub

2 1

Can't get route[AUTH] working as expected.
by Aqs Younas 25 Mar '18

25 Mar '18

Greetings list. I can see that I was able to bypass the default route[AUTH] if I send an invite containing from_uri which is not local but requested line containing a local user. llisten=udp:172.16.40.10:5060 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; } Below INVITE get passed above auth route. INVITE sip:60129879190@172.16.40.10 SIP/2.0 Via: SIP/2.0/UDP 139.5.177.91:5060;branch=z9hG4bK31edc7f4;rport Max-Forwards: 70 From: <sip:0128888877@139.5.177.99>;tag=as2274e806 To: <sip:60129879190@172.16.40.10> Contact: <sip:0128888877@139.5.177.91:5060> Call-ID: 7b6d32bc6c679bb23eb248b955c0ac8b@139.5.177.91:5060 CSeq: 102 INVITE User-Agent: FPBX-13.0.194.2(13.17.0) Date: Fri, 23 Mar 2018 09:33:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 321 v=0 o=root 237494576 237494576 IN IP4 139.5.177.99 s=Asterisk PBX 13.17.0 c=IN IP4 139.5.177.99 t=0 0 m=audio 15332 RTP/AVP 0 18 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv >From INVITE and route[AUTH] I can see why it is being passed. But should not it by default authenticate every request if IP address is not allowed in permission module. Br, Aqs.

2 4

Best approach for Kamailio behind nat
by Evgeny Kuchkov 25 Mar '18

25 Mar '18

Sorry for my previous mail with typos. I have the current configuration located on google cloud servers: [Alice a.a.a.a]--->(GCN NAT x.x.x.x)--->(Instance with Kam 10.0.0.1)--->(next proxy 10.0.0.2) Currently I have configured a fake alias with IP x.x.x.x in Debain and set listeners this way: listen=udp:10.0.0.1:5060 advertise x.x.x.x:5060 listen=udp:x.x.x.x:5060 200OK from Kam to Alice: Via: a.a.a.a Record-Route: <sip:alias1@10.0.0.2;lr> (used for further internal routing) Record-Route: <sip:alias2@10.0.0.2;lr> Record-Route: <sip:x.x.x.x;lr> Contact: <sip:dnis@x.x.x.x> ACK recived from Alice with reversed RR: ACK sip:dnis@x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr>,<sip:alias1@10.0.0.2;lr> So what Kam does - it goes through loose_route() and dispatches two ACKs to x.x.x.x (from which I'd like to get rid of) and 10.0.0.2 First one is: ACK sip:alias1@10.0.0.2 Via: x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr> And the next one: ACK sip:alias1@10.0.0.2 Via: x.x.x.x Via: x.x.x.x (Doubles Via) Via: a.a.a.a Route: <sip:alias2@10.0.0.2;lr> So my question is - how to make Kam think that x.x.x.x it's own address and stop dispatching in-dialog messages to itself? Currently I'm thinking about chooping off one Route/Record-Route in route[WITHINDLG] tight before loose_route(). But maybe there's a better way to solve my problem?

1 0

Best approach for Kamailio behind nat
by Evgeny Kuchkov 24 Mar '18

24 Mar '18

Hi all, I have the current configuration located on google clod servers: [Alice a.a.a.a]--->(GCN NAT x.x.x.x)--->(Instance with Kam 10.0.0.1)--->(next proxy 10.0.0.2) Currently I have configured a fake alias with IP x.x.x.x in Debain and set listeners this way: listen=udp:10.0.0.1:5060 advertise x.x.x.x:5060 listen=udp:x.x.x.x:5060 200OK from Kam to Alice: Via: a.a.a.a Record-Route: <sip:alias1@10.0.0.2;lr> (used for further internal routing) Record-Route: <sip:alias2@10.0.0.2;lr> Record-Route: <sip:x.x.x.x;lr> Contact: <sip:dnis@x.x.x.x> ACK recived from Alice with reversed RR: ACK sip:dnis@x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.2;lr>,<sip:alias1@10.0.0.2;lr> So what Kam does - it goes through loose_route() and dispatches two ACKs to x.x.x.x (from which I'd like to get rid of) and 10.0.0.2 First one is: ACK sip:alias1@10.240.0.67 Via: x.x.x.x Via: a.a.a.a Route: <sip:x.x.x.x;lr>, <sip:alias2@10.0.0.1;lr> And the next one: ACK sip:alias1@10.240.0.67 Via: x.x.x.x Via: x.x.x.x (Doubles Via) Via: a.a.a.a Route: <sip:alias2@10.0.0.1;lr> So my question is - how to make Kam think that x.x.x.x it's own address and stop dispatching in-dialog messages to itself? Currently I'm thinking about chooping off one Route/Record-Route in route[WITHINDLG] tight before loose_route(). But maybe there's a better way to solve my problem?

1 0

dlg_manage() and dlg_flag
by Alex Balashov 24 Mar '18

24 Mar '18

Hi, One aspect of the dialog module that is very confusing to newbies is the relationship between the dlg_flag modparam and dlg_manage(). According to the documentation, dlg_manage() effectively seems to obsolete the requirement for a flag[1]: Process current SIP request with dialog module. It is an alternative to setting dialog flag for initial INVITE and Route-parameter-callback execution for within-dialog requests. Yet, if one does not set a dlg_flag modparam, the dialog module fails to load: Mar 23 15:42:29 evaristesys-legacy /usr/local/sbin/kamailio[2900]: ERROR: dialog [dialog.c:500]: mod_init(): invalid dlg flag -1!! Perhaps the flag modparam should be made optional, and any arguments for using one over the other more clearly explained in the documentation. I'd make the latter contribution, but unfortunately am not entirely sure what those are. Cheers, -- Alex [1] https://kamailio.org/docs/modules/5.1.x/modules/dialog.html#dialog.f.dlg_ma… -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 3

have successfully installed but ..
by abartlett＠vixtel.com.au 23 Mar '18

23 Mar '18

having trouble configuring kamalio to move calls between servers and carriers

3 3

tcpconn_main_timeout
by Kjeld Flarup 23 Mar '18

23 Mar '18

We are developing an app with a pjsip client. I see this error when trying to send an invite from kamailio. Mar 17 00:29:46 raspberrypi /usr/sbin/kamailio[18720]: ERROR: <core> [tcp_main.c:4258]: tcpconn_main_timeout(): connect 62.44.134.85:43829 failed (timeout) A few seconds before 62.44.134.85:43829 did a register. I can understand that either the TCP connection was close by the other end or no one is listening. How much can I see in a pcap from this scenario? I cannot see the invite, but somehow there must be some TCP communication to show what is failing. -- -------------------- 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

3 10

htable not visible in another thread
by Kjeld Flarup 22 Mar '18

22 Mar '18

Im trying to do a t_suspend, but cant get htable to work I save to $sht $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); xlog( "L_ALERT", "Suspended transaction sht(vtp=>join::$rU) = [$sht(vtp=>join::$rU)]\n" ); Mar 19 23:36:28 raspberrypi /usr/sbin/kamailio[2167]: ALERT: <script>: Suspended transaction sht(vtp=>join::004540294149) = [23690:1238111367] But in my register thread nothing is read xlog("L_ALERT","sht(vtp=>join::$tU) = $sht(vtp=>join::$tU)\n"); Mar 19 23:36:31 raspberrypi /usr/sbin/kamailio[2114]: ALERT: <script>: sht(vtp=>join::004540294149) = <null> # -------- htable params -------------- modparam("htable", "db_url", DBURL) modparam("htable", "htable", "vtp=>size=10;autoexpire=120;dbtable=htable;dbmode=1") modparam("htable", "htable", "a=>size=6;") I never see any data in htable in the database. What am I doing wrong? -- -------------------- 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

3 3

phonenum module erratic behaviour with 5.1.2
by George Diamantopoulos 22 Mar '18

22 Mar '18

Hello all, It seems that the phonenum module in at least 5.1.2 (haven't tried previous versions) is behaving somewhat erratically. More specifically, if there are any references to any phn PVs with the key other than "number": if ( phonenum_match("1-484-555-8888", "src") ) { xlog("number norm: $phn(src=>number)\n"); xlog("number country: $phn(src=>country)\n"); dbg_pv_dump(30, "L_ERR"); } kamailio will not start: Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) DEBUG: <core> [core/pvapi.c:494]: pv_spec_lookup(): PV <$phn(src=>country)> is not in cache Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) DEBUG: phonenum [phonenum_pv.c:160]: pv_parse_phonenum_name(): phonenum [src] - key [country] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: phonenum [phonenum_pv.c:205]: pv_parse_phonenum_name(): error at PV phonenum name: src=>country Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: <core> [core/pvapi.c:951]: pv_parse_spec2(): pvar "phn" has an invalid name param [src=>country] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: <core> [core/pvapi.c:1106]: pv_parse_spec2(): wrong char [)/41] in [$phn(src=>country)] at [17 (5)] Mar 19 20:06:49 modcom-sbc-dev1 kamailio[5042]: 0(5044) ERROR: xlog [xlog.c:513]: xdbg_fixup_helper(): wrong format[number country: $phn(src=>country) If I comment the offending line, all is well, but when dumping PVs with dbg_pv_dump(30, "L_ERR");, only $phn(src=>number) is listed, not the other keys (country, operator, region according to the docs) However, what is more disturbing is that phn(src=>number) always contains the string provided to phonenum_match() function, even if invalid characters are included, with no normalization. For example: phonenum_match("14~8%^88", "src") will result in dbg_pv_dump reporting: "$phn(src=>number)": "14~8%^88" Please note that I'm using sipwise's debian repository for stretch for kamailio, and debian's stable stretch repository for libphonenumnber (currently at 7.1.0-5+b1). I have also tried building current libphonenumber myself (version v8.9.2), but the result is the same. Is this a bug? Or am I missing something? Thanks! BR, George

3 4

Kamailio htable.dump command returns - reply too big
by José Seabra 22 Mar '18

22 Mar '18

Hello there, I have an htable with 3000 entries and when i try dump the htable through the command kamcmd htable.dump htable, this command returns "ERROR: reply too big". I have tried to increase the body size using the following modparam: modparam("ctl", "binrpc_max_body_size",1024) But the result still the same "ERROR: reply too big", do I need to configure another parameter to be able dump this htable? The kamailio version that i'm using is 4.4.2 Thank you. Regards José Seabra

2 3

Kamailio behind NAT no Audio
by Benjamin Marty 22 Mar '18

22 Mar '18

Hi, I am new to Kamailio and coming a bit from the Asterisk side. Currently I'm using the Amazon EC2 Infrastructure and set up an Kamailio with standard configuration. EC2 uses Destination NAT on their side. Thus I enabled WITH_NAT and WITH_NATSIPPING. I'm using the rtpproxy from master branch of https://github.com/miconda/rtpproxy. Besides those two parameters I also changed the listen parameter in the Kamailio config: listen=udp:10.0.0.221:5060 advertise 52.59.XXX.XXX:5060 My rtpproxy is running with following parameters: ./rtpproxy -A 52.59.XXX.XXX -F -l 10.0.0.221 -m 20000 -M 30000 -s udp:*:7722 -d INFO When I use two Softclients (Zoiper) on different Networks (with Source NAT on the Firewall) the Softclient tries to transmit the RTP Packets to the Public IP of the other Network. But the Client obviously thus doesn't get the RTP Packets. Is there an easy and documented way to solve that issue? If I understood it correctly Kamailio together with rtpproxy should be able to receive the RTP Packets and forward them to the other client. So it should signal the IP of the Kamailio to send the RTP packets to. Or am I on the complete wrong way? Regards, Benjamin

1 0

avpops and mongo problem
by hdssdsdsdsfsdf hdssdsdsdsfsdf 22 Mar '18

22 Mar '18

Hi all, I have kamailio setup that uses a mongo cluster, and everything works fine. Only I have problems with the avpops module. The avp_db_load fails with "field [attribute] not found in result iterator", even if the attribute field is present in the object in mongo. Does anybody know what could be going wrong? Thanks in advance, and kind regards, =============================================================================== kamailio.cfg: ... # ---- avpops params ---- modparam("avpops", "db_url", DBURL) modparam("avpops", "avp_table", "usr_preferences") ... xlog("trying avp_db_load\n"); if (avp_db_load("$fu/username", "$avp(s:attributevalue)")) { xlog("if succeeded\n"); } xlog("if not succeeded\n"); ... =============================================================================== mongo: rs:SECONDARY> db.usr_preferences.find({"username":"1234567890"}) { "_id" : ObjectId("5ab22a5b384cd2aa0d4cf403"), "attribute" : "attributevalue", "username" : "1234567890", "domain" : null, "value" : "750", "uuid" : "", "type" : 0 } =============================================================================== syslog: ... Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: <script>: trying avp_db_load Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:851]: db_mongodb_query(): query to collection [usr_preferences] Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:885]: db_mongodb_query(): query filter: { "username" : "1234567890", "attribute" : "attributevalue" } Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:904]: db_mongodb_query(): columns filter: { "value" : 1, "attribute" : 1, "type" : 1 } Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:116]: db_new_result(): allocate 56 bytes for result set at 0x7f43b613c480 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:349]: db_mongodb_get_columns(): 3 columns returned from the query Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:154]: db_allocate_columns(): allocate 24 bytes for result names at 0x7f43b613c318 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 12 bytes for result types at 0x7f43b614d888 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[0] named: value Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:396]: db_mongodb_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7f43b614d9b8 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:431]: db_mongodb_get_columns(): use DB1_STRING result type Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:460]: db_mongodb_get_columns(): RES_NAMES(0x7f43b614d9b8)[0]=[value] (2) Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[1] named: attribute Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): field [attribute] not found in result iterator Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): failed to set the columns Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to store result Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to do the query Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 3 columns Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7f43b614d9b8 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result names at 0x7f43b613c318 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result types at 0x7f43b614d888 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7f43b613c480 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: avpops [avpops_impl.c:381]: ops_dbload_avps(): db_load failed Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: <script>: if not succeeded ... ===============================================================================

3 3

drouting & rd
by Jean Cérien 22 Mar '18

22 Mar '18

Hello I have setup 2 gateways in my dr_rules: 5,6 - so the 2nd one is tried if the first fails. The second one requires credentials. When the first one fails: I execute the following code: failure_route[ROUTEFAIL] { if ( t_check_status("[345][0-9][0-9]") or (t_branch_timeout() and !t_branch_replied() )) { t_on_failure("ROUTEFAIL"); if (use_next_gw()) { xlog("L_INFO","failure_route(ROUTEFAIL) @@ routing to: rd: $rd, ru:$ru, t_relay"); t_relay(); exit; } then in the same ROUTEFAIL (before this code), I have if (t_check_status("401|407")) { xlog("L_INFO","failure_route(ROUTEFAIL) @@ call to uac_auth(), $si rd: $rd\n"); ..... So, when the first gateway replies with a 404, it executes correctly the first block, finds the second gw and I see in the log that rd is set to its address. Yet, when the 2nd gw replies with 401, rd is set to the very first gw. However, the pcap traces confirm it is indeed the second that replies this. If this behaviour is correct, I cant seem to find the source IP of the gateway that replied to me, to retrieve credentials from the db. version: kamailio 5.0.5 / 5.0.6 (tested on both) Thank you for your help J

2 1

XMLRPC with params
by David Villasmil 22 Mar '18

22 Mar '18

Hello guys, I'm trying to call an xmlrpc method with parameters like: <?xml version="1.0"?> <methodCall> <methodName>dispatcher.set_state</methodName> <params> <param> <value><string>state:i</string></value> </param> <param> <value><string>group:1</string></value> </param> <param> <value><string>sip:1.2.3.4:5060</string></value> </param> </params> </methodCall> I'm getting back "Invalid Parameters". I can't find any documentation as to _how_ to call it with parameters. Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 2

(no subject)
by hdssdsdsdsfsdf hdssdsdsdsfsdf 21 Mar '18

21 Mar '18

Hi all, I have kamailio setup that uses a mongo cluster, and everything works fine. Only I have problems with the avpops module. The avp_db_load fails with "field [attribute] not found in result iterator", even if the attribute field is present in the object in mongo. Does anybody know what could be going wrong? Thanks in advance, and kind regards, =============================================================================== kamailio.cfg: ... # ---- avpops params ---- modparam("avpops", "db_url", DBURL) modparam("avpops", "avp_table", "usr_preferences") ... xlog("trying avp_db_load\n"); if (avp_db_load("$fu/username", "$avp(s:attributevalue)")) { xlog("if succeeded\n"); } xlog("if not succeeded\n"); ... =============================================================================== mongo: rs:SECONDARY> db.usr_preferences.find({"username":"1234567890"}) { "_id" : ObjectId("5ab22a5b384cd2aa0d4cf403"), "attribute" : "attributevalue", "username" : "1234567890", "domain" : null, "value" : "750", "uuid" : "", "type" : 0 } =============================================================================== syslog: ... Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: <script>: trying avp_db_load Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:851]: db_mongodb_query(): query to collection [usr_preferences] Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:885]: db_mongodb_query(): query filter: { "username" : "1234567890", "attribute" : "attributevalue" } Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:904]: db_mongodb_query(): columns filter: { "value" : 1, "attribute" : 1, "type" : 1 } Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:116]: db_new_result(): allocate 56 bytes for result set at 0x7f43b613c480 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:349]: db_mongodb_get_columns(): 3 columns returned from the query Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:154]: db_allocate_columns(): allocate 24 bytes for result names at 0x7f43b613c318 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:165]: db_allocate_columns(): allocate 12 bytes for result types at 0x7f43b614d888 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[0] named: value Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:396]: db_mongodb_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7f43b614d9b8 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:431]: db_mongodb_get_columns(): use DB1_STRING result type Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:460]: db_mongodb_get_columns(): RES_NAMES(0x7f43b614d9b8)[0]=[value] (2) Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: db_mongodb [mongodb_dbase.c:377]: db_mongodb_get_columns(): Found a field[1] named: attribute Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): field [attribute] not found in result iterator Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): failed to set the columns Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to store result Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to do the query Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:79]: db_free_columns(): freeing 3 columns Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:83]: db_free_columns(): freeing RES_NAMES[0] at 0x7f43b614d9b8 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:92]: db_free_columns(): freeing result names at 0x7f43b613c318 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:97]: db_free_columns(): freeing result types at 0x7f43b614d888 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: DEBUG: <core> [db_res.c:134]: db_free_result(): freeing result set at 0x7f43b613c480 Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: avpops [avpops_impl.c:381]: ops_dbload_avps(): db_load failed Mar 21 17:15:16 ccccc /opt/kamailio/sbin/kamailio[2544]: ERROR: <script>: if not succeeded ... ===============================================================================

1 0

Problems with dispatcher reload
by Arik Halperin 21 Mar '18

21 Mar '18

Hello, I’ve installed kamailio 5.1 but when I run: kamctl dispatcher reload I get: { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" }, "id": 31222 } How do I fix this? Thanks, Arik Halperin

4 6

How does Kamailio decide which msg to store when receiving it double from 2 servers?
by Zigelman, Amit 21 Mar '18

21 Mar '18

I am using Captagent on 2 sip servers communicating with each other. Both Captagents then route the SIP traffic to the same Kamailio. Therefore, if I understand, Kamailio will get every msg between them twice. Once from server A and once from server B. Let's say that each of them has a different capturing code (100 for server A and 200 for server B) How will the data look in the DB? Will it hold both entries pretty much the same with a difference is some fields such as the capture node? Thanks a lot, Amit -- *Amit Zigelman* | *Software Developer* Mobile (+972) 54-570-9129 <+972%254-570-9129> | amit.zigelman(a)vonage.com

2 1

authenticating to mongodb fails
by hdssdsdsdsfsdf hdssdsdsdsfsdf 21 Mar '18

21 Mar '18

When I connect using mongo, the credentials work: $ mongo kamailio -u "kam" -p "kam" MongoDB shell version v3.4.9 connecting to: mongodb://127.0.0.1:27017/kamailio MongoDB server version: 3.4.9 > db.subscriber.find() { "_id" : ObjectI ... But when I use the following DBURL in kamailio.cfg, kamailio fails to even login to mongo: #!define DBURL "mongodb://kam:kam@localhost/kamailio" If I disable mongodb authentication, kamailio starts up just fine again. Any idea what's going wrong here?

3 10

NATDETECT Flag 16
by Duarte Rocha 21 Mar '18

21 Mar '18

Greetings, I have a doubt about the flag 16 used in nat_uac_test() belonging to the module NATDETECT. In the documentation, the following is stated : *16* - Test if the source port is different from the port in the “Via” header I have some situations that I receive a request from port 5060 and the Via Header contains no port. I think it probably assumes that it is the default port. Will this situation trigger the "16" flag ? Thanks in advance, Cheers

2 2

S-CSCF can not allocate memory
by tyd 21 Mar '18

21 Mar '18

Hello, I haved install debian 8 and kamailio 4.4.7 for IMS testing. After registering 10000 users, SIPp generates CPS 25 SIP INVITE message and BYE message after 100 seconds. After about 240000 calls, there are "cannot allocate memory" messgae on S-CSCF. The command of S-CSCF is "kamailio -m 512 -f /etc/scscf/kamailio.cfg" The parameter mem_join=1 is included in kamailio.cfg. I have modified the share memory to 1024 and 2048, but the result is the same, just increasing the pass calls. Would you please tell me how to solve the memory problem? Thanks. Mar 21 12:25:03 Kamailio440 kamailio[16483]: ERROR: <core> [sip_msg_clone.c:981]: msg_lump_cloner(): cannot allocate memory Mar 21 12:25:03 Kamailio440 kamailio[16483]: ERROR: tm [t_fwd.c:1656]: t_forward_nonack(): ERROR: t_forward_nonack: failed to save the message lumps Mar 21 12:25:03 Kamailio440 kamailio[16483]: ERROR: tm [t_reply.c:545]: _reply_light(): ERROR: _reply_light: cannot allocate shmem buffer Mar 21 12:25:03 Kamailio440 kamailio[16483]: ERROR: tm [t_lookup.c:1489]: t_unref(): ERROR: t_unref: generation of a delayed stateful reply failed Mar 21 12:25:03 Kamailio440 kamailio[16446]: ERROR: tm [t_reply.c:1893]: relay_reply(): ERROR: relay_reply: cannot alloc reply shmem Mar 21 12:25:03 Kamailio440 kamailio[16446]: ERROR: tm [t_reply.c:545]: _reply_light(): ERROR: _reply_light: cannot allocate shmem buffer Mar 21 12:25:03 Kamailio440 kamailio[16435]: ERROR: tm [t_reply.c:316]: update_totag_set(): ERROR: update_totag_set: no memory Mar 21 12:25:03 Kamailio440 kamailio[16406]: ERROR: ims_dialog [dlg_profile.c:472]: set_dlg_profile(): no more shm memory Mar 21 12:25:03 Kamailio440 kamailio[16406]: ERROR: ims_dialog [ims_dialog.c:619]: w_set_dlg_profile(): failed to set profile Mar 21 12:25:03 Kamailio440 kamailio[16406]: ERROR: tm [t_lookup.c:1289]: new_t(): ERROR: new_t: out of mem: Mar 21 12:25:03 Kamailio440 kamailio[16406]: ERROR: tm [t_lookup.c:1429]: t_newtran(): ERROR: t_newtran: new_t failed root@Kamailio440:~# kamcmd tm.stats { current: 69 waiting: 0 total: 991300 total_local: 0 rpl_received: 2183891 rpl_generated: 20232 rpl_sent: 1475856 6xx: 0 5xx: 2704 4xx: 10124 3xx: 0 2xx: 978329 created: 991300 freed: 991231 delayed_free: 0 } root@Kamailio440:~# kamctl stats shmem shmem:fragments = 1630 shmem:free_size = 4303040 shmem:max_used_size = 536870912 shmem:real_used_size = 532567872 shmem:total_size = 536870912 shmem:used_size = 415668000

1 0

execute only on INITIAL INVITE
by David Villasmil 20 Mar '18

20 Mar '18

Hello guys, I want to execute an sql only for INITIAL INVITEs, I'm trying like: #!ifdef WITH_DID_VALIDATE if(!has_totag()) { sql_query("acd", "select * from did_numbers where didNumber = '$tU' and active = 'Y'", "ra"); xlog("L_ERR", "[REQUEST_ROUTE] number of rows in table domain: $dbr(ra=>rows)\n"); if( $dbr(ra=>rows)==0 ) { xlog("L_ERR", "[REQUEST_ROUTE] DID Number validation returned [$dbr(ra=>rows)], not responding to this request\n"); exit; } sql_result_free("ra"); } #!endif But, say I remove the DID from the allowed list while a call is going on. Then responses to the allowed INVITE would not be processed. How would I limit that to ONLY the INITIAL INVITE? Should I use has_totag? Thanks! David Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

3 4

kamailio 4.4.7 websockets
by draden vandewind 20 Mar '18

20 Mar '18

Hello, Can you explain to me how configure kamailio for to use websocket. I search a kamailio.cfg file with websocket support.... best regards draden

2 1

Security announcement related to Kamailio
by Henning Westerholt 19 Mar '18

19 Mar '18

Hello, as already announced from Daniel-Constantin Mierla on the lists last Wednesday [1], we strongly advise you to update your Kamailio installation to the latest stable release for security reasons. All supported releases (4.4, 5.0. and 5.1) contains two important security fixes related to the tmx and lcr module. Technical details for the tmx issue: A specially crafted REGISTER message with a malformed branch or From tag triggers a so called "off-by-one heap overflow". This vulnerability existed in the tmx module and makes it possible to remotely crash the Kamailio service. If an attacker sends many of this messages this would lead to a Denial of Service of the attacked infrastructure. This is especially critical as no authentication for the remote source is needed. This vulnerability was found from Sandro Gauci and Alfred Farrugia from the Security Company Enable Security. Many thanks to them for finding the issue and reporting it to us. You find all the details including a proof of concept code in the published security announcement from them: https://github.com/EnableSecurity/advisories/tree/master/ES2018-05-kamailio… Technical details for the lcr issue: A vulnerability existed in the lcr next_gw() function. It happens when a very long R-URI username is sent with an INVITE due to an mistake in the function error code handling. It can be triggered from a remote source, but should be only from a trusted peer, as it expected that calls going through lcr are authenticated by user or IP address. This vulnerability was reported from an user in the Netherlands to us, thanks as well for the bug report. So far we are not aware of any public exploits of this errors. But as already mentioned, we advise you to update your Kamailio servers to the latest stable release as soon as possible, especially as the tmx vulnerability will reported to more security lists later today. Please address any detailed technical questions related to the two bugs to the developer list at sr-dev(a)lists.kamailio.org . In case of confidential remarks related to this or other security issues, please address them to the Kamailio Management at management(a)kamailio.org . Best regards, Henning Westerholt Kamailio Project [1] https://lists.kamailio.org/pipermail/sr-users/2018-March/100672.html

1 2

msilo on proxy in front of registrar
by Stefan Sayer 19 Mar '18

19 Mar '18

Hello, I have a bit of a strange situation where I want to add SIP MESSAGE reliability (store and forward) for a proxy that is in front of the (message-wise unreliable) registrar/message originator. So I m_store MESSAGEs in the failure_route of MESSAGE requests. By default, msilo's m_dump works only for REQUEST_ROUTE, thus I can't use that in the 200 to REGISTER. But when I add a function to msilo for ONREPLY_ROUTE (without the check for MESSAGE support) it works. So, is there a reason that there's m_dump only for REQUEST_ROUTE? And, is this: if ((msg->first_line).type == SIP_REQUEST) the correct way to check whether we're handling a request or reply? Also, is there a way to check the expires of contacts in the 200 to REGISTER from the script, so I don't try to m_dump on de-REGISTER? Or does this have to go into the m_dump function for the ONREPLY_ROUTE case? Many thanks for hints! Stefan

2 1

Some clarifications needed on async_task_route
by George Diamantopoulos 19 Mar '18

19 Mar '18

Hello all, I've been exploring the async module a little, and async_task_route() more specifically. I have async_workers set, and the async module loaded. However, whenever this async function is involved in SIP processing, kamailio returns a 500 error to the UAC. I was hoping that someone could confirm that this function is suitable for my needs, I'm not sure I understand the docs very well on this one. So the plan is to have some of the processing of the transaction performed asynchronously, as it is not required for routing etc. Here's an example of what I'm trying to achieve, it's not really critical if NICE_TO_HAVE route functions fail etc, I just need request_route processing to continue normally without waiting for async_task_route("NICE_TO_HAVE") to finish whatever it's going to be doing: request_route() { ... SIP_PROCESSING_A ... async_task_route("NICE_TO_HAVE") ... SIP_PROCESSING_B t_relay(); } route(NICE_TO_HAVE) { dlg_var(sth) = something_derived_from_global_vars(); } Is this possible, or am I completely out of scope here? Thanks! BR, George

2 1

INFO: Relevant fixes in the last releases
by Daniel-Constantin Mierla 19 Mar '18

19 Mar '18

Hello, I want to highlight that the last stable versions (for the latest 3 release series: 4.4, 5.0 and 5.1) include fixes for two issues that can crash a running instance of Kamailio, therefore it is strongly recommended to upgrade if you are using tmx or lcr modules. Next week a CVE report is going to be created with more details about one of these issues. The issues were reported privately, one by security researchers and one by a community member, and were fixed quickly. The code related to the reported issues is rather old (few years by now) and there are no known incidents of exploiting these issues so far. However, once the CVE report comes public, there could be a higher risk of exploitation. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

2 2

Kamailio World 2018 - grants for students and underrepresented people
by Daniel-Constantin Mierla 19 Mar '18

19 Mar '18

Hello, we are glad to announces that, thanks to the sponsors, we can offer again couple of free seats at Kamailio World Conference, May 14-16, 2018, in Berlin, for students as well as for underrepresented people. In this way we are continuing the program from last years, based on the roots and the tight relation of Kamailio project with the academic environment, the eligible people being students enrolled in universities or research institutes (bachelor, master and PhD programs qualify) as well as people from underrepresented groups. If you think you are eligible and want to participate, email me directly (miconda(a)gmail.com) or to registration(a)kamailio.org . Participation to all the content of the event (workshops, conference and social event) is free, but you will have to take care of expenses for traveling and accommodation. Write a short description about your interest in real time communications and, when it is the case what is the university or the research institute you are affiliate to. Also, if you are not a student, but you are in touch with some or have access to students forums/mailing lists, it will be very appreciated if you forward these details. More information about Kamailio World is available on the web site: - https://www.kamailioworld.com Many thanks to the event sponsors that allowed to continue this program, respectively: Evosip, 2600hz, Sipwise, Sipgate, Simwood, NG-Voice, Evariste Systems, Digium, LOD.com, Pascom, Core Network Dynamics, FhG Fokus and Asipto. Expect a full house event! Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Select last URI from Record-Route
by John Petrini 19 Mar '18

19 Mar '18

Hello list, Does anyone how I can extract the last URI from a compact Record-Route header? I've tried using a negative index in the parameter of my select but it returns the first URI in the header instead of the last. $var(rr_last_uri) = @record_route.uri[-1]; Maybe there's another way instead of using selects? Thanks, John

4 7

Select last URI from Record-Route
by Ilie Soltanici 18 Mar '18

18 Mar '18

Hi, Try $(hdr(Record-Route)[-1]), to extract URI only you can use: $(hdr(Record-Route)[0]{nameaddr.uri})

1 0

Logging unregister
by Kjeld Flarup 17 Mar '18

17 Mar '18

Is there a way to get a log entry when a client unregisters incl. when tcp connection is lost for a registration. -- -------------------- 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 as load balancer
by Seyed Mohammad Mirheydari 16 Mar '18

16 Mar '18

Hi I have 5 asterisk servers. And I want to use kamailio as load balancer. How can route sip calls from outside to asterisk servers? Note: I installed kamailio and mairadb , etc. I configured subscriber for kamailio. And add my 5 servers to databse. Hust to know how to change the file kamailio.cfg to route all incoming calls to my 5 asterisk server (round robin). thanks

3 2

Need useful graphics ideas
by Karsten Horsmann 16 Mar '18

16 Mar '18

Hello List, I thought about some kind of Kamailio stats source (like registered users, calls active and some other things) to collect them into influx dB and draw them with grafana. How do you solved that? Timer based routes or statsd or whatever? Kind regards Karsten Horsmann

5 10

Lua script reload
by Igor Olhovskiy 16 Mar '18

16 Mar '18

Hi! I’m playing with new KEMI framework and trying to reload script without reloading Kamailio. Lua code is mostly one from example. ... modparam("app_lua", "load", "/etc/kamailio/lua/kamailio.lua") cfgengine "lua" ... So, I’m changing lua code and giving command kamctl rpc app_lua.reload But in the answer { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": 11098 } In syslog there is a string /usr/sbin/kamailio[10939]: ERROR: app_lua [app_lua_api.c:399]: lua_sr_reload_script(): reload is not activated What is need to be done to activate reload? Regards, Igor

2 12

logging sip messages
by Karthik Srinivasan 15 Mar '18

15 Mar '18

Hello: I am trying to find a simple way to log each and every SIP message that Kamailio receives or Kamailio sends. In a nutshell, here is what i'm trying to accomplish: - log sip message received; log ip address and port of where message was received from - log sip message to be sent or sent; log ip address and port of destination. I am trying to achieve this in the context of my general log file that contains routing/event logic from the kamailio.cfg file. By this i mean i don't want to have to run an external program like tcpdump to capture the packets and log them. I have tried a few thing but haven't been able to capture all conditions. examples: i have included things like event_routes for [tm:local-response]; [sl:local-response]; [sl:filtered-ack], onreply_route... I can capture most replies and responses via those but am not able to capture a 200ok relayed response from Kamailio back to the originating UA for example. I can log incoming messages without problem. I have also tried to activate IO interrupt to route to event_route[NETWORK:MSG] but unfortunately i can't access the source/dest ip address/port info or other pseudo variables. So back to the original question: How does one log every SIP message that Kamailio gets? How does one log every SIP message that Kamailio sends? via the context of the routing/replies/event routes in kamailio.cfg. Thanks, Karthik

3 3

UAC Restore Mode
by Duarte Rocha 15 Mar '18

15 Mar '18

Greetings, I have a question about the UAC module and its restore mode. If i use the "auto" restore mode in the "restore_mode" parameter will the entire info be saved in the From Tag for restore? I'm asking because i will implement High Avaliability and need to know if in event of a failure and i switch to a kamailio in a different machine i will be able to do the To\From restore. Thanks in advance, Duarte Rocha

1 0

Kamailio Edge Proxy & Kamailio Core Sip Server behind NAT
by Ed 14 Mar '18

14 Mar '18

Hi all, I would like to thank you all for the work you are doing. Having a Kamailio Edge proxy (2 interfaces - 1 pvt, 1 public) & Kamailio Core Sip server behind NAT, is it necessary to set up rtpengine on both servers? Thanks, Ed

2 1

kamailio implamentation
by kossivi HOLONOU 14 Mar '18

14 Mar '18

Hello, I am developing a project on MSRP Kamailio with webRTC client for my end of training project, can you help me with documentation or configuration side kamailio and webRTC client to implement to get there? Thank you for your support BesT Regards,Kossivi HolonouBO VAS EngineerDakar-Sénégal

1 0

Dimensioning ds_hash_size value for dispatcher module
by Alvaro Araos 14 Mar '18

14 Mar '18

Hi, In the documentation of the dispatcher module it is said that in order to use the algorithm 10 (load balancing) the parameter ds_hash_size must be set, in the definition it is said that the value is in powers of 2, that is to say if it is set to 8, this is equal to 2 ^ 8 = 256 slots, but still I do not understand how to dimension this value for my needs. I do not know whether to put 8, 4, 9, etc. Saludos, ALVARO ARAOS MÉNDEZ Director de Tecnología en FONOIP +56 2 2947 9000 alvaro(a)fonoip.com | www.fonoip.com [1] Moisés Chacón 46, El Monte, RM Links: ------ [1] http://www.fonoip.com

2 2

Kamailio With SMPP
by Amar Tinawi 14 Mar '18

14 Mar '18

Hello Is there a method to integrate kamailio to an smpp smsc ? Thanks

2 1

kamailio Rx_AAR qusetions
by Lv, Victor (NSB - CN/Qingdao) 14 Mar '18

14 Mar '18

Hi All, I am trying to send two Rx_AAR after Kamailio receive 200 OK with SDP: The first one for termination part is worked well, but in MT_aar_reply route block, it's not works. The one I marked with red color, is there anyone know how configure Kamailio to send two Rx_AAR after receive 200 OK SDP? Thanks a lot!! Victor +++++++++++++++++++++++++++++++++++++++++++++++++++++++ if (t_check_status("200")){ xlog("L_DBG", "IMS: Received 200 nside orig_initial_reply\n"); $var(aarret) = Rx_AAR("MT_aar_reply", "term","",2); xlog("L_DBG", "AAR return code is $var(aarret)\n"); } on MT_aar_reply block: route[MT_aar_reply] { #this is async so to know status we have to check the reply avp switch ($avp(s:aar_return_code)) { case 1: xlog("L_DBG", "Diameter: Term AAR success on media authorization\n"); break; default: xlog("L_ERR", "IMS: AAR failed Term\n"); xlog("L_ERR", "IMS: ttag: "+ "$avp(TTAG_CUSTOM_AVP)"); xlog("L_ERR", "IMS: ftag: "+ "$avp(FTAG_CUSTOM_AVP)"); xlog("L_ERR", "IMS: callid: "+ "$avp(CALLID_CUSTOM_AVP)"); #comment this if you want to allow even if Rx fails if(dlg_get("$avp(CALLID_CUSTOM_AVP)","$avp(FTAG_CUSTOM_AVP)","$avp(TTAG_CUSTOM_AVP)")){ dlg_terminate("all", "Sorry no QoS available"); exit; } $var(aarret) = Rx_AAR("MO_aar_reply", "term","",2); xlog("L_DBG", "AAR return code is $var(aarret)\n");

4 8

kamailio does not start with db_mongdb and usrloc
by Stephane Focant 13 Mar '18

13 Mar '18

Hello, I have an issue when I try to use kamailio 5.1 with usrloc and db_mongdb modules. When I try to start kamailio, I get the following error: 0(22216) DEBUG: <core> [db.c:314]: db_do_init2(): connection 0x7f735b5c4100 not found in pool 0(22216) DEBUG: db_mongodb [mongodb_connection.c:55]: db_mongodb_new_connection(): connection open to: mongodb://10.0.12.149 ,10.0.12.150, 10.0.12.151/kamailio?replicaSet=sre_location&slaveOk=true&readPreference=primaryPreferred 0(22216) DEBUG: db_mongodb [mongodb_dbase.c:864]: db_mongodb_query(): query to collection [version] 0(22216) DEBUG: db_mongodb [mongodb_dbase.c:898]: db_mongodb_query(): query filter: { "table_name" : "location" } 0(22216) DEBUG: db_mongodb [mongodb_dbase.c:917]: db_mongodb_query(): columns filter: { "table_version" : 1 } 0(22216) DEBUG: <core> [db_res.c:120]: db_new_result(): allocate 56 bytes for result set at 0x7f735b5c4450 0(22216) DEBUG: db_mongodb [mongodb_dbase.c:746]: db_mongodb_store_result(): no result from mongodb 0(22216) DEBUG: <core> [db.c:404]: db_table_version(): no row for table location found 0(22216) ERROR: <core> [db.c:450]: db_check_table_version(): invalid version 0 for table location found, expected 9 (check table structure and table "version") 0(22216) ERROR: usrloc [dlist.c:659]: register_udomain(): error during table version check. 0(22216) DEBUG: <core> [db_pool.c:100]: pool_remove(): removing connection from the pool 0(22216) ERROR: registrar [registrar.c:651]: domain_fixup(): failed to register domain 0(22216) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:660 0(22216) ERROR: <core> [core/rvalue.c:3860]: fix_rval_expr(): failure in cfg at line: 660 col: 22 0(22216) ERROR: <core> [core/rvalue.c:3860]: fix_rval_expr(): failure in cfg at line: 660 col: 22 0(22216) ERROR: <core> [core/route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/kamailio-5.1/etc/kamailio/kamailio.cfg:663 ERROR: error -1 while trying to fix configuration 0(22216) DEBUG: tm [t_funcs.c:85]: tm_shutdown(): start 0(22216) DEBUG: tm [t_funcs.c:88]: tm_shutdown(): emptying hash table 0(22216) DEBUG: tm [t_funcs.c:90]: tm_shutdown(): removing semaphores 0(22216) DEBUG: tm [t_funcs.c:92]: tm_shutdown(): destroying tmcb lists 0(22216) DEBUG: tm [t_funcs.c:95]: tm_shutdown(): done 0(22216) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(22216) DEBUG: <core> [core/mem/shm.c:261]: shm_destroy_manager(): destroying memory manager: q_malloc 0(22216) DEBUG: <core> [core/mem/q_malloc.c:1165]: qm_shm_lock_destroy(): destroying the shared memory lock 0(22216) DEBUG: <core> [core/mem/pkg.c:91]: pkg_destroy_manager(): destroying memory manager: q_malloc In mongoDB, I have the collection "version" in the db kamailio, and the entries needed for usrloc (location and location_attrs): sre_location:PRIMARY> use kamailio switched to db kamailio sre_location:PRIMARY> db.getCollection("version").find() { "_id" : ObjectId("5a8ec43c601b262bef82a595"), "table_name" : "location", "table_version" : 9 } { "_id" : ObjectId("5a8edb72a3ec40427577b8d0"), "table_name" : "location_attrs", "table_version" : 1 } And logs in mongo show the following info when kamailio starts: 2018-02-22T17:13:02.604+0100 I NETWORK [conn90] received client metadata from 10.0.12.180:56320 conn: { driver: { name: "mongoc", version: "1.9.2" }, os: { type: "Linux", name: "CentOS", version: "6.9 (Final)", architecture: "x86_64" }, platform: "cfg=0x1563000 posix=200809 CC=GCC 4.4.7 20120313 (Red Hat 4.4.7-18) CFLAGS="" LDFLAGS=""" } 2018-02-22T17:13:02.604+0100 I COMMAND [conn90] command admin.$cmd command: isMaster { isMaster: 1, client: { driver: { name: "mongoc", version: "1.9.2" }, os: { type: "Linux", name: "CentOS", version: "6.9 (Final)", architecture: "x86_64" }, platform: "cfg=0x1563000 posix=200809 CC=GCC 4.4.7 20120313 (Red Hat 4.4.7-18) CFLAGS="" LDFLAGS=""" }, compression: [], $readPreference: { mode: "secondaryPreferred" }, $db: "admin" } numYields:0 reslen:716 locks:{} protocol:op_query 0ms 2018-02-22T17:13:02.604+0100 I COMMAND [conn90] command kamailio.$cmd command: find { find: "version", filter: { table_name: "location" }, table_version: 1, $db: "kamailio", $readPreference: { mode: "primaryPreferred" }, $clusterTime: { clusterTime: Timestamp(1519315979, 1), signature: { hash: BinData(0, 0000000000000000000000000000000000000000), keyId: 0 } } } numYields:0 reslen:531 locks:{} protocol:op_msg 0ms 2018-02-22T17:13:02.605+0100 I NETWORK [conn90] end connection 10.0.12.180:56320 (9 connections now open) I probably miss something: it is not clear to me why kamailio is not able to retrieve the value of table_version for the table_name location. Any idea would be very welcome ! I use mongo-c-driver 1.9.2 and mongodb 3.6 Thank you. Stephane

3 14

permissions.allow in conjunction with trusted table
by Ben Hood 12 Mar '18

12 Mar '18

Hi, I'd like to trust HTTP requests from internal subnets and use the trusted table to manage permissions from external networks. Is there a way to configure permissions.allow to trust everything on a local subnet and use the DB table for everything else? Does the permissions module even work in that way (i.e. to use two sources of trust)? If so, what is the permissions.allow syntax to trust a CIDR? Or is it better to split the trust up based on listening ports (i.e. define the route blocks so that requests to the port listening for internal traffic is not subject to the trusted table)? I guess I could also add the trusted internal subnet to the trusted table, but this would be mixing application level stuff with devops stuff. TIA, Ben

2 2

Tracking NAT Helper keepalives
by Mark Boyce 12 Mar '18

12 Mar '18

Hi All Is there any way to extract round trip lime state from keepalives? I have a customer who’d like to duplicate the qualify feature of Asterisk. ie latency / distance of endpoint from server. Can’t see any way to hook in from the docs. Thanks Mark

2 1

SBC rejects SIP INVITE from Kamailio+topos
by José Lopes 12 Mar '18

12 Mar '18

Hello, I am using Kamailio working as edge proxy with topos module active. Kamailio is proxing requests from Freeswitch's to the SBC Metaswitch. The Freeswitch sends SIP REGISTER requests and it is ok (Contact: < sip:1003@172.29.210.164:5070>). When Freeswitch sends SIP INVITE quest, it receives a reply with Forbiden (Contact: <sip:btpsh-5aa0219a-2a32-2@172.29.210.164:5070>). You can check the sip trace at https://pastebin.com/raw/JZUNgYx5 I analysed this issue on SBC Metaswitch and it requires that the INVITE and REGISTER have the same Contact Header. I know that this is not required from SIP RFC, but that it is SBC default security policy. There is any way to configure Kamailio using topos to have the same Contact Header on INVITE and REGISTER? Best regards, José Lopes

2 1

Dialog Terminator
by Leonardo Assunção 12 Mar '18

12 Mar '18

Greetings, Does Kamailio has a method or variable that indicates who cancelled,rejected or terminated a call and, therefore ended the dialog? Currently i'm looking for the sources of BYE's, CANCEL and replies greater than 300 to first INVITE, but i would safer if such a method existed. Thanks in advance

2 1

Presence not notifying?
by Mark Boyce 11 Mar '18

11 Mar '18

Hi All Got a presence problem with what should be a very simple config. Would much appreciate someone having a look at the following and seeing if I’ve missed something obvious :-) Thanks Mark I’ve gone right back to basics and using the example at http://kb.asipto.com/kamailio:presence:k43-blf <http://kb.asipto.com/kamailio:presence:k43-blf> Changes to that example are; - Kamailio 5.1.2 - RTPEngine - NAT Enabled - JSON RPC replacing mi_rpc - HA1 Passwords Setup is; Three phone 20,21,22 all on site behind a NAT (10.1.3.x local IP) at 1.2.3.4 External IP. Phone 21 has BLF Subscriptions on 20 and 22 Domain is set as “mykamailio.com” Kamailio is an Ubuntu server “In the cloud” on 5.6.7.8 (Assuming I have got my domain & ip rewrites correct on the cut’n’paste) What’s Happening; All three phones register 21 Issues SUBSCRIBE:Dialog for 20 and 22. Which get 202 Accepted back after a 407 Proxy Auth SUBSCRIBE sip:22@mykamailio.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.3.84:5065;rport;branch=z9hG4bK3894730624 From: "21@SBC-2" <sip:21@mykamailio.com>;tag=1678729960 To: <sip:22@mykamailio.com> Call-ID: 1544456221(a)10.1.3.84 CSeq: 1 SUBSCRIBE Contact: <sip:21@10.1.3.84:5065> Accept: application/dialog-info+xml Max-Forwards: 70 User-Agent: Yealink SIP-T26P 6.73.0.50 Expires: 1800 Event: dialog Content-Length: 0 I can see these arrive in active_watchers table; *************************** 2. row *************************** id: 2 presentity_uri: sip:20@mykamailio.com watcher_username: 21 watcher_domain: mykamailio.com to_user: 20 to_domain: mykamailio.com event: dialog event_id: to_tag: a6c744f901d779619c580c9ef48ab5e9-2651 from_tag: 1809677495 callid: 2674059037(a)10.1.3.84 local_cseq: 0 remote_cseq: 2 contact: sip:21@10.1.3.84:5065;alias=1.2.3.4~5065~1 record_route: expires: 1520711354 status: 1 reason: version: 0 socket_info: udp:5.6.7.8:5060 local_contact: sip:5.6.7.8:5060 from_user: 21 from_domain: mykamailio.com updated: -1 updated_winfo: -1 flags: 0 user_agent: Yealink SIP-T26P 6.73.0.50 If I make a call from 20 to 22 it works as expected but no notify is sent. For completeness the changes I’ve made to the kamailio.cfg file are; # diff kamailio.cfg_presence_example kamailio.cfg 6a7 > #!define WITH_NAT 119c120 < #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" --- > #!define DBURL "mysql://kamailiorw:password@localhost/test_kamailio" 169a171 > alias="mykamailio.com" 221c223 < mpath="/usr/local/lib/kamailio/modules/" --- > mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules/" 228c230 < loadmodule "mi_fifo.so" --- > loadmodule "jsonrpcs.so" 245d246 < loadmodule "mi_rpc.so" 279c280 < loadmodule "rtpproxy.so" --- > loadmodule "rtpengine.so" 304,307d304 < # ----- mi_fifo params ----- < modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") < < 375,376c372,374 < modparam("auth_db", "calculate_ha1", yes) < modparam("auth_db", "password_column", "password") --- > modparam("auth_db", "calculate_ha1", no) > modparam("auth_db", "password_column", "ha1") > modparam("auth_db", "password_column_2", "ha1b") 414c412 < modparam("presence", "server_address", "sip:192.168.178.54:5060" ) --- > modparam("presence", "server_address”, "sip:5.6.7.8:5060") 454c452 < modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") --- > modparam("rtpengine", "rtpengine_sock", "udp:127.0.0.1:2223") 835c833 < rtpproxy_manage("co"); --- > rtpengine_manage("co”);

2 5

Re: [SR-Users] From URI == Myself
by Fred Posner 09 Mar '18

09 Mar '18

If you're not listening or identifying that IP and specific port as an alias, then yes, this is expected. One reason why would be to run another program on that port. Fred Posner direct: +1 (224) 334-FRED (3733) > On Mar 9, 2018, at 6:52 AM, Nicolas Breuer <Nicolas.Breuer(a)belcenter.biz> wrote: > > --_000_VI1PR0402MB359898CF4C2CDFCD4EBA5DA189DE0VI1PR0402MB3598_ > Content-Type: text/plain; charset="us-ascii" > Content-Transfer-Encoding: quoted-printable > > Hello the List, > > I saw a strange behavior on the last Kamailio v4. > > The Kamailio is binding on port 5060. > > If (from_uri=3D=3Dmyself){ > ...... > } > > > * When the From uri in the INVITE is <xxxx@myalias:5070> > * When the myalias is an alias > No issue the IF routine is executed. > * When the myalias is the ip address of the server > The IF Routine is = > NOT executed because there is a mismatch in the port (5060 >< 5070) > > > Is it the expected behavior ? > > Thanks > > > > > > > --_000_VI1PR0402MB359898CF4C2CDFCD4EBA5DA189DE0VI1PR0402MB3598_ > Content-Type: text/html; charset="us-ascii" > Con

1 0

From URI == Myself
by Nicolas Breuer 09 Mar '18

09 Mar '18

Hello the List, I saw a strange behavior on the last Kamailio v4. The Kamailio is binding on port 5060. If (from_uri==myself){ ...... } * When the From uri in the INVITE is <xxxx@myalias:5070> * When the myalias is an alias > No issue the IF routine is executed. * When the myalias is the ip address of the server > The IF Routine is NOT executed because there is a mismatch in the port (5060 >< 5070) Is it the expected behavior ? Thanks

1 0

ways to track all messages of a dialog
by Fabian Borot 09 Mar '18

09 Mar '18

I need to track all messages (SIP Requests, SIP Responses) for some SIP dialogs. Let's say I need to log something specific on each state for calls initiated from a certain IP (initial request, temporary replies, rejection with a negative response, call was accepted with 200, termination with BYE, termination with CANCEL etc. ). I am thinking on using flags but I am not sure after reading the tutorial on flags (http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-flag-operations) if that is going to work properly. The doc says: "They provide a very easy and fast way of keeping states during processing a request or during a transaction". I am afraid that the ACK if the call is accepted and the BYE won't be tracked properly because they are not part of the initial transaction. Also the doc does not mention if the flag is preserved among branches spawned during the processing of the call. The idea is for example: request_route { if (is_method("INVITE")){ if ($si == "192.168.168.100"){ #condition that would set the flag, INVITE came from "my_IP" #flag/mark dialog setflag(1); #log call from "my_IP" was received } route("INCOMING); } If(loose_route() ){ if (is_method("ACK")){ if(isflagset(1)){ #log ACK related to "my_IP" was received } } if (is_method("CANCEL")){ if(isflagset(1)){ #log CANCEL related to "my_IP" was received } } if (is_method("BYE")){ if(isflagset(1)){ #log BYE related to "my_IP" was received } } } } Then : route["INCOMING"] { # do some processing # do some processing t_onreply("INCOMING"); t_on_failure("INCOMING"); t_relay(); } onreply_route[INCOMING] { #do some processing if(isflagset(1)){ #log a reply related to "my_IP" was received } } failure_route[INCOMING] { #do some processing if(isflagset(1)){ #log a negative reply related to "my_IP" was received } } Will the flags work on this case ? My 2nd approach is to use a htable and use the call_id for my calls as the key. The on each route section (reply, failure_reply, ACK processing etc.) find the $ci on the htable and if it is there do the appropriate action. But I need to delete the key:value once the call ends or leave it autoexpire with a expire value bigger than the max_call_duration_time that we have in our system.

3 6

sql_pvquery free result
by Ben Hood 09 Mar '18

09 Mar '18

Hi, Does sql_pvquery() clean up after itself? The other query APIs return a result handle that you clean up with sql_result_free(), but sql_pvquery() doesn’t appear to return any handle, so is it self-collecting? Cheers, Ben

2 2

Why does all the examples on configuring Kamailio install PostgreSQL
by Wilkins, Steve 09 Mar '18

09 Mar '18

Hello All, I have noticed that almost all, if not all examples of installing Kamailio install the PostgreSQL libraries. If one is using MySQL, is there a need for this? Is there something in Kamailio that requires these PostgreSQL libraries to be installed? Thank you

4 3

Resend Authenticate REGISTER to two or more Asterisk PBX
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'd like know which is the best way to resend a Authenticate REGISTER to two or more Asterisk. Repeating rewritehostport and forward commands for the number of Asterisk Servers, there is a problem with the Via Header; Kamailio add one with his IP for each forward. I try to remove the first Via before send the REGISTER to the second Asterisk without success. Any hint or tip? Thank you Regards -- --- I'm SoCIaL, MayBe

2 2

how to check if pseudo-variable contains a String
by Laurent Schweizer 08 Mar '18

08 Mar '18

Dear all, I have a question regarding pseudo-variable, how can I check if a pseudo-variable contains a specific string ? Like to check if $ct contains "transport=tcp". Thanks BR Laurent

1 0

Info: Meeting during FOSSASIA 2018
by Daniel-Constantin Mierla 08 Mar '18

08 Mar '18

Hello, soon I am going to travel to Singapore for FOSSASIA 2018 conference (March 22-25 - https://2018.fossasia.org/), giving a talk about Kamailio. Given that it is rather rare I travel to south-east Asia, I wonder if there are other people from Kamailio community attending FOSASSIA or happen to be in the city during the event. Maybe we can plan something like a small group meetup to greet, have a drink or so and chat about the project and real time communications. Write me if you can join such a meetup. It would be great to meet some kamailians around there! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Kamailio Memory Settings
by Duarte Rocha 08 Mar '18

08 Mar '18

Hey. I have some questions about memory settings. I can see the default values with "kamailio -I" and change the current values with "kamailio -M 12 -m 128". How do i see the active settings and how do i change the default value? I want to setup a proxy capable to handle 150CPS, 6k active calls, with Dispatcher, Database, Number Normalizations and other operations. What are good settings for CPU, RAM Memory, PKG_SIZE and SHM_SIZE? Thanks for the help, Best Regards, Duarte Rocha

2 1

Failed auth
by Aidar Kamalov 08 Mar '18

08 Mar '18

Hello! I have kamailio 5.0.5 and some cisco spa 50x phones. Sometime phone can't register to kamailio. REGISTER packets with right nonce,reponse, etc gets to kamailio but it response 401 Unauthorized with code -3 (user not found). After some attempts(or time, phone send register every 1200 seconds) or after reboot phone it register succesfully Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -5) fd=voip.xxxx.org, adu=<null>, aa=<null>, ar=<null>, au=<null>, ad=<null>, aU=<null>, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] -- Aydar A. Kamalov

2 1

Issue starting with kamctl
by Ruben Lezama 08 Mar '18

08 Mar '18

Hi, I am having issues to start Kamailio using kamctl. root@kamailio2:/etc/systemd/system# kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' Using #Kamailio starts works. I can register peers and make calls between them. But I cannot update configuration. root@kamailio2:/var/log# kamailio Listening on udp: 127.0.0.1:5060 udp: 192.168.1.100:5060 tcp: 127.0.0.1:5060 tcp: 192.168.1.100:5060 Aliases: tcp: kamailio2:5060 tcp: localhost:5060 udp: kamailio2:5060 udp: localhost:5060 I would appreciate any help provided. Attached the syslog output Ruben

2 1

KeepAlive Module
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'm using keepalive module with drouting module... the event route works fine: NOTICE: <script>: Gateway up: OPTIONS sip:1.2.3.4 NOTICE: <script>: Gateway up: OPTIONS sip:5.6.7.8 but kamcmd command not: *kamcmd keepalive.list* foobar 42 foobar 42 { text: foobar number: 42 } { list: { nn: 17 nn: 22 } } { uri: sip:1.2.3.4 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:13 2018* } { uri: sip:5.6.7.8 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:12 2018* Thank you Regards... -- --- I'm SoCIaL, MayBe

2 1

Strongswan on same Server in EC2 as kamailio 4.4
by Jonathan Hunter 08 Mar '18

08 Mar '18

Hi Guys, I have an issue with sending a BYE message back down a strongswan VPN tunnel on a server in the Amazon EC2 environment. To get the configuration working correctly I have the Public IP address bound to a loopback virtual interface, and I also have kamailio listening to it. Calls establish without issue, but when the called party send a BYE, I cant seem to force the SIP message down the correct interface. Due to client confidentiality I need to be careful what information I send but does anyone have a similar configuration set up, and working, I am just looking for reasons as to why force_socket and mhomed=1 doesnt appear to force the BYE message via the Public Interface but the private IP address on the server as its a NAT'd environment. So essentially call establishes; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B | | | | |<---------(VIA IPSEC)-100 TRYING----|<----------100 TRYING---------------|<----------100 TRYING----------------------------| | | | | |<---------(VIA IPSEC)-180 RINGING---|<----------180 RINGING--------------|<----------180 RINGING---------------------------| | | | | |<---------(VIA IPSEC)-200OK --------|<----------200OK -------------------|<----------200OK --------------------------------| | | | | |----------(VIA IPSEC)-ACk---------> |-----------ACk--------------------->|-----------ACk---------------------------------->| How can I ensure the BYE is forced out via the PUBLIC IP? So I do this; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B |<---------BYE (VIA IPSEC)----------------------|<----------BYE ----------------------------|<----------BYE -------------------| As it appears to be going from Private IP and not being pushed down the IPSEC tunnel CLIENT---------(VIA IPSEC)-INVITE----->KAMAILIO PUBLIC IP-------->KAMAILIO PRIVATE IP------------------------->CLIENT B |<---------BYE ----------------------<----------BYE ----------------------------|<----------BYE -------------------| on the BYE I have tried; force_send_socket(KAMAILIO PUBLIC IP); However doesnt appear to work, is this an issue for ip routing on the server to push the BYE correctly down the IPSEC tunnel? or is it configuration related? Any comments welcome if people have had this issue before. Many thanks Jon

2 1

Question about t_branch_timeout behavior.
by José Seabra 08 Mar '18

08 Mar '18

Hi there, I have some situations where Kamailio receives replies (100 and then 180) from the other end and after 2 seconds (timer defined on function t_set_fr(0,2000)) after receive the last reply(180) the INVITE transaction enters on the failure route with variable $T_reply_code=408. What can be wrong here to kamailio generates a 408 internally after receive 1xx replies? Best Regards José Seabra

2 1

SIP header when 911 called
by Abdul Basit 08 Mar '18

08 Mar '18

Hi Team, Does kamailio has compatibility with E911, i.e. through location in SIP header when 911 called? Is there any reference document? -- regards, abdul basit

2 1

Choosing a Testing and Validation Tool
by Duarte Rocha 08 Mar '18

08 Mar '18

Hey I need to create several testing and validation processes for the deploy of an SBC and i would like to know your opinion on the best tools to use. I need to create an automated group of testing scenarios and validation of the packets sanity and SIP compliance. I also would like to validate the packets according to some parameters of mine (ex. All numbers in E164, etc). I'm looking at Star Trinity's SIP Tester right now but i'm not sure it can fulfill all my needs. I've also looked into SIPp and considered creating a Kamailio to do the validation, but i'm not sure it's worth the work if i can find a tool prepared to do this. Cheers :)

2 1

Disable non-existent user warning
by Stefan Rüetschli 08 Mar '18

08 Mar '18

Hello sr-users! I'm using Kamailio 5.1.2. If I execute the command "serctl acl grant 1234 mygrant" I get a warning which I have to accept with "Y": Non-existent user '1234'. Still proceeding? [Y|N] Is it possible to deactivate this warning? I know why this warning is there (it's not a user, it's an alias), so I think I know what I'm doing ;-) Regards Stefan

3 4

Kamailio not send NOTIFY for the state of the subscription when using Event: dialog (BLF)
by Borislav Gergov 08 Mar '18

08 Mar '18

Hello, The initial NOTIFY that shows the server accepts our subscription is not sent: SUBSCRIBE sip:002@10.7.2.22;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---837a95d50bef81d4;rport. Max-Forwards: 70. Contact: <sip:444@10.2.4.38:63718;transport=TCP>. To: <sip:002@10.7.2.22;transport=TCP>. From: <sip:444@10.7.2.22;transport=TCP>;tag=810bbd39. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 1 SUBSCRIBE. Expires: 120. Accept: application/dialog-info+xml. Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE. User-Agent: Z. Event: dialog. Allow-Events: presence, kpml, talk. Content-Length: 0. . SIP/2.0 202 OK. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---837a95d50bef81d4;rport=63718;received=10.2.4.38. To: <sip:002@10.7.2.22 ;transport=TCP>;tag=339c21513a585618341888f199c2a0f6-c10d. From: <sip:444@10.7.2.22;transport=TCP>;tag=810bbd39. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 1 SUBSCRIBE. Expires: 120. Contact: <sip:10.7.2.22:5060;transport=tcp>. Server: kamailio (5.1.1 (x86_64/linux)). Content-Length: 0. . SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.2.4.38:63718 ;branch=z9hG4bK-524287-1---5b9c08901116e843;rport=63718;received=10.2.4.38. To: <sip:444@10.7.2.22 ;transport=TCP>;tag=75017b5d51cabe3de4b519c896167ee7.88a9. From: <sip:444@10.7.2.22;transport=TCP>;tag=d831f844. Call-ID: p-vce2LWcasOPThxdFZP6A... CSeq: 3 REGISTER. Contact: <sip:444@10.2.4.38:63718 ;transport=TCP;rinstance=045d4a61cf8385dc>;expires=120. Server: kamailio (5.1.1 (x86_64/linux)). Content-Length: 0. . NOTIFY sip:444@10.2.4.38:63718;transport=TCP SIP/2.0. Via: SIP/2.0/TCP 10.7.2.22;branch=z9hG4bK94d5.046c6940000000000000000000000000.0. To: <sip:444@10.7.2.22>;tag=810bbd39. From: <sip:002@10.7.2.22>;tag=339c21513a585618341888f199c2a0f6-c10d. CSeq: 2 NOTIFY. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... Content-Length: 0. User-Agent: kamailio (5.1.1 (x86_64/linux)). Max-Forwards: 70. Event: dialog. Contact: <sip:10.7.2.22:5060;transport=tcp>. Subscription-State: terminated;reason=timeout. . SIP/2.0 200 OK. Via: SIP/2.0/TCP 10.7.2.22;branch=z9hG4bK94d5.046c6940000000000000000000000000.0. Contact: <sip:444@10.2.4.38:63718;transport=TCP>. To: <sip:444@10.7.2.22>;tag=810bbd39. From: <sip:002@10.7.2.22>;tag=339c21513a585618341888f199c2a0f6-c10d. Call-ID: 8fMhJ4GnMrkSu8qxez70aw... CSeq: 2 NOTIFY. User-Agent: Z. Content-Length: 0.

2 1

consume_credentials question
by Aymeric Moizard 07 Mar '18

07 Mar '18

Hi list! I'm trying to remove Proxy-Authorization dedicated to local subscriber without -and not the other ones dedicated to other server- I realized that "consume_credentials" doesn't work because it requires that a call to "www_authorize, proxy_authorize, www_authenticate or proxy_authenticate" has been made before. I have a use-case where this isn't called. I don't want to use such code: if (is_present_hf("Proxy-Authorization")) { remove_hf("Proxy-Authorization"); } Because this may remove a "Proxy-Authorization" for another service. I have seen this: has_credentials but it doesn't help to know which header is the one containing the header... Any good way to find the header and index? And then remove it with remove _hf_value2? Regards Aymeric -- Antisip - http://www.antisip.com

2 2

SIP RFC behaviour for 180/200 to_tag mismatch
by Mititelu Stefan 07 Mar '18

07 Mar '18

Hi people, I am trying to search for the RFC behavior in case of 180/200 to tag mismatch of an INVITE. From my readings so far, I'm not 100% convinced of the correct behavior. However, there may be a scenario, where this might be plausible: 1. parallel forked two calls 2. 180 ringing comes for first one, but not for second one(various reasons) 3. 200 ok comes for second one Any opinions about this? Do you have some references that will make this 100% clear? Thank you, Stefan

2 2

sqlops module error
by Fabian Borot 06 Mar '18

06 Mar '18

I am using the sqlops module with odbc to INSERT,UPDATE only and as the documentation says I can omit result parameter in "sql_query". However I always see these ERRORS when processing the call: This is an example of a query: sql_query("c04", "insert into SIPInfo302CDR (SIPCallID,SessionID) values('$ci',$var(SessionID))"); And these are the error lines in the log: ERROR: db_unixodbc [res.c:60]: db_unixodbc_get_columns(): no columns returned from the query ERROR: db_unixodbc [res.c:261]: db_unixodbc_convert_result(): getting column names failed ERROR: db_unixodbc [dbase.c:216]: db_unixodbc_store_result(): failed to convert result ERROR: <core> [db_query.c:188]: db_do_raw_query(): error while storing result ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [insert into SIPInfo302CDR (SIPCa] I have also tried using the parameter like shown below but the ERROR events happen anyways. sql_query("c04", "insert into SIPInfo302CDR (SIPCallID,SessionID) values('$ci',$var(SessionID))", "ra"); sql_result_free("ra"); Even though the errors line show up and say that " cannot do the query .." the query does work and the records are inserted/updated accordingly. I tried using the "sql_query_async" function that does not need the result parameter but I can't use it with unixODBC, I see these lines in the log when trying to use it: "the db driver module doesn't support async query" and the record is not inserted/updated. I also added the "async_workers" core parameter as well just in case. I also tried declaring them in the module sqlops initialization section even though in my case there is no need: "modparam("sqlops", "sqlres", "ra")" but the errors show up anyways. When I test with SEELCT query the result parameter can be used, I can count rows etc. But I am only doing INSERT/UPDATE. Am I am missing something here or this is something that can be safely used like that and I just need to ignore them. I am using " Kamailio (OpenSER) SIP Server v5.0" Thank you Fabian

2 1

Re: [SR-Users] cnxcc module - make auth for redis connection
by Donat Zenichev 06 Mar '18

06 Mar '18

Hi Daniel, first of all thanks for your attention. Perhaps I missed something. I read documentation about cnxcc and ndb_redis again, but I haven't found information that can help me with my goal. ndb_redis is separate module that implements connections to redis databases (and also is able to perform authentication), has it's own functions like redis_cmd() to interact directly with redis. But cnxcc stores real-time amounts in redis hash keys using it's own parameter - modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") And I can't see possibility to use ndb_redis to perform authentication for cnxcc module, correct me if I'm wrong. Do you offer me to read ndb_redis code sources to use them to edit cnxcc code sources? Thanks in advance for your advice. 2018-03-01 9:28 GMT+02:00 Donat Zenichev <donat.zenichev(a)gmail.com>: > Hi Daniel, first of all thanks for your attention. > > Perhaps I missed something. I read documentation about cnxcc and ndb_redis > again, but I haven't found information that can help me with my goal. > ndb_redis is separate module that implements connections to redis > databases (and also is able to perform authentication), has it's own > functions like redis_cmd() to interact directly with redis. > > But cnxcc stores real-time amounts in redis hash keys using it's own > parameter - modparam("cnxcc", "redis", "addr=127.0.0.1;port=6379;db=1") > And I can't see possibility to use ndb_redis to perform authentication for > cnxcc module, correct me if I'm wrong. > > Do you offer me to read ndb_redis code sources to use them to edit cnxcc > code sources? > Thanks in advance for your advice. > > > > 2018-02-28 14:33 GMT+02:00 Daniel-Constantin Mierla <miconda(a)gmail.com>: > >> Hello, >> >> maybe authentication support was not implemented in cnxcc. Eventually it >> can be extracted from the ndb_redis, that module has it implemented already. >> >> * https://www.kamailio.org/docs/modules/stable/modules/ndb_red >> is.html#ndb_redis.p.server >> >> Cheers, >> Daniel >> >> On 27.02.18 14:39, Donat Zenichev wrote: >> >> Nevertheless, I want to find the needed solution. >> In my opinion - If module is able to connect to redis databases, it'd >> better to perform it with authentication. >> >> Because, there are cases when db located on localhost is improper >> solution or other users mustn't see hkeys, amounts inside hkeys it etc. >> So I think no one will be mind, to have whole possibility - real-time >> billing (cnxcc) working with redis securely. >> >> So please don't be apathetic to my request. >> >> 2018-02-26 9:44 GMT+02:00 Donat Zenichev <donat.zenichev(a)gmail.com>: >> >>> Hi community. >>> This message is specially dedicated to developers of the kamailio >>> project. >>> >>> I need to find solution how to make authentication for connections of >>> cnxcc with redis. >>> For that moment, cnxcc module is able to connect to redis only without >>> password (Carlos Ruiz Díaz will add this possibility later?). >>> So that, this makes database vulnerable. >>> >>> Well, furthermore I don't want to use local redis database, cuz I think >>> machine that contains kamailio daemon, must perform just this kind of >>> actions - SIP proxying (kamailio server is not to be a database server or >>> anything else). >>> >>> For that moment we use separate redis db for cases like that, but this >>> is not best practice I think. >>> I already tried to find solution (for e.g. connections via another >>> module like db_redis: >>> https://github.com/kamailio/kamailio/pull/1432) but I got advice - >>> achieving my goal by sr-users mailing list. >>> >>> Please pay attention for my request. I think this will be very helpfull >>> for others. >>> >>> >>> >>> -- >>> -- >>> BR, Donat Zenichev >>> Wnet VoIP team >>> Tel Ukraine: +380(44) 5-900-800 >>> Tel USA: +164(67) 8-174-17 (ext 1320) >>> https://w-net.us/ <http://wnet.ua> >>> >> >> >> >> -- >> -- >> BR, Donat Zenichev >> Wnet VoIP team >> Tel Ukraine: +380(44) 5-900-800 >> Tel USA: +164(67) 8-174-17 (ext 1320) >> https://w-net.us/ <http://wnet.ua> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com >> >> > > > -- > -- > BR, Donat Zenichev > Wnet VoIP team > Tel Ukraine: +380(44) 5-900-800 > Tel USA: +164(67) 8-174-17 (ext 1320) > https://w-net.us/ <http://wnet.ua> > -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 (ext 1320) https://w-net.us/ <http://wnet.ua>

2 1

Setup timeout for in function jsonrpc_request
by Marco Capetta 06 Mar '18

06 Mar '18

Hi All, I'm currently in the process of upgrading a kamailio instance from 4.4.6 to 5.1.2. Since"janssonrpc-c" has been renamed in "jsonrpcc" I have to adapt my configuration. In version 4.4.6, the function "janssonrpc_request" allowed me to setup a specific timeout for the request. Is it possible to do the same in version 5.1.2 with the function "jsonrpc_request"? Because the documentation doesn't mention it. Thank you Best regards Marco

2 2

Re: [SR-Users] Not documented parameter server_id_filter inside usrloc module
by Konstantin Polyakov 05 Mar '18

05 Mar '18

Victor, We are going to use this parameter, but we found that cleanup process should use such filter too. Otherwise different instances of proxy delete records of each other and it leads to the problems when db_mode 1 or 2 (cached) is used. Soon I am going to propose a pull request with filter in cleanup procedure. Could you tell me please, if it is necessary to add a new parameter for cleanup process or I can use this server_id_filter? Is there some case when we need to use this filter on startup but we don't it during cleanup? Thank you. Best regards, Konstantin

2 1

Not documented parameter server_id_filter inside usrloc module
by Konstantin Polyakov 05 Mar '18

05 Mar '18

Hello everybody, Could somebody clarify why server_id_filter parameter is not listed in the usrloc module article? It is not ready for usage? Or it was forgotten to be added into description? Thank you Best regrads Konstantin

2 1

Error using newest Siremis and kamailio 5.0.3
by Olli Attila 05 Mar '18

05 Mar '18

Hello, I installed Siremis today from master branch alongside with Kamailio 5.0.3 and entering "XMLRPC Commands" menu gives me this error: [2018-03-01 12:28:34 (GMT)] An exception occurred while executing this script: Error message: #256, Unable to location template file system_internal_error.tpl. Script name and line number of error: /var/www/html/https/siremis/openbiz/bin/Resource.php:283 function: errorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/html/https/siremis/openbiz/bin/Resource.php", 283, Array(11) ) @ /var/www/html/https/siremis/openbiz/bin/sysheader.inc 117 function: userErrorHandler ( 256, "Unable to location template file system_internal_error.tpl.", "/var/www/html/https/siremis/openbiz/bin/Resource.php", 283, Array(11) ) @ function: trigger_error ( "Unable to location template file system_internal_error.tpl.", 256 ) @ /var/www/html/https/siremis/openbiz/bin/Resource.php 283 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/html/https/siremis/openbiz/bin/BizSystem.php 560 function: getTplFileWithPath ( "system_internal_error.tpl", "email" ) @ /var/www/html/https/siremis/siremis/modules/service/userEmailService.php 108 function: SystemInternalErrorEmail ( Array(2), Null ) @ /var/www/html/https/siremis/siremis/modules/common/form/ErrorForm.php 34 function: Report ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 310 function: invoke ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 110 function: dispatchRequest ( ) @ /var/www/html/https/siremis/openbiz/bin/BizController.php 32 function: include_once ( "/var/www/html/https/siremis/openbiz/bin/BizController.php" ) @ /var/www/html/https/siremis/siremis/bin/controller.php 6 (I'm running PHP 7.0.27-0+deb9u1 and MariaDb Server version: 5.1.73 Source distribution) What is going wrong here? I found this online but I think this is already fixed on the Siremis master branch? https://lists.kamailio.org/pipermail/sr-users/2016-December/095373.html Cheers, Olli

2 7

Invoking RPC commands over HTTP
by Ben Hood 05 Mar '18

05 Mar '18

Hi, What is the idiomatic way to call an RPC from a remote HTTP client in the 5.1.x series? I'd like to invoke mtree.reload from a remote HTTP client. My goal is to avoid a dependency on kamcmd on the remote client system - it would be nice to be able to issue a HTTP RPC which could be implemented in any environment. Or should I be looking to implement a BINRPC client on the remote system? TIA, Ben

3 8

Kamailio iOS Push Notifications
by Amine Mensori 04 Mar '18

04 Mar '18

Hello, We need help implementing push notifications within Kamailio. I have found a module relating to this: https://github.com/tvntsr/push <https://github.com/tvntsr/push> However, the developer did not provide any examples of the cfg file, only parts of certain functions. We need some help implementing the functions in our config file or a full example of another method to get push notifications to work when the application is closed on the device. Please note we are not very proficient in C programming so a clear example would help a lot. Many Thanks, Amine

4 3

Non-standard SIP-over-TLS port
by Steve 03 Mar '18

03 Mar '18

Hello sr-users, How does one assign a non-standard SIP-over-TLS port rather than the usual port 5061 for client registration? Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

2 2

dialplan module reload fail
by Paulo Ferreira 02 Mar '18

02 Mar '18

Hi list, I'm using several modules, including dialplan. I can reload modules data from database to kamailio with kamctl reload, except the dialplan module. kamctl dialplan reload { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" }, "id": 21705 } looking at the error, it seems that this method was not implemented. This isn't making sense, so any idea in what can be wrong? thanks,and regards, Paulo -- ------------------------------------------- Paulo Ferreira VoIP@RCTS - Área de Infraestruturas Aplicacionaisz FCCN http://www.fccn.pt/ Av. do Brasil, n.º 101 1700-066 Lisboa - Portugal Telefone|Phone: +351 218440100; Fax: +351 218472167 Aviso de Confidencialidade/Disclaimer Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha rececionado indevidamente esta mensagem,solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218440100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218440100 and delete it immediately.

2 2

question
by Syrine Riahi 02 Mar '18

02 Mar '18

bonjour je suis une étudiante en mastère professionnel des systèmes des réseaux et télécommunication j'ai choisie comme un sujet de mémoire mise en oeuvre de solution des VoIP et après des grands recherches j'ai choisie comme solution kamailio alors comme je suis débutante dans le domaine j'ai besoin des étapes exacte de l'instalation et la configuration de kamailio v 5.1.2 et j'utilise la distribution ubuntu v 16.04 merci énormément de me voir aider

3 2

5.1.2 rtpengine regressions - broke working setup?
by Anthony Alba 02 Mar '18

02 Mar '18

Hi list, upgrade to 5.1.2 broke my setup due to some different behaviour in rtpengine.so. I use: rtpengine_manage("DTLS=off ICE=remove"); 1. I don't send any rtpengine_manage flags (RTP UDP SAVP). UAC sends RTP/SAVP in SDP, rtpengine changes to UDP/TLS/RTP/SAVP. Reverted to 5.1.1, and see that rtpengine sends RTP/SAVP 2. To SDP body, rtpengine is adding: a=setup:actpass a=fingerprint:sha-1 D6:D7:DA:AA:2D:DB:AD:94:29:46:FA:C4:D8:AE:0F:03:96:6F:8A:4F For some reason this broke FreeSWITCH which does not like this. Anthony

2 6

Re: [SR-Users] rtpengine transcoding benchmarks?
by Jeff Brower 01 Mar '18

01 Mar '18

Andreas- > We've seen around 400 concurrent calls for G711<->Opus on a rather > slow 12-core E5-2420(a)1.90GHz for some first preliminary benchmarks. Was that using narrowband (8 kHz) for Opus ? Otherwise sampling rate conversion is needed. Also I assume Opus DTX was disabled. Also at what Opus bitrate ? Thanks. -Jeff > On 2018-02-26 21:27, Brooks Bridges wrote: >> I realize the transcoding support is rather new, however has anyone run any real world tests on it yet?Â Trying to get a sense of what the performance will be. >> >> Â >> >> Iâm aware that itâs forced into userspace for transcoding so itâs not **nearly** as efficient as the iptables module, however even if I have to throw a rack full of whiteboxes at it to get anywhere close to decent numbers thatâs still going to be comparable to a lot of the commercial options out there right now. >> >> Â >> >> *Brooks Bridges | *Engineering Manager >> >> *O^1 Communications* >> >> 4359 Town Center Boulevard, Suite 217 >> >> El Dorado Hills, California 95762 >> >> *office:*916.235.2097 |*main:*888.444.1111, Option 2 >> >> *email:*bbridges@o1.com <mailto:bbridges@o1.com> |Â *web:*www.o1.com <http://www.o1.com/> >> >> Â

2 2

Kamailio v5.1.2 Released
by Daniel-Constantin Mierla 01 Mar '18

01 Mar '18

Hello, Kamailio SIP Server v5.1.2 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.1. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.2. For more details about version 5.1.2 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/03/kamailio-v5-1-2-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.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Planing next Kamailio releases - v4.4.7 v5.0.6 and v5.1.2
by Daniel-Constantin Mierla 01 Mar '18

01 Mar '18

Hello, I am considering to do releases from last three stable branches during next week. v4.4.7 is going to be released out of latest code in branch 4.4 to mark the end of official maintenance for series v4.4.x. The version 4.4.0 was released in March 2016, so it had more or less two years of maintenance. Commits can still go in 4.4 branch if developers wants to push there, end of official maintenance means that there is no plan to do packaging and release another version from the branch in the future, so you will have to build it from sources or generate packages yourself. v5.0.6 and v5.1.2 are the usual minor releases we do from the last two stable branches. If there are any issues you are aware of and not reported on bug tracker, create the items as soon as possible to get a better chance to review and eventually fix. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 3

Siremis and kamailio 5.x.x
by Olli Attila 01 Mar '18

01 Mar '18

Hello, Question about Kamailio web GUI (Siremis): I believe the Siremis updates are still a ongoing work or has there been any updates when the new Siremis update (compatible with Kamailio 5.x.x) will be released? https://www.kamailio.org/w/2017/07/siremis-ongoing-updates/ Cheers, Olli

2 1

Kamailio ims with TCP
by Amar Tinawi 01 Mar '18

01 Mar '18

Hello list For some reason i couldn't register on kamailio ims when i use only TCP protocols between nodes, the error was solved when i enabled UDP between I-CSCF and S-CSCF is it mandatory for those two nodes to talk over UDP only ? or i'm missing something ? and in general for sip protocol which is better ? UDP or TCP ? Regards,

2 5

Trunks with registration
by Володимир Іванець 28 Feb '18

28 Feb '18

Hello all! I was testing Registration capability of UAC module and have two questions that hopefully someone have answers to: - Is it possible to tell Kamailio which transport protocol to use other than by configuring auth_proxy column of uacreg table with domain name that has SRV record? - What are the priorities of different transport protocols in SRV record when it is used? Thanks a lot!

2 2

Changing contact header
by Cibin Paul 28 Feb '18

28 Feb '18

Hello, In my setup, Kamailio acts as a registration server and route the calls to different asterisk boxes. The clients using IPV6 private ip and IPV4 public ip address are rejected by the asterisk server. On printing the contact header, I am getting the private ip (ipv6). Is there a way to change the contact header with public ip in all scenarios. I tried updating the contact header as below and unfortunately not working. Is there a better way to handle the scenario. As of now we don/t have a plan to implement ipv6 network. $var(p1) = "<sip:"; $var(p2) = "@"; $var(p3) = ":"; $var(p4) = ";ob>"; $var(x) = "" + $var(p1) + $fU + $var(p2) + $si + $var(p3) + $sp + $var(p4); remove_hf("Contact"); append_hf("Contact: $var(x)\r\n”); Thanks & Regards Cibin

2 1

Question on contribution of documentation improvements
by Floimair Florian 28 Feb '18

28 Feb '18

Hi guys! While reading through documentation I realized there are a lot of typos (typing errors) and some misspellings so I decided to try to correct them. I dit this in all of Kamailios module and lib documentation (the xml in the doc folders) and they turned out to be quite a lot. Right now this sits on my fork of kamailo on github, but I would like to get this back into the main repository. What's puzzling me though, is how to most effectively get this in via a pull request due to the following: For now I have made a single commit for each module I touched. Therefore quite a lot of commits have accumulated. Actually 152 commits in modules and 4 commits in lib. Since most of the commits in modules are rather small (ranging from 1 line changed to 4) with only a few having more, I am not sure if that's the way to go to have a single commit for each module. Therefore I am unsure if a squashed version would be more appropriate. Therefore I created a second branch on my forked repository. Unsquashed: https://github.com/ffloimair/kamailio/tree/wip-fix-typos Squashed: https://github.com/ffloimair/kamailio/tree/wip-fix-typos In total a diff --stat to master results in 205 files changed, 521 insertions(+), 521 deletions(-) So please let me know what would be the wanted (and most efficient) way to proceed. With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 Tel: +43-662-85 62 25 Fax: +43-662-85 62 26 http://www.commend.com Security and Communication by Commend FN 178618z | LG Salzburg

2 1

cnxcc module - make auth for redis connection
by Donat Zenichev 28 Feb '18

28 Feb '18

Hi community. This message is specially dedicated to developers of the kamailio project. I need to find solution how to make authentication for connections of cnxcc with redis. For that moment, cnxcc module is able to connect to redis only without password (Carlos Ruiz Díaz will add this possibility later?). So that, this makes database vulnerable. Well, furthermore I don't want to use local redis database, cuz I think machine that contains kamailio daemon, must perform just this kind of actions - SIP proxying (kamailio server is not to be a database server or anything else). For that moment we use separate redis db for cases like that, but this is not best practice I think. I already tried to find solution (for e.g. connections via another module like db_redis: https://github.com/kamailio/kamailio/pull/1432) but I got advice - achieving my goal by sr-users mailing list. Please pay attention for my request. I think this will be very helpfull for others. -- -- BR, Donat Zenichev Wnet VoIP team Tel Ukraine: +380(44) 5-900-800 Tel USA: +164(67) 8-174-17 (ext 1320) https://w-net.us/ <http://wnet.ua>

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kamailio 5.1 compilation error
by Aqs Younas 28 Feb '18

28 Feb '18

Greetings list, I am trying to install Kamailio 5.1 from source code on Debian 8.10. But It is failing with below error. Don't know what could be wrong. Any suggestion is much appreciated. Thanks. make --no-print-directory -C ./squirrel/squirrel SQMARCH64="-m64" CC_EXTRA_FLAGS="-fPIC" \ OUT="../../libsquirrel.a" sq64 gcc -O2 -m64 -D_SQ64 -fno-exceptions -fno-rtti -Wall -fno-strict-aliasing -c sqapi.cpp sqbaselib.cpp sqfuncstate.cpp sqdebug.cpp sqlexer.cpp sqobject.cpp sqcompiler.cpp sqstate.cpp sqtable.cpp sqmem.cpp sqvm.cpp sqclass.cpp -I../include -I. -Iinclude -fPIC gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory gcc: error trying to exec 'cc1plus': execvp: No such file or directory Makefile:51: recipe for target 'sq64' failed make[3]: *** [sq64] Error 1 Makefile:44: recipe for target 'libsquirrel.a' failed make[2]: *** [libsquirrel.a] Error 2 Makefile:511: recipe for target 'modules' failed make[1]: *** [modules] Error 1 make[1]: Leaving directory '/usr/src/kamailio/src' Makefile:27: recipe for target 'all' failed make: *** [all] Error 2 Br, Aqs.

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please downgrade warning message
by Juha Heinanen 28 Feb '18

28 Feb '18

I noticed this when I start a correctly configured kamailio: Feb 27 21:55:45 rautu /usr/bin/sip-proxy[29529]: WARNING: registrar [registrar.c:369]: mod_init(): empty reg_xavp_cfg or sock_flag not set -> resetting Neither of those two are mandatory. I'm not using registered() function. Please downgrade the message to info or debug level. -- Juha

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rtpengine transcoding benchmarks?
by Brooks Bridges 27 Feb '18

27 Feb '18

I realize the transcoding support is rather new, however has anyone run any real world tests on it yet? Trying to get a sense of what the performance will be. I'm aware that it's forced into userspace for transcoding so it's not *nearly* as efficient as the iptables module, however even if I have to throw a rack full of whiteboxes at it to get anywhere close to decent numbers that's still going to be comparable to a lot of the commercial options out there right now. Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

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Kamailio v5.0.6 Released
by Daniel-Constantin Mierla 27 Feb '18

27 Feb '18

Hello, Kamailio SIP Server v5.0.6 stable release is out. This is a maintenance release of the previous stable branch, 5.0, that includes fixes since the release of v5.0.5. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.6 or v5.1.x. For more details about version 5.0.6 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/02/kamailio-v5-0-6-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.1, at this moment with its latest release v5.1.1 (and v5.1.2 planned to be out in the next few days). See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Kamailio and dialogs..
by Laura 27 Feb '18

27 Feb '18

Dear list.. I'm using kamailio on a live enviroment with dialogs support.. what I made is a script that permit me to see the concurrent call (live) with all parameters using the info saved on dialogs tables. The problem is that if the call is not going on the first termination of LCR but for examples in the second, the dialog is not updated with the paramenter of the second supplier.. Is this problem only on my system or maybe i forgot to make some configuration steps ? Regards Laura

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Kamailio v4.4.7 Released
by Daniel-Constantin Mierla 26 Feb '18

26 Feb '18

Hello, Kamailio SIP Server v4.4.7 stable release is out. This is the last planned maintenance release of the old stable branch, 4.4, that includes fixes since the release of v4.4.6. There is no change to database schema or configuration language structure that you have to do on upgrading previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.7, or even better to v5.0.x or 5.1.x. For more details about version 4.4.7 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/02/kamailio-v4-4-7-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: theb branch 4.4 is going out of maintenance life time, the latest stable branch is 5.1, at this moment with its latest release v5.1.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-1-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Kamailio json_rpc isn't reloading htable data from database into memory
by José Seabra 26 Feb '18

26 Feb '18

Hello there, I'm using xHTTP and json_rpc module to receive http instructions to reload htable's from database to memory. The way i'm calling the json_rpc in kamailio script is the following: $var(json_rpc)='{"jsonrpc": "2.0", "method": "htable.reload", "id":"htable"}'; jsonrpc_exec($var(json_rpc)); The execution of this command doesn't give any errors, but the changes made on htable database weren't transmitted to the memory. I have attached the kamailio log where you can see that there isn't any erros on json_rpc execution during reload instructions, the strange thing here is that the htable module only says: 2018-02-23 14:33:44.968 DEBUG: htable [ht_db.c:96]: ht_db_open_con(): database connection opened successfully doesn't report if the reload was success or not. Those logs attached were printed with kamailio debug set to level 4. The kamailio version is 4.4.2. Thank you -- Best Regards José Seabra

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks, Juha. You are correct. That feature request is directly relevant to this thread. I will upvote the feature request, and add a link back to this thread. The proposed workaround of using async workers is the one I opted to use.

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

>A common design which avoids this is to use TCP at the client edge and >UDP inside the network core. This is one of the reasons why TCP is not >optimal for use inside the core. That makes sense, but is unfortunately not an option for me due to strict security requirements. I need to use TLS on the whole path. Another option I explored was to have the edge proxies not always use the same TCP connection for sending to the registrar. If I could find a way to load balance across a number of TCP connections, that would probably work for me. Perhaps there is a way the DISPATHCER module can be configured to accomplish this. Maybe the dispatcher configuration can list multiple copies of the same destination, but each having a different send socket address, and then can load balance across those. Does that make any sense?

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Load-balance on 2 proxies
by David Villasmil 23 Feb '18

23 Feb '18

Hello guys, I hava a setup where there are 2 kamailios in active-active connected to the same db (no registration yet). They both forward requests load balancing to several freeswitches. My question is, is the load balance algorithm independent on the 2 servers, or do they act as 1 load balancing in term of sending the next request to the next FS in the list independent of which kamailio sent the last request? I.e. Kam1 -> fs1 Kam2 -> fs2 Kam1 -> fs3 Kam1 -> fs1 Kam2 -> fs2 And so on? Thanks for your help David

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks. Regarding UDP, that all makes sense. For my short-term needs, to minimize risk and infrastructure changes, it looks like the ASYNC module will work. Here's the basic outline of what I'm doing: -------- async_workers=8 route { route(CHECK_IS_EXPENSIVE_OPERATION); if ($var(isExpensiveOperation)) { async_task_route("COMMON"); } else { route(COMMON); } } route[COMMON] { # Do all my normal routing logic. } -------- I detect at the very top of the main route whether the operation will be expensive. If so, I process the message with an async worker. If not, I directly process the message on the TCP worker. The rest of the routing logic remains unchanged. The only difference is that I execute the expensive operations using the async workers. It's not the prettiest thing in the world, but it seems like an OK short-term fix to relieve pressure. I will revisit the bigger architectural ideas in a later phase. Thanks again for all your help and good ideas. Happy Friday. -Cody

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Re: [SR-Users] How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Thanks very much for the quick replies, Alex and Brandon. The main reason I'm hitting a bottleneck is because my architecture is not optimal. I have a number of edge proxies which communicate with all my clients. The clients are usually distributed pretty evenly across all the proxies. On those proxies, client TCP connections are distributed pretty evenly across the TCP workers, so that's all fine. The problem occurs when the edge proxies communicate with the central registrar. When that happens, the SIP messages from a very large number of clients are all multiplexed onto a single TCP socket connection between the proxy and the registrar. That narrowing results in the registrar worker processes not being utilized efficiently. For example, say I have 3 edge proxies, and my registrar has 8 cores and 8 worker processes. I want to spread the message processing across all 8 workers, but I'm only able to utilize 3, because all the messages from a given edge proxy are being processed by a single TCP worker on the registrar. The long-term solution is to change my architecture. I should not be doing all the expensive work on a single central registrar. I'm planning to move to a mesh architecture, where all servers are the same, and the registration processing is divided amongst all the servers. That design makes sense, but it's more complex and will require more information sharing amongst the servers. I'm looking for a short-term improvement that will give me some breathing room, which led me to looking at the ASYNC module. >...one alternate suggestion that could help spread load on actual Kamailio TCP workers is by firing up additional workers on alternate ports. That makes sense. I could have my central registrar listen on several different ports, and could perhaps use the dispatcher module on my edge proxies to try and evenly divide the traffic across those ports. I will look into that. Thanks again. -Cody

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How are SIP messages divided amongst TCP workers?
by Cody Herzog 23 Feb '18

23 Feb '18

Hello. I'm curious as to how SIP messages received over TLS are divided amongst the available TCP workers. Based on some searching, I was hoping and expecting that the messages would be load balanced amongst the TCP workers at the level of individual SIP messages. However, with my configuration, it seems that the work is load balanced at the TCP connection level. I will now elaborate on my basic architecture. I have several edge proxies to which the clients connect. Those proxies do not handle registration, or do much of anything, they just forward traffic to a central registrar server. On the central registrar server, it seems that a particular TCP worker is assigned to handle all the traffic arriving from a particular edge proxy. As long as the TCP connection between the servers remains open, it seems that the particular worker is locked into handling all traffic on that connection. Does that sound like expected behavior? That behavior is not ideal for me, because I would like to evenly divide the SIP message load across all the TCP workers on my central registrar. Perhaps I'm doing something bad in my config file that is preventing the load balancing from working at the SIP message level. If that behavior is expected, is there anything I can do to achieve more even load balancing? Perhaps the ASYNC module might work for my needs. It seems like I could use async_task_route() to divide certain messages evenly amongst the async workers. Does that sound reasonable? In my use case, certain messages require a lot of CPU time to perform crypto calculations, and I want those calculations to be distributed evenly across all CPU cores. Thanks. -Cody

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rtpengine codecs transcoding: performance
by Abdoul Osséni 23 Feb '18

23 Feb '18

Hello, Is there a study comparing the performance (quality and cpu/ram consumption) of codecs transcoding between rtpengine and asterisk/freeswith? Is rtpengine more efficient? Regards Abdoul

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RFC: db_redisusrloc module - review and how to deal with
by Daniel-Constantin Mierla 23 Feb '18

23 Feb '18

Hello, (cross-posting being something I want to get developers and users express their opinions). We were recently in a (lucky) situation to have two new modules submissions targeting more or less the same purpose: allowing to use Kamailio with a Redis backend via database API. One was submitted by Andreas Granig (Sipwise) and already merged with the name db_redis, because it was designed from the beginning as a generic DB connector, so the module can be used with auth_db, acc, usrloc, ... * https://www.kamailio.org/docs/modules/devel/modules/db_redis.html The second one was submitted by Surendra Tiwari (Plivo), initially having a naming conflict with db_redis, but renamed to db_redisusrloc, see the pull request at: * https://github.com/kamailio/kamailio/pull/1446 Now, this email is about deciding the way to go forward with the second module. It was designed to be used only for usrloc in the first phase, with many specific usrloc attributes hard coded inside db_redisusrloc. Surendra said (in a private chat) that the long term plan is to make it work for other modules. Anyhow, at this moment is very tied to usrloc, therefore the name of the module. Given that the backend (Redis) is not an SQL engine, mapping over Kamailio's DB API needs some schema definition (see the readme of db_redis) in order to make it generic and work for all our modules that use a DB backend. On the other hand, to squeeze the best of the backend, specially in no-SQL cases, having a dedicated DB connector module optimized for a specific module might help to get more performances/high-availability/scalability from the backend itself. In this case, for example the expires value for a contact record can be set inside redis, so kamailio module doesn't have to run a timer routine to clean up (it doesn't mean db_redisusrloc does it right now, I didn't have time for a proper review, just giving an example). Surendra said they use it in production for couple of months now and it is several times faster than using usrloc with db_postgres (iirc, not db_mysql) for db_mode=3 (database only mode). But of course, the reverse of the medal with a dedicated db connector for a module: it adds overhead to code maintenance (besides generic updates due to external library changes, I expect changing something relevant in usrloc, like adding new columns, would require updates in this module as well). So, there are few things I want feedback on: 1) how do you fill about splitting from a generic-only DB connectors to have also some dedicated ones? This is more from confusion point of view, as a general rule so far, we do not deny contributions if there are other options for same kind of feature (e.g., many lcr or nat traversal options). As long as the contributor is willing to maintain the code, we were fine. 2) I guess usrloc, presence and dialog modules would be the main suspects that would benefit for such dedicated connectors, in other cases might not worth adding dedicated connectors. Missing any other module one would like to squeeze more performances with a dedicated connector? 3) should we set a different naming policy for such modules, for example: use *dbs_* prefix instead *db_*, to suggest better it is a DataBase Specific connector? 4) Andreas said he plans to do some performance testing of usrloc module with the two modules and see the differences. Anyone else that wants to do it? It can be a good metric to see if it worth going one way or another? 5) Helping to review the pull request, specially if you use Redis, is appreciated. Personally I am very short in available time these days, next week I plan do to new Kamailio stable releases, so the schedule is not getting lighter in my side. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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Checking dispatcher gw state
by Daniel Greenwald 23 Feb '18

23 Feb '18

Is there a way thru kamctl or kamcmd to check which dispatcher gateways are currently inactive? "kamctl dispatcher show" doesn't seem to have that info and I can't find anything in the docs. Thanks

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How to Re-transmit the current Request when a negative reply arrives?
by Arun NV 21 Feb '18

21 Feb '18

I am using Kamailio 4.4. And I am forwarding an INVITE to my client. but sometimes this client will send a negative response.And I am able to drop that negative response from Kamailio. But I need to re-transmit the same request to the client again. How do I do that? Thank you, Arunbalan <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> Virus-free. www.avast.com <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2>

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Return codes within a route
by Jeff Bilyk 20 Feb '18

20 Feb '18

Hello, Running into odd behaviour in Kamailio 5.0.2 and I'm hoping to clarify. I have a simple route that uses the textops functions subst and remove_hf. Other than that, there is nothing else done within the route. In some cases, the remove_hf function returns false as there are no matching headers. Documentation in https://www.kamailio.org/wiki/cookbooks/5.0.x/core#return seems to show that if there are no explicit return statements, the route should return 1. However, this route appears to be returning -1. My question is: if a function within a route returns a negative value, and there are no explicit return statements, will the entire route return -1 (as it appears is happening), or should the route return 1 as the core cookbook indicates? Thanks in advance, Jeff

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Kamailio 4.3 decrypting SIP-TLS
by Remsik,Robert 20 Feb '18

20 Feb '18

Hello! I'm working on setting up our kamailio instance to use TLS for inter-phone communication and so far I've been able to configure the GUI and get it to work which is great. However, I'm trying to grab a packet capture and decode the TLS session to get the SIP ladder (so I can troubleshoot issues) and so far I've been unable to. I've been following the article at: http://packetpushers.net/using-wireshark-to-decode-ssltls-packets/ and using the kamailio self-signed cert as I think that's the cert being used with no success. I'm getting the Cipher Suite: TLS_RSA_WITH_AES_256_CBC_SHA (0x0035) which doesn't use perfect forward secrecy so it should be able to be decoded. Is there a way to decode the kamailio TLS encoded communications so I can troubleshoot issues that may come up? Thank you in advance, Robert Robert Remsik Telecom Desk Phone: 970 491 7120 Robert.Remsik(a)colostate.edu

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Inserting a message into a transaction?
by Daniel Tryba 20 Feb '18

20 Feb '18

For a retarded endpoint (an old ISDN PBX behind a Smartnode) I need to insert a 180 reply before sending a 183 (with SDP), otherwise the PBX fails to playback the inband audio to the caller behind the PBX. I can't find hints on how to do this. uac_req_send() can be used to start a new transaction, but I only need to generate a 180 based on the 183 in the on_reply route. Current Q&D hack is to send a 180 on the initial INVITE, works but not ideal.

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Re: [SR-Users] lost mysql connections
by Jan Gaida 20 Feb '18

20 Feb '18

Hello juha, Sorry for getting back on this old topic. Can you confirm that with Kamailio v4.3 your problem was solved? Regards, Jan

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Create ISUP Body
by Duarte Rocha 20 Feb '18

20 Feb '18

Hey Everyone I'm setting up a load testing scenario for my Kamailio. In order to do that i've created another Kamailio to answer "200 OK" to an Invite in order to simulate a call being answered. One of the scenarios I need to test is reading values from the ISUP IAM or ACM body. Here's the code on the Kamailio that replies with 200 : $var(acm) = "06 02 04 01 2c 01 fb 0c 08 83 10 19 7c 82 12 15 00 36 01 19 40 01 00 39 06 2c c0 36 c0 40 c0 00"; set_reply_body("$var(acm)","application/isup;base=itu-t92+;version=etsi356"); if( is_method("INVITE") && !has_totag() ) { · xnotice("First Invite"); send_reply("200","OK"); } However, when i use this and i do a pcap i get a "Message Type: Overload", which doesn't support optional parameters. What am i doing wrong ? Are there other methods to setup a simple ISUP part without hex ? Thanks in advance

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Kamailio mod ALIAS_DB: multiple aliases
by Stefan Rüetschli 20 Feb '18

20 Feb '18

Hello Kamailio-Users I'm using kamailio 5.1 with Alias_DB module. My subscribers register with a username like 0000001, 0000002 etc... Every user has a list of multiple aliases which they can use on their SIP-Trunk. I want to avoid that the user can use any alias for outgoing calls, only the aliases which are boud to his username in the aliases-db. So, if the user uses an alias which does not belong to him, i want to overwrite it to the first alias for his username in the aliases-db. At the moment, the function alias_db_find returns me the last alias found in the db. Is there any way to do it? Or is there a way to extend the dbaliases with a new column marking the "primary" alias for an account?|||| This would be the perfect solution. Regards Stefan

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Info: Kamailio World 2018 - Preview With 15 Sessions
by Daniel-Constantin Mierla 20 Feb '18

20 Feb '18

Hello, less than 3 months till the start of the 6th edition of Kamailio World Conference, time if flying fast! About one week ago we published the details for a group of accepted speakers, today we made a selection of sessions at the Kamailio World 2018. We had more proposals than we could accommodate, we are trying hard to fit in as many as possible, taking also in consideration the feedback from the participants at the past editions. For now you can head to the Schedule page and see the details of 15 sessions, from both workshops and conference days: https://www.kamailioworld.com/k06/schedule/ A very divers range of topics, from using Kamailio for emergency services (112/911), scaling with Redis backend, deploying in a containerized environment with Docker and Kubernetes, how to migrate the SIP routing logic to rich KEMI languages such as Lua, Python or Javascript, unit testing for Kamailio and test driven deployments, to blockchains in telephony, using Kamailio and FreeSwitch together, or latest updates from Asterisk PBX. The IMS/VoLTE workshop is going to show what you can do with latest Kamailio in mobile networks. And, of course, we have the very popular two sessions that never missed a Kamailio World edition: Dangerous Demos with James Body and VUC Visions with Randy Resnick. The details for other speakers and sessions will be published in the near future, stay tuned! Do not miss Kamailio World Conference 2018, it is going to be another great edition! You can register now at: https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at the next Kamailio World Conference, during May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

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IMS: maybe a bug in S-CSCF: should we create an issue?
by Valentin Christoph 20 Feb '18

20 Feb '18

Hi all, We are using kamailio 5.1.0 as S-CSCF as follows: We have two machines and two kamailio instances, one on each machine, both working on the same database (together they build one highly available S-CSCF). After registering a user via the first kamailio instance, then the first instance is stopped (simulating an outage), and the HSS sends an RTR to the second instance. However, the second instance does not find the registered user in its cache and it does not access the database --> the deregistration fails. Configuration: kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) Does anyone immediately "see" a mis-configuration, or should we create an issue? Details in the mail below. Thanks, Christoph From: Friedrich Peter Sent: Friday, February 16, 2018 6:38 PM To: Valentin Christoph <Christoph.Valentin(a)kapsch.net> Cc: Onic Roman <Roman.Onic(a)kapsch.net>; Habashizadeh Mahmoud <Mahmoud.Habashizadeh(a)kapsch.net> Subject: AW: Kein RTR Message nachdem SCSCF-1 outage Hello, following scenario at REGISTER of catt subscriber: start with empty db and empty usrloc cache node1(.23): MariaDB [scscf]> select * from impu; Empty set (0.00 sec) [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After catt REGISTER: catt_subscriber visible in DB and usrloc cache of .23 but missing in usrloc cache of .24 MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) MariaDB [scscf]> [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 3542 client: n/a } } } [root@vict-ims ~]# [root@vict-ims2 etc]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims2 etc]# After kamctl stop + kamctl start of .24 catt_subscriber is visible in usrloc cache of (.24): preload_data loads data from db at module init [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2949 client: n/a } } } [root@vict-ims2 ~]# Now deregister catt_subscriber from HSS1: catt_subscriber disappears from db and usrloc cache of (.23) but persists on (.24) as registered in usrloc cache HSS >hss.deregister_subscriber catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> 0 [root@vict-ims ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net [root@vict-ims ~]# MariaDB [scscf]> select * from impu; Empty set (0.00 sec) MariaDB [scscf]> [root@vict-ims2 ~]# kamctl kamcmd ulscscf.showimpu sip:catt178@vict-ims.net { impu: sip:catt178@vict-ims.net state: registered barring: 0 ccf1: ccf2: ecf1: ecf2: subscription: { impi: catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> service profiles: { 1: { impus: { 1: sip:catt178@vict-ims.net 2: sip:+4350255872178@vict-ims.net 3: tel:+4350255872178 } } } } contacts: { sip:catt178@193.154.113.40:5060;transport=UDP;rinstance=317-catt178: { expires: 2571 client: n/a } } } [root@vict-ims2 ~]# Scenario of RTR from HSS after REGISTER of catt_subscriber at (.23) and outage of (.23): subscr. visible in db but not visible in usrloc cache of .24 HSS sends diameter RTR message to (.24) and deregister callback in (.24) searches in usrloc cache and does not find catt_subscriber and fails. MariaDB [scscf]> select * from impu; +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | id | impu | barring | reg_state | ccf1 | ccf2 | ecf1 | ecf2 | ims_subscription_data | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ | 196 | sip:catt178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 197 | sip:+4350255872178@vict-ims.net | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | | 198 | tel:+4350255872178 | 0 | 1 | NULL | NULL | NULL | NULL | catt178(a)vict-ims.net<mailto:catt178@vict-ims.net> sip:catt178@vict-ims.net sip:+4350255872178@vict-ims.net tel:+4350255872178 | +-----+---------------------------------+---------+-----------+------+------+------+------+----------------------------------------------------------------------------------------------------------------------+ 3 rows in set (0.00 sec) Cfg file of (.24): kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "enable_debug_file", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "matching_mode", 0) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", "cluster://cluster1") kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_url", DB_URL) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "db_mode", 2) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_interval", 10) kamailio-scscf/kamailio-scscf-modules.cfg:modparam("ims_usrloc_scscf", "timer_procs", 4) lg P The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 4

issue with running kamailio
by justin barnes 20 Feb '18

20 Feb '18

I am installing from githum kamailio 5.0 on my ubuntu-16.04-x86_64 VPS. when I start the kamailio server I get this error : /# /etc/init.d/kamailio start [....] Starting kamailio (via systemctl): kamailio.serviceJob for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. failed! Ilooking up the status, I get this kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/systemd/system/kamailio.service; disabled; vendor preset: enabled) Active: failed (Result: exit-code) since Mon 2018-02-19 15:23:41 EST; 4h 10min ago Process: 18969 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GR Feb 19 15:23:41 khynepassion systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Feb 19 15:23:41 khynepassion systemd[18969]: kamailio.service: Failed at step EXEC spawning /usr/sbin/kamailio: No such file or directory Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Control process exited, code=exited status=203 Feb 19 15:23:41 khynepassion systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Unit entered failed state. Feb 19 15:23:41 khynepassion systemd[1]: kamailio.service: Failed with result 'exit-code'.

1 0

LCR options as of 2018
by Francisco Valentin Vinagrero 19 Feb '18

19 Feb '18

Hello, I am configuring a Kamailio node that will interface with 3 carriers and will implement Least Cost Routing for around 900 destinations. I have already implemented a quick prototype with the LCR module and I have been reading the CarrierRoute doc and the CGrates site. I don't really need advanced functionality, just: - Re-routing to a backup carrier when the main one is not available. - Ping with OPTIONS to all carriers so I can detect if any of them is not available (à la Dispatcher) - Translate prices from a central DB to routing priorities. - Get some accounting post-call info about the routing decision (acc module) So far, while testing the LCR module I have had some issues with the rule IDs being integers that auto-increment, so the LCR rules need to be referenced from the rule_target table using this numeric ID. I can see some issues when scaling this to 900 lcr_rules and 2700 lcr_rule_targets with 3 priorities. In addition to that, I think the OPTIONS mechanism is just used by the lcr module when a gateway is marked as defunct, but the gateways are not actively monitored in the normal status. Is this correct? I have been considering CarrierRoute, which has a slightly simpler table structure, but I can't find any other advantage for my use case. Regarding CGrates, it looks great but it looks like it exceeds my current needs. For those who have implemented LCR in your kamailios, am I missing something regarding performance, scalability, etc? Are there any newer modules that can implement this, or maybe just a dispatcher-on-steroids setup will do the trick? Cheers, Francisco.

3 6

User state
by Magnus Löfqvist 19 Feb '18

19 Feb '18

Hi, We are running for a customer a kamailio server for user registration and routing of calls. Also using Asterisk for queue management and IVR. Now we are in the progress of writing a web portal where the customers users should be able to see if there collueges are busy in phone or availible. In asterisk I would use hints and listing in events in AMI. But what way are the best in Kamailio? Best Regards Magnus Sent from Nine<http://www.9folders.com/>

2 1

PR1447: first attempt at app_python3
by Anthony Alba 18 Feb '18

18 Feb '18

Hi Daniel and kamailio contributors, I have submitted a PR for app_python3 here: https://github.com/kamailio/kamailio/pull/1447. It is very rough, but would appreciate your feedback. Notes: 1. Almost a verbatim port of app_python; just changing obvious semantics. String->Unicode is practically the only large change. No change to symbol names and export functions so the two modules cannot coexist. (Don't loadmodule "app_python.so" and loadmodule "app_python3.so) 2. Module initialization semantics in Py3 have changed: add the top level modules to the inittab before calling Py_Initialize(). I have done this for KSR and Router. I.e. PyImport_AppendInittab() is called for Router and KSR *before* Py_Initialize() 3. Makefile looks for a python3 binary (PYTHON3 in the Makefile) to get include paths and link libraries. Here is an example compile line for IUS Python 3.6 on CentOS 7 (no native Python3 from RHEL) ## default: needs to find python3 and python3-config ## needs to find both python3.6 and python3.6-config make -C src/ PYTHON3=python3.6 SKIP_MODULES="${SKIP_MODULES}" quiet=verbose module_group_py="app_python3" group_include=py every-module 4. The semantics for GIL and thread management are too different for one-to-one replacements. I have therefore decided to comment out all threading calls. Fortunately, kamailio does not used threads at the moment. This is more complicated than it seems: even with Py2+thread management: there would be deadlocks if kamailio->python2 script->Cython extension with threads(e.g. grpcio). For an MVP it would be best to get it in without thread management. 5. It should backport to 5.1. 6. Tested locally, with a routing scripting making many python_exec() calls Thank you. Anthony Alba

1 1

PUBLISH and ETag
by Cyrille Demaret 16 Feb '18

16 Feb '18

Hi, I'm using Kamailio with presence enabled and Asterisk PJSIP and outbound-publish. My problem is happening when I place 2 consecutive calls from Asterisk : When I make a first call Asterisk sent the following: PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e69723a610 82 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Event: dialog Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early.. Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPj4f9c19eb-26d8-4bb1-8f00-e6972 3a61082;received=192.168.100.37 From: sip:201@mydomain.com;tag=a560e088-9e8a-49f2-a9b1-4a0ec31340bf To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-ff39 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10697 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19863.16.0 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 When the call is done, Asterisk sent another PUBLISH telling that the call if terminated : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97ab8c7 52 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19863.16.0 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 230 <?xml version="1.0" encoding="UTF-8"?> terminated.. And Kamailio replies : SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.37:5080;rport=5080;branch=z9hG4bKPja93efb01-a518-445e-9e9b-f6f97 ab8c752;received=192.168.100.37 From: sip:201@mydomain.com;tag=165fb3b2-ec0e-4786-889f-eb194ad456ce To: sip:201@mydomain.com;tag=b596189c6de9c38f624fd84638f43be6-48b4 Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10698 PUBLISH Expires: 180 SIP-ETag: a.1518775074.19873.18.1 Server: kamailio (5.0.5 (x86_64/linux)) Content-Length: 0 Here, the SIP ETag is a.1518775074.19873.18.1. The problem is if I make a new call before the expiration of the previous SUBSCRIBE, Asterisk reuse this SIP ETag according to the RFC : PUBLISH sip:201@192.168.100.37 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.37:5080;rport;branch=z9hG4bKPj9d13bb82-31d9-48db-9672-bd4b6b4f22 f0 From: sip:201@mydomain.com;tag=33e6b028-0444-4b3a-8bc2-4a987a291528 To: sip:201@mydomain.com Call-ID: 5adcf0a0-f138-44d6-8c56-eaf7c3b3b183 CSeq: 10699 PUBLISH Event: dialog SIP-If-Match: a.1518775074.19873.18.1 Expires: 180 Max-Forwards: 70 User-Agent: Asterisk PBX 14.6.0 Content-Type: application/dialog-info+xml Content-Length: 247 <?xml version="1.0" encoding="UTF-8"?> early. Kamailio refuse it with this error : "Trying to update an already terminated state. Skipping update." because the call is considered as terminated. The RFC is stating : When updating previously published event state, PUBLISH requests MUST contain a single SIP-If-Match header field identifying the specific event state that the request is refreshing, modifying or removing. This header field MUST contain a single entity-tag that was returned by the ESC in the SIP-ETag header field of the response to a previous publication. Why Kamailio is acting like that? Best regards, Cyrille

2 2

Problem with "transport=tcp" and rtpproxy
by Pau Giralt 16 Feb '18

16 Feb '18

Hi! I'm working with kamailio and rtpproxy. When I configure the clients to send sip messages through TCP adding the ";transport=tcp" in the sip message the rtpproxy is not working anymore. The code is never passing the point pasted below, inside route[NATMANAGE] in kamailio.cfg if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; Therefore, rtpproxy_manage("co"); is never reached. Could it be a parsing problem? ( check_route_param("nat=yes") ) The config file is exactly the same as the one provided in the kamailio repository (the default one) Thanks!

2 7

Raspberry Pi deb packages -> built
by Mititelu Stefan 15 Feb '18

15 Feb '18

Hi people! After a bit of research and lib installations, I've managed to build the full kamailio deb packages for a raspberry pi 3 (version B) running raspbian stretch (lite) that can be found at [1] (with a really few module exceptions that might be used in really specific scenarios -> I think) Just for fun, I'm planning to do some CPS stats tomorrow (if I got the time), with a basic config and also share here, but, if someone else tried it before and is able to share some stats/benchmarks, would be greatly appreciated (and interesting to compare, given my "decent" rpi overclock), because I'm also curious about VoIP stuff on raspberry pi! :D [1] https://github.com/smititelu/rpi/tree/master/kamailio/5.1.1 Thank you, Stefan

2 3

KEMI
by Aleksandar Sosic 15 Feb '18

15 Feb '18

Hi everyone, we're valuating on rewriting our kamailio routes in python. We're doing a lot of string manipulation and GraphQL API queries so the kamailio scripting language is a little bit limited for this use case. Can someone tell us if the KEMI framework is stable enough for production? Kind regards, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

5 11

Re: [SR-Users] RTPEngine Transcoding
by Social Boh 15 Feb '18

15 Feb '18

Hello, I'm trying to use this configuration: Softphone A audio codec g722 and Opus Softphone B audio codec PCMA and PCMU Kamailio with rtpengine_manage with codec-transcode-PCMU codec-transcode-PCMA in the request Softphone A call Softphone B The RTPEngine DEBUG show many Decoder Error: https://pastebin.com/k1Zhw8HS Thank you Regards --- I'm SoCIaL, MayBe El 13/02/2018 a las 20:05, Richard Fuchs escribió: > On 2018-02-13 07:22 PM, Social Boh wrote: >> Thank you. >> >> which is the best way to implement ir >> >> codec-strip-all and the add codecs? >> >> Is there a document about how work the transcoding? > It's really up to you and depends on what you want to achieve. Doing a > strip-all and then adding codecs you want to offer is definitely one > way. Another is to just add codecs that you know or think the peer may > want to see. > > Cheers >

2 2

Invalid payload location
by Soltanici Ilie 15 Feb '18

15 Feb '18

Hi All, Can someone explain why i'm receiving such errors in syslog: [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:228]: set_sdp_payload_fmtp(): Invalid payload location [4839]: {INVITE (1)] | Seq=101 | Source IP=46.166.139.7 (udp)} ERROR: <core> [core/parser/sdp/sdp.c:212]: set_sdp_payload_attr(): Invalid payload location This is how the SIP Packet is looking: 2018/02/15 10:51:40.916040 46.166.139.7:30189 -> X.X.X.X:5060 INVITE sip:01031203697179@X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP 46.166.139.7:30189;branch=z9hG4bK-5342365299746297160962993236828333618949;rport=30189 From: <sip:88893855391@46.166.139.7:30189>;tag=0vkn019bur To: <sip:01031203697179@X.X.X.X:5060> Contact: <sip:88893855391@46.166.139.7:30189> Call-ID: 0889633580294714035892119863939070195463 CSeq: 101 INVITE User-Agent: Hello Annonymous Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 356 v=0 o=SIP-Session 950029666 1307322426 IN IP4 46.166.139.7 s=SIP c=IN IP4 46.166.139.7 t=0 0 m=audio 5075 RTP/AVP 18 0 8 101 a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=fmtp:9 bitrate=64000 a=rtpmap:104 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv Thank You.

1 0

python_exec return 0 causes script hangs; return -1 is treated like 0
by Anthony Alba 15 Feb '18

15 Feb '18

5.1.1: if python_exec method returns 0, then the script hangs If the python_exec method return -1, then expressions treat it as if it returned zero. Somewhere there seems to be an offset by 1 issue. if (!python_exec("method")) { # if you return -1, the script does not hang but treated like zero } ################## Example 1: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return 0 Calling: python_exec("hello") will cause script hang ################### Example 2: class kamailio(object): def hello(self, msg): KSR.info("hello, world\n") return -1 Calling python_exec("hello") if (!python_exec("hello")) { ## method returns -1 ## why did we get here? xlog("L_INFO","method return zero!\n"); } shows INFO: <script>: method return zero!

2 1

[IGNORE] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Apologies for the noise; it is working now. Not sure how I got the extra " 0" appended to the method arguments for python_exec(). AAlba

1 0

[BUG?] 5.1.1 python_exec, extra chars ' 0' appended to args
by Anthony Alba 15 Feb '18

15 Feb '18

Using 5.1, the python_exc args string is getting an extra [_0' appended When I call python_exec("method", "string"); In python: def method(self, msg, args): KSR.info("DBG: {} {} {}".format(msg, args)) This will show 'string 0'; instead of string. AAlba

1 0

RTPEngine Transcoding
by Social Boh 14 Feb '18

14 Feb '18

Hi List, I just read on the RTPEngine page the Proxy RTP now support transcoding: https://github.com/sipwise/rtpengine#transcoding is there a plan to support it? Regards -- --- I'm SoCIaL, MayBe

5 10

RTPEngine QoS statistics and leg labelling
by Alex Balashov 14 Feb '18

14 Feb '18

Hello, I am excited about the new media statistics from RTPEngine, exposed conveniently into PVs, e.g. jitter, packet loss, MOS, time offset of lowest and highest MOS scores, etc. Great work! I am, however, having a bit of trouble understanding how to properly apply this new functionality, as it is not clearly documented (IMHO): 1. What is the meaning of "global" statistics which do not take into account an "A" and a "B" side? That is to say, if I have a packet loss or a jitter or MOS score statistic using just the generic toplevel values, e.g. https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… What does that mean? A functional call ordinarily consists of an inbound and outbound RTP stream direction relative to the endpoints on both sides of RTPEngine. Inbound A --> server --> Outbound A Outbound B <-- server <--- Inbound B Where do the "global" stats situate themselves in relation to all this? 2. I am given to understand that the statistics for the "A" and "B" sides of a bidirectional RTP stream can be separated based on an arbitrary label, using these PVs: https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… https://kamailio.org/docs/modules/5.1.x/modules/rtpengine.html#rtpengine.p.… How do I use it properly? My intuition was to try something like this: rtpengine_offer("replace-origin replace-session-connection ICE=remove label=outbound"); And in the answer/manage handler on the SDP answer: rtpengine_answer("replace-origin replace-session-connection ICE=remove label=inbound"); Later, when processing a BYE and obtaining subsequent media statistics, I tried this: if(check_route_param("proxy_media=yes")) { $avp(rtpengine_a_label) = 'outbound'; $avp(rtpengine_b_label) = 'inbound'; rtpengine_delete(); route(LOG_RTPENGINE_STATS); } But this only gets me stats for the "A" leg in the JSON object I emit: { "server_addr": "udp:127.0.0.1:5050", "a_label": "outbound", "b_label": "inbound", "a_leg": { "min": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "max": { "mos": "4.4", "at": "0:04", "packetloss": "0", "jitter": "0", "roundtrip": 0 }, "avg": { "mos": "4.4", "packetloss": "0", "jitter": "0", "roundtrip": 0, "samples": 2 } }, "b_leg": { "min": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "max": { "mos": "<null>", "at": "<null>", "packetloss": "<null>", "jitter": "<null>" }, "avg": { "mos": "<null>", "packetloss": "<null>", "jitter": "<null>" } } } I also tried rtpengine_delete("label=inbound") (and outbound) individually. Some clarity on how this all fits together would be appreciated. :-) Many thanks to Sipwise for great work on it! -- 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 14

How to add a new module to existing Kamailio Server?
by Arun NV 14 Feb '18

14 Feb '18

I am using a *Kamailio 4.4* server. And I would like to install *http_async_client* module to this server. I have looked in the module directory(*/usr/local/lib64/kamailio/modules*) to check whether this *http_async _client* module already exists there, but it was not there. And I searched for a solution and only found <http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-modular-install> the method to include new module at the compile time. I could find the modules source code on GitHub <https://github.com/sipwise/kamailio/tree/master/modules/http_async_client>. But I don't know the steps to install a new module to the existing Kamailio Thank you, Arunbalan

2 1

Use RTPProxy in 302 scenario
by Fabian Borot 13 Feb '18

13 Feb '18

I have this scenario Customer --> Kamailio --> Provider1 The route where l route the call to Provider1 is called route["INCOMING"] and I "arm" the replies and failures to : t_on_reply("INCOMING") and t_on_failure("INCOMING"). Provider1 always replies with a 302 Redirect and in the failure_route["INCOMING'] I have this code: if (t_check_status("302")) { get_redirects("*"); t_on_branch("REDIRECT_BRANCH"); t_load_contacts(); t_next_contacts(); t_on_failure("SERIAL_FORKING"); t_relay(); exit; } In the t_on_failure("SERIAL_FORKING") block I do t_relay() as long as there are more contacts to be processed. Lets say I need to try Provider_2 and Provider_3 failure_route["SERIAL_FORKING"] { xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Reply [$rs]\n"); if (!t_next_contacts()) { exit; } xlog("L_INFO","mylog: In Failure Route SERIAL_FORKING. Trying next Contact.\n"); t_on_failure("SERIAL_FORKING"); t_relay(); } I want to use RTPProxy in this scenario and I am using the function rtpproxy_manage() to let kamailio handle it automatically. RTPProxy ↑ | | Customer --> Kamailio --> Provider1 | | ↓ Provider N After the 302 response is processed I would like to have the signaling flow "Customer -> Kamailio --> ProviderN" and the RTP flow "Customer --> RTPProxy --> ProviderN" I have tried inserting the function in several places but can't make it work and what I mean is that sometimes the rtpproxy creates the 1st session and then deletes it as soon as I call it the 2nd time with a message similar to this one (log of rtpproxy) : "INFO:416f20b74a58fb75168175e61b2edece@192.168.1.61:5060:rtpp_cmd_delete_ematch: forcefully deleting session 1 on ports 63502/0". Sometimes the kamailio log shows this message: "ERROR: rtpproxy [rtpproxy.c:2536]: force_rtp_proxy(): incorrect port 0 in reply from rtp proxy" I think I must call rtpproxy_manage() before I send the INVITE to each of the Providers in the Contact list so they receive a SDP offer with an IP:Port combination provided by the RTPProxy (instead of the ones coming from the Customer). That would be in my route["INCOMING'] Then once Provider_N replies with a 183 with SDP or a 200 OK I would call it again to get a new session to be passed on the reply to my customer. But is the part I am struggling with. Should I add a new reply route inside the failure_route["INCOMING'] only if I am dealing with 302 and do it there ? I am out of ideas guys and any idea is welcome

2 2

Can rtpengine change network interface of estabished call?
by Yufei Tao 13 Feb '18

13 Feb '18

Hi, Is it possible to define two pairs of network interfaces for rtpengine, e.g. int/ext and up/down? If so during a call established between interfaces 'int' and 'ext', is it possible to send a re-INVITE from the calling party on interface 'ext', to update the RTP destination to an IP on a different interface 'up', for example? Or does the updated RTP destination have to be on the same interface as when the call was established? I have some success updating the RTP destination if it is on the same interface (different IP:port) however haven't managed to divert it to a different interface. I wonder if it is possible at all or me missing something? Cheers, Yufei

1 0

(no subject)
by Tihomir Culjaga 13 Feb '18

13 Feb '18

Hi, Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=) i have a setup like this: (Kamailio version 5.1) UA > Kamailio > Freeswitch Kamailio forwards registration to freeswitch without saving location. I have configured sip over websockets and im able to register to Freeswitcha via kamailio. Also, i use add_path_received() so kamailio adds a path header to register messages recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809: ------------------------------------------------------------------------ REGISTER sip:192.168.50.60 SIP/2.0 Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 Max-Forwards: 69 To: <sip:1001@192.168.50.60> From: <sip:1001@192.168.50.60>;tag=vq30modpgd Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Authorization: Digest algorithm=MD5, username="1001", realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90", uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef", qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001 Contact: <sip:1001@192.0.2.110 ;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600 Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: path, gruu, 100rel, outbound User-Agent: SIP.js/0.7.0 BB Content-Length: 0 Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6> ------------------------------------------------------------------------ send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0 Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435 From: <sip:1001@192.168.50.60>;tag=vq30modpgd To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ Call-ID: dlnrna9o4ngn25fb9d1vi2 CSeq: 196 REGISTER Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600 Date: Fri, 09 Feb 2018 21:38:07 GMT User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Content-Length: 0 ------------------------------------------------------------------------ Once, registered, im trying to originate a call to the registered endpoint from freeswitch this results in an invite like this: send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549: ------------------------------------------------------------------------ INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0 Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6 Max-Forwards: 70 From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK To: <sip:1001@192.0.2.110;transport=ws> Call-ID: 8212b700-8885-1236-08b3-000c293ae94e CSeq: 118770326 INVITE Contact: <sip:mod_sofia@192.168.50.65:15060> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 891 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:1002@192.168.50.65 >;party=calling;screen=yes;privacy=off Freeswitch appends a route header to the invite specifying kamailio as route and a received with original uri. well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio my RELAY route is like this 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")) { xlog("L_ALERT", "Its a Branch, go manage that !!! <$ru>\n"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_ALERT", "its onreply_route, go manage !!! <$ru>\n"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_ALERT", "Its failure_route, go manage !!! <$ru>\n"); t_on_failure("MANAGE_FAILURE"); } } $fs = "udp:192.168.50.60:5060"; xlog("L_ALERT", "Trying to rely !!! <$ru>\n"); if (!t_relay()) { xlog("L_ALERT", "failed to rely !!! <$ru>\n"); sl_reply_error(); } exit; } here is a part of the log: Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced udp:192.168.50.60:5060, to ws:192.0.2.110:5060) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core> [core/msg_translator.c:424]: clen_builder(): content-length: 916 (916) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]: prepare_new_uac(): could not build request Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2) Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: *** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws> is that because i set a $fu before t_relay() ? any idea how to approach the issue ?

2 4

onend_route() ?
by Wilkins, Steve 13 Feb '18

13 Feb '18

I meant to reply to Kamailio mailing list. From: Wilkins, Steve Sent: Friday, February 9, 2018 12:40 PM To: 'miconda(a)gmail.com' <miconda(a)gmail.com> Subject: RE: [SR-Users] t_relay()? Hi Daniel, I am not sure what I am doing wrong but I cannot get the forwarded message to print. I am using onend_route() and trying to print the results using xdbg. I have searched for examples and have had no luck. Thank you From: Daniel-Constantin Mierla [mailto:miconda@gmail.com] Sent: Friday, February 9, 2018 7:48 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org>>; Wilkins, Steve <swwilkins(a)mitre.org<mailto:swwilkins@mitre.org>> Subject: Re: [SR-Users] t_relay()? Hello, before t_relay() you can't see what is going to the network, because some headers are added when knowing target address and local socket (e.g, Via, Record-Route). You can use onsend_route for seeing the request just before writing to the network. https://www.kamailio.org/wiki/cookbooks/devel/core#onsend_route $snd(buf) will give outgoing buffer -- for more see $sndfrom(...) and $sndto(...) vars. Cheers, Daniel On 09.02.18 11:24, Wilkins, Steve wrote: Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com<http://www.asipto.com> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com<http://www.kamailioworld.com>

2 1

Request Timeouts
by Wilkins, Steve 13 Feb '18

13 Feb '18

Hello, This is actually a follow up to my last question about some providers calls being dropped. I noticed that when calls are dropped, I see a Kamailio log message "Request Timeout" about 5-10 seconds before the call is dropped. I am wondering what could cause this while a call is in progress. Does Kamailio need to respond to Asterisk with a '200 OK' for an INFO request sent to Kamailio from Asterisk? I am not sure what else could cause this call dropping behavior. Note: It does not appear to be caused from a missing 'ACK'. Thank you, -Steve

3 6

What is the mechanism used for sending push notification from Kamailio, For both IOS and Android?
by Arun NV 13 Feb '18

13 Feb '18

I am creating custom phone client, which uses *Kamailio *as a proxy and it's required to send *push notification* from *Kamailio *to the clients when a call towards the client reached on *Kamailio*. I found a module for APNS here* https://github.com/tvntsr/push <https://github.com/tvntsr/push>*. But I am developing both Android and IOS clients. So What mechanism I can use for the Android side? Is there any module available for *GCM*(Google Cloud Messaging) in Kamailio? Thank you, Arunbalan

2 1

is_audio_on_hold() doesn't support rfc3264
by Julia Boudniatsky 13 Feb '18

13 Feb '18

FYI, After some tests found, that function "*is_audio_on_hold()*" (textops) *supports only rfc 254*3, where "hold" is indicated by setting the "c" destination addresses for the media streams to zero (0.0.0.0). The newest *rfc3264*, where "hold" is indicated by a=sendonly/inactive, *is not supported*. Kamailio version 4.4.6. Best regards, Julia

4 8

Load testing: Taking CPU & MEM stats
by jeevan Chaitanya 12 Feb '18

12 Feb '18

Hi All, Found python library to take system stats during the load. >>> from getstats import GetStats >>> stat_obj = GetStats.getstats(dur=300,rate=10) >>> stat_obj.run_stats() Running.......... ***** Completed capturing the STATS ***** ######################################## # stats_2018-02-12_16-47-47_.csv # ######################################## Doc URL: http://getstats.readthedocs.io/en/latest/ If you require use and let me know. Thanks, K.Jeevan.

1 0

Very inconsistent calls
by Wilkins, Steve 12 Feb '18

12 Feb '18

Hello, I have found that calls are very inconsistent. I use Kamailio 5, Asterisk 14. When certain Providers like (Sorenson, ZVRS) make calls into a WebRTC client (tryit-jssip), sometime the calls stay up until I close them (10-15 minutes), others times those calls drop in 30 seconds. This is extremely confusing...does anyone else experience this type of behavior? -Steve

4 7

t_suspend usage in branch_route
by Aymeric Moizard 09 Feb '18

09 Feb '18

Hi, I'm trying to use t_suspend in my kamailio.cfg from a branch_route. I'm using kamailio 5.1.1 on debian jessie from http://deb.kamailio.org/kamailio51 >From the documentation: http://www.kamailio.org/docs/modules/5.1.x/modules/tmx.html#tmx.f.t_suspend i can see that "This function can be used from ANY_ROUTE ." However, the INVITE is still forwarded for this branch. So obviously the "outgoing transaction" is not suspended. May be "t_suspend" is only suspending "incoming transactions"? I tried to call "drop()". This prevent the INVITE from being sent. But in such case, i have an answer is sent immediatly to the caller (SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL)) I want the script to wait as expected after my "t_suspend". And in the block indicated by t_continue, i want to append a new branch. Any way to do this? Otherwise, do you advise me to control the list of branch before t_relay? I guess I can access the list. Not sure how.. so if you have any tips... Tks Aymeric -- Antisip - http://www.antisip.com

3 9

I: - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 09 Feb '18

09 Feb '18

Hello Everyone, I've been resolved the problem. It was the setflag(FLT_ACC) at the beginning of the block if (is_method("INVITE")) { setflag(FLT_ACC); <----- using instead following function now the dialog is correctly created. setflag(FLT_DIALOG); Best Regards Angelo Da: Angelo Colucci Inviato: giovedì 8 febbraio 2018 18:33 A: 'sr-users(a)lists.kamailio.org' <sr-users(a)lists.kamailio.org> Oggetto: [SR-Users] - w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it Hello Everyone, I'm new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We've trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn't find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { ... if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } ... } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Regular exepression issue in permissions module
by Paulo Ferreira 09 Feb '18

09 Feb '18

Hi list, I'm testing permissions module with kamailio 5.1.1 and I was struggling with regular expressions in trusted requests part of module. I noticed that 9[0-9]{4} doesn't work as expected so I had to change it for 9[0-9][0-9][0-9][0-9] to work. Is it supposed to be like that? Is there other way in this module to simplify the expression since it is not accepting the POSIX way? Thanks, Paulo -- ------------------------------------------- Paulo Ferreira VoIP@RCTS - Área de Infraestruturas Aplicacionaisz FCCN http://www.fccn.pt/ Av. do Brasil, n.º 101 1700-066 Lisboa - Portugal Telefone|Phone: +351 218440100; Fax: +351 218472167 Aviso de Confidencialidade/Disclaimer Esta mensagem é exclusivamente destinada ao seu destinatário, podendo conter informação CONFIDENCIAL, cuja divulgação está expressamente vedada nos termos da lei. Caso tenha rececionado indevidamente esta mensagem,solicitamos-lhe que nos comunique esse mesmo facto por esta via ou para o telefone +351 218440100 devendo apagar o seu conteúdo de imediato. This message is intended exclusively for its addressee. It may contain CONFIDENTIAL information protected by law. If this message has been received by error, please notify us via e-mail or by telephone +351 218440100 and delete it immediately.

2 4

t_relay()?
by Wilkins, Steve 09 Feb '18

09 Feb '18

Hi All, Could someone please tell me, what variable to I use to see what t_relay() fowarding in its message? Thank you, -Steve

3 2

- w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it
by Angelo Colucci 08 Feb '18

08 Feb '18

Hello Everyone, I’m new in Kamailio environment. We trying to setup a IMS Credit Control Client as per RFC 4006. We’ve trying to implement the suggesting configuration in ims_charging module doc. The Kamailio version is 5.1.1 Stable. The Server is configured with two local users (A and B) registered on local mysql database. The scenario is the following: 1. In the request_route object we trying to intercept the first INVITE and trigger a diameter credit control request using Ro_CCR function. 2. When A calls B without triggering mechanism the call is fine. 3. When A calls B with credit control triggering, the calls failed and the diamater request is not generated. In the debug we find the following error: exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=540 a=25 n=xlog ERROR: <script>: Individuato primo INVITE: generata nuova transazione exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=63 n=assign exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio.cfg] l=541 a=29 n=Ro_CCR DEBUG: ims_charging [ims_charging_mod.c:494]: w_ro_ccr(): Ro CCR initiated: direction:orig, reservation_units:30, route_name:CHARGING_CCR_REPLY, incoming_trunk_id:1 outgoing_trunk_id:1 DEBUG: ims_dialog [dlg_hash.c:926]: internal_get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_hash.c:956]: get_dlg(): no dialog callid='yH5kdBGSrM6CowdkkRHTUw..' found DEBUG: ims_dialog [dlg_handlers.c:1950]: dlg_get_msg_dialog(): dlg with callid 'yH5kdBGSrM6CowdkkRHTUw..' not found ERROR: ims_charging [ims_charging_mod.c:511]: w_ro_ccr(): Unable to find dialog and cannot do Ro charging without it The error indicates that w_ro_ccr function doesn’t find any dialog. Following you can find the relative configuration of kamailio.cfg. The request_route block is equal to default configuration except the following command: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_DEBUG loadmodule "cdp" loadmodule "cdp_avp" loadmodule "ims_charging" loadmodule "ims_dialog" modparam("ims_charging", "origin_host", "kcc-t-lu-03.mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "origin_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging", "destination_realm", "mnc049.mcc222.3gppnetwork.org") modparam("ims_charging","interim_update_credits",30) modparam("ims_charging","timer_buffer",5) # -- ims_dialog params -- modparam("ims_dialog", "dlg_flag", FLT_DIALOG) request_route { … if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting t_newtran(); $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", 30, "1", "1"); if ($var(cc_ret) < 0) { xlog("Chiamata a Ro_CCR() fallita\n"); sl_send_reply("402","Payment required"); exit; } } … } route[CHARGING_CCR_REPLY]{ xlog("cca_return code is $avp(s:cca_return_code): $avp(s:cca_result_code)\n"); switch ($avp(s:cca_return_code)){ case 1: #success break; case -1: #failure switch ($avp(s:cca_result_code)){ case 4010: send_reply("402", "Payment required - Unsufficient funds"); break; case 5006: send_reply("486", "Line limit exceeded"); break; default: send_reply("402","Payment required ($avp(s:cca_result_code))"); break; } default: xlog("L_ERR", "Unknown return code from CCR: [$avp(s:cca_return_code)] \n"); send_reply("500", "Charging Error"); break; } } What's goes wrong? Thanks in advance Angelo Colucci

1 0

Feature/help android fcm and ios related push notification module
by Dipen Patel 08 Feb '18

08 Feb '18

hello.. Now most sip soft client in required push wakeup phone. So for that required android, ios push notifications module. I have seen some one have developed 'Tsilo' module based on GCM but it is old. So if possible so please upgrade for Android FCM so it iwll supported for Android and ios both. Thanks

3 2

Using t_relay_to_tcp()
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hi All, Some providers only allow tcp so I switched from t_relay to t_relay_to_tcp(); however, nothing now gets relayed. Has anyone else had luck switching? Thank you, -Steve

3 2

Random 'BYE" sent?
by Wilkins, Steve 08 Feb '18

08 Feb '18

Hello All, I am running Kamailio 5 and using Asterisk 14 as PBX. I have a strange issue. When I receive calls from some providers to a WebRTC client, calls hang up after about 25 seconds. This does not happen with all providers and does not happen with WebRTC client to WebRTC client calls. It appears the "BYE" is coming from Kamailio. Any ideas? Thank you, -Steve

3 2

Re: [SR-Users] How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi. Yep! I found issue was in if. Fixed it. thx for response! On Feb 8, 2018 18:44, "Daniel-Constantin Mierla" <miconda(a)gmail.com> wrote: It looks like you are using it in some conditional expression (like if(...)) or some assignment with integer expression. Can you paste here the part of the config where you use it? Cheers, Daniel On 08.02.18 04:08, Yuriy Gorlichenko wrote: Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int? _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio World 2018: First Group Of Speakers Announced
by Daniel-Constantin Mierla 08 Feb '18

08 Feb '18

Hello, the details for the first group of speakers at Kamailio World 2018 have been published: - https://www.kamailioworld.com/k06/speakers/ So far they come from three continents: Europe, North America and Asia, many presenting for the first time at our event. The two sessions present at all editions so far will be there also in 2018, at our 6th edition, respectively Dangerous Demos with James Body and VUC Visions with Randy Resnick. Besides covering various use cases for Kamailio, Asterisk or FreeSwitch, the sessions go into WebRTC, VoLTE/IMS, IoT, blockchains for telecommunications or scalability using NoSQL data storage systems. Definitely another edition with very interesting content – soon we will publish more details about the sessions as well. You can register now to benefit of the early registration price: - https://www.kamailioworld.com/k06/registration/ Looking forward to meeting many of you at Kamailio World Conference, May 14-16, 2018, in Berlin, Germany! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

How to avoid automatic string to int conversion?
by Yuriy Gorlichenko 08 Feb '18

08 Feb '18

Hi im using kamailio 5.1 with native cfg file and using json_get_value( method) form jsonso module When i get json string that contains strings and i need to have these strings kamailio tries to convert string i find into int and affcource it can't then i see warning and can't get vaules of the string i have rval_expr_eval_int(): rval expression conversion to int failed (24,6-24,14) How to say kamailio to not convers strng to int?

2 1

dmq_usrloc - which server?
by Alex Balashov 07 Feb '18

07 Feb '18

Hi, The Socket parameter of location entries is not replicated under dmq_usrloc. To make matters worse, the technical characteristics of the registrar module are such that I cannot straightforwardly modify the incoming Contact to add a rider indicating the server on which the register was received. The AoR domain is the same for all registrations received on any server in the DMQ mesh. To reach the registrants, I need to forward the registration to the correct server that corresponds to the true provenance of the registration. What is a reasonable way of finding out what that is? I don't see any method of injecting such values into the replicated contacts, as dmq_usrloc does not provide hooks for this. 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/

3 19

any interest in segmentation fault at startup ?
by Jean Cérien 07 Feb '18

07 Feb '18

Hi I guess I'm a bit too creative when it comes to kamailio syntax.... just edited the cfg file and got the following error upon startup: kamctl start INFO: Starting Kamailio : /usr/sbin/kamctl: line 1915: 5294 Segmentation fault $OSERBIN -P $PID_FILE -f $ETCDIR/kamailio.cfg $STARTOPTIONS > /dev/null 2> /dev/null ERROR: PID file /var/run/kamailio.pid does not exist -- Kamailio start failed I assume it is my syntax in cfg that creates this, I just need to correct it, but if there is interest, I can provide the faulty cfg file J.

4 4

Re: [SR-Users] Blind Transfer WS Error
by Matteo Campana 07 Feb '18

07 Feb '18

Dear Kamailians, does anyone have any idea on the issue below? Alessio told me that is always reproducible, and he can also provide captures and more logs. Thanks in advance! Matteo *From: *sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Alessio Miranda <smirolandia(a)gmail.com> *Reply-To: *"Kamailio (SER) - Users Mailing List" < sr-users(a)lists.kamailio.org> *Date: *Tuesday, 16 January 2018 at 14:18 *To: *"sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> *Subject: *[SR-Users] Blind Transfer WS Error Hi, I have a problem with the blind transfer via WebSocket, debugging from the browser we see that in some cases when the REFER is done and the sip message "202 Accepted" arrives, the message "180 Ringing" arrives first and then the "100 Trying" and the transfer goes in error with "500 JsSIP Internal Error" I have an architecture with Kamailio in frontend and asterisk in backend on the same network, the sip dialogue between FE and BE is in SIP / UDP. The WebRTC client connects to WebSocket at Kamailio's public address. We have verified that the sip messages exchanged between kamailio and asterisk are always correct. Kamailio Ver. 4.4.4 Asterisk Ver. 13.18.3 PJSIP Ver. 2.6 JSSIP Ver. 3.2.2 Thanks Alessio

2 2

Contact and Record-Route are using Private IP Address- is this correct?
by Wilkins, Steve 07 Feb '18

07 Feb '18

Hello All, Contact and Record-Route are using Private IP Address - is this correct? I though the Public IP should be used. I do not know what setting is causing this to happen. Does anyone have any ideas? Thank you, -Steve

3 4

Info: Markus Monka replacing Marcus Hunger in management group
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, Markus Monka has just replaced Marcus Hunger in Kamailio project management group. Marcus (still at sipgate) has moved to work more on frontend applications than backend, no longer interacting with Kamailio project. Markus Monka has managed the VoIP operations at sipgate for more than 15 years, helping the project over the years with various resources, mainly in respect of organizing events and testing infrastructure, sipgate being one of the oldest VoIP services using Kamailio (since first releases of SER). The change is reflected now on the website: * https://www.kamailio.org/w/management/ Warm welcome to Markus and many thanks to both of them for what they did so far for Kamailio project! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Info: Kamailio project administration updates
by Daniel-Constantin Mierla 07 Feb '18

07 Feb '18

Hello, after several discussions at some of the past IRC devel meetings, finally we started to build a team to be involved more actively in the administration of Kamailio. The project has grown steadily, not only in terms of code, but also packaging, continuous integration, social networking interactions as well as participation to events world wide. For a better coordination and ability to handle related tasks, I invited the most active developers and community members to join so called Kamailio Administration Team, the initial details about it are published as part of management page on kamailio.org: * https://www.kamailio.org/w/management/ Some more details about its rules and purpose: * https://www.kamailio.org/w/kadmin/ It will still take some time to get it properly rolling, more or less now looking to see if the community has suggestions/improvements on what can be done in these aspects. I wrote to all mailing lists for the initial announcement, but you can just reply to sr-users, if you are subscribed there, otherwise it's fine to write back to any of the lists. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Re: [SR-Users] Contact and Record-Route are using Private IP Address- is this correct?
by Ammar Tinawi 07 Feb '18

07 Feb '18

Is it mandatory to have an interface with public ip in order to overcome NAT problems ?

1 0

Issue with Socket selection on Forwarding ACK message
by Thomas Carvello 06 Feb '18

06 Feb '18

Hello, i have an issue with my Kamailio 4.1.9 configuration. This configuration is multi-homed, we have*two network* interfaces, one on a private network and on the public Internet. Kamailio is configured to listen on port 5060 and 5066 on both interfaces. We register two users Alice and Bob on the public Internet using port 5066. Both users are behind a NAT and we capture the SIP exchange on the proxy server. We have set the parameter mhomed=1 When Alice calls Bob, we have Alice Proxy Bob src=5063 dst=5066 INVITE ------------------> src=5066 ------ INVITE ---------------> dst=5060 dst=5066 <------- 200 OK -------------- src=5060 dst=5063 <------- 200 OK --------- src=5066 src=5063 dst=5066 -------- ACK -----------> *src=5060 (blocked by NAT)* ------ ACK-----x dst=5060 The ACK packet gets relayed with the wrong source port. Then the NAT rejects the packet and the call cannot be established. For some reason, when Bob calls Alice, the call is correctly established. Could it be because Bob happend to use 5060 as local port? Also, if we set nhomed=0 it works BUT we are not sure that multi homed is handled correctly. I was wondering if you have encounter this issue before? I have investigated the code for selection socket and what is the logic of this selection ? /*How does kamailo knows that it should choose 5066 as src port if the user is registered using port 5066 instead of 5066?* / Thank you for your time. Thomas

4 8

rtpproxy - no media connection
by Steve 05 Feb '18

05 Feb '18

Dear sr-users, I am trying to set up a simple Kamailio server with rtpproxy on the same host computer, behind a NAT router. “Run your own Skype-like service” is my set-up guide, but I’m using Kamailio version 5.1 so I modified the config files that came with the download. My problem is that SIP clients can register over the internet via TLS encryption with the server and establish a call session connection, but the ZRTP media connection is not being made. Once the call session connection is made, both clients send out UDP packets to each other, but neither receives them—the media traffic is one way and being dropped somewhere. I’ve tried different rtpproxy start settings. If I start the rtpproxy service with “sudo /etc/init.d/rtpproxy start” then it uses the default “unix:/var/run/rtpproxy/rtpproxy.sock”. I’ve also tried starting with “sudo rtpproxy –l /my_public_ip/ –s udp:localhost:7722” which then asks for either –F (superuser) or –u (user). I’ve tried it both with options “–F” and “-u kamailio kamailio”. All the variations tried so far have the same result as noted above. The server behind the router is in a DMZ zone that bypasses the router firewall. Is there anything in the user-list archives that addresses this issue? Or some likely place to start troubleshooting the problem? Any suggestions would be greatly appreciated. Thanks in advance. Steve --- This email has been checked for viruses by Avast antivirus software. https://www.avast.com/antivirus

1 0

IMS (ng-voice vs github)
by Amar Tinawi 04 Feb '18

04 Feb '18

Hello Dears, can you please tell me what are the differences between the github version of ims modules and the ng-voice modules introduced by Mr Carsten repo ? Thanks in advance

2 1

Fosdem dinner
by Daniel-Constantin Mierla 03 Feb '18

03 Feb '18

Hello, in the past 10 years or so, many of us met at Fosdem conference in Brussels (the edition this year happens during February 3-4) and had a dinner on Saturday evening. I know already couple of people from community that will go there, so the question is if there is any interest to organize again such event. If yes, it will be again on Saturday, on the evening of the 3rd of February. At the past editions, it turned into an all-RTC dinner, but we can consider to have it alone again (eventually with few other invited friends), if there are too many or people want a smaller meeting. Reply this week if you want to join, so we can estimate how many seats we need and have enough time to find a place that can accommodate us. More about Fosdem can be found at: - https://fosdem.org There is a devroom on Sunday with topics covering real time communications. I will have a presentation there. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

10 13

SDP Rewrite issues
by Steven Meyer 02 Feb '18

02 Feb '18

Our topology is: Client <> Client Router <> Internet <> Firewall (NAT) <> Kamailio <> FS Unfortunately, when enabling SDP Rewrite: fix_nated_sdp, 0x08, it rewrites packets to the firewall external IP, not the client router external IP. Right now we have to do a local NATed IP on each phone.

1 0

Routing based on A and B numbers
by Grant Bagdasarian 02 Feb '18

02 Feb '18

Hi, Which module would be best suited for routing calls based on A and B numbers? We're currently using the LCR module, with some additional htables to add this functionality, but I was wondering if there is a module which does exactly this? Or perhaps a combination of modules? Regards, Grant Bagdasarian Senior Developer +31765727054<tel:+31765727054> cm.com<https://cm.com> [cid:image001.png@01D39C36.945A76C0]

2 2

Distributed Registrations.
by Володимир Іванець 02 Feb '18

02 Feb '18

Hello! Yesterday I read email conversation "Distributed Presence between Multiple Kamailios" where it was suggested to use Presence module with new dmq parameter. So was wondering if someone was thinking about approach to share registrations between multiple Kamailio servers without using database? Thanks!

2 1

Contact Header need update?
by Duarte Rocha 02 Feb '18

02 Feb '18

Greetings, My proxy does some number normalization in order to guarantee E164 numeration. It also verifies for valid Calling Numbers and replaces them if needed. In order to this, i'm updating the $fU pseudo-variable. Should i update the "Contact" Header too? I've made some testing and i can get calls to work with keeping the Contact intact and changing it to the new number. What is the most correct approach? Cheers, Duarte Rocha

1 0

Distributed Presence between Multiple Kamailios
by SamyGo 01 Feb '18

01 Feb '18

Hi, I'm working on multiple Kamailios that have users distributed evenly among them, I now need to enable presence in those Kamailio boxes. The scenario is a mesh where any user may have subscribed to BLF of other user registered & calling on any other proxy. What modules can help me achieve this? can this be done without creating a new Presence-Node for the whole environment? Hope to get some suggestions on how experts here have done this. Looking forward to positive response. Regards, Sammy

3 5

Case insensitive regexp in textops::search() function
by Andrey Deykunov 01 Feb '18

01 Feb '18

Hi, I have an htable containing some black list words (i.e. user IDs or phone models). To find matches of the words in every message I use the following script: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { if ( search($shtitval(i1)) ) { // do something } } sht_iterator_end("i1"); How can I modify search($shtitval(i1)) to get case insensitive matching? The example below does not work: sht_iterator_start("i1", "black_list"); while(sht_iterator_next("i1")) { $var(re) = $shtitval(i1) + "/i"; if ( search($var(re)) ) { // do something } } sht_iterator_end("i1"); Other possible modifications of $var(re) aren't working too. Thanks, Andrey

2 2

Partial/regex match on hash table key?
by Anthony Joseph Messina 31 Jan '18

31 Jan '18

Is is possible (without iterating through all entries in a hash table) to return the value based on a partial or regex match to the key? Something like... $var(one)=$sht(a=>*$var(two)) -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

2 2

question about force_send_socket
by Fabian Borot 31 Jan '18

31 Jan '18

our kamailo has 2 IPs, a public (87.x.x.x) and a private (192.168.1.1). It is listening on all of them. For all calls it listens on the public IP but for some calls I need to forward the incoming INVITE to a server in the private network so I do a " force_send_socket(privateip:port)" before the route that forwards the INVITE to the server in the private network and that works. But then when the server in the private network replies with 302 I have to contact the servers inside the Contact header using the Public IP now so I do this hoping that it will switch back to the public IP: if (t_check_status("302")) { force_send_socket(PUBLIC_IP:5060); get_redirects("*"); t_relay(); } But it stays in the private, so the INVITEs to the servers inside the Contact header are sent from the Private IP and they are blocked. This would be the flow: customer INVITE -> kamailio(publicIP) kamailio (publicIP) 100 trying -> customer kamailio(privateIP) INVITE -> Server(privateIP) Server(privateIP) 302 redirect -> kamailio(privateIP) kamailio(privateIP) ACK -> Server(privateIP) after I would like this to happen kamailio(publicIP) INVITE -> Server1(publicIP) but instead this happens: kamailio(privateIP) INVITE -> Server1(publicIP) how can I switch back to the public IP to continue with this call flow? thank you

2 2

[4.2] how to send out BYE successfully to close a TCP/TLS connection which misses the connection ID?
by Jack Wang 31 Jan '18

31 Jan '18

For some reasons, My settings will cause kamailio to be restarted even if there were connections connected to it. And when one SIP client in a connection wanted to hangup the call with a WebRTC Client (SIP over websocket) after the proxy-restarting, local client was closed, however , the other side was not. >From the log below it seems that the connection missed. Jan 30 18:19:31 ./kamailio[27857]: WARNING: <core> [msg_translator.c:2777]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1365]: assemble_via(): assemble_via: via building failed Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [t_msgbuilder.c:1545]: build_uac_req(): build_uac_req(): Error while assembling Via Jan 30 18:19:31 ./kamailio[27857]: ERROR: tm [uac.c:379]: t_uac_prepare(): t_uac: Error while building message So, Is it equivalent to connection ID missing? And is there any way to make Websocket found the connection? Btw, after proxy-restarting for seconds , I typed `sh kamctl stats` : websocket:ws_current_connections = 1 It showed that the WebRTC Client will re-connect automatically, So I think this connection could be found by proxy, Any suggestion? Thanks :)

1 2

Normal behaviour with loose_routing
by Loic Chabert 30 Jan '18

30 Jan '18

Hi list, I have a setup configurered has diagram below: UAC -> Proxy LB (proxy lb) -> Proxy Router/registrar (proxy B) -> Asterisk SBC My UAC is SIPP and sending a classic call scenario: INVITE - ACK - BYE. But i have some trouble to understand which queries should be loose routed and which queries should be dispatched by dispatcher module. My first invite request has been dispatched to proxy B by proxy LB. After that, ACK and BYE request should be loose_routed or should be dispatched as other request ? My proxy B is a statefull proxy, logging cdrs and other stuff ... >From my point of view, subsequent request should be loose_routed to keep dialog informations consistent. My subsequent requests have record-route headers. Should i use theses headers to route my calls ? Loose routing features only work with "Route" header ? What is the difference beetwen Route headers and Record-route headers ? Thanks in advance, Regards.

3 5

rtjson + uac + event - OK, ACK and BYE display name not updated
by Diego Nadares 30 Jan '18

30 Jan '18

Hi Guys, We are modifying "from" and "to" headers with rtjson module via a json struct. It works fine in INVITE, TRYING, RINGING and OK but not in the following messages. Kamailio It's changing ONLY "from" and "to" URI BUT NOT the DISPLAY NAME. I think it's the tm module but I can't figure out how to confirm this. Maybe I need to configure something else in my cfg or could it be a bug? My json struct that is handled by rtjson module response.routing = "serial"; //response.routing = "parallel"; response.routes = []; response.routes[0] = {}; response.routes[0].uri = "sip:1111111111@172.16.213.31:5060"; response.routes[0].headers = { // Headers to be modified "from": { "display": "1111111111", "uri": "sip:111111111@172.16.213.38" }, "to": { "display": "2222222222", "uri": "sip:2222222222@172.16.213.38" }, }; response.routes[0].headers.extra = {};//"X-Hdr-A: abc\r\nX-Hdr-B: bcd\r\n"; This is part of my cfg .... modparam("dialog","dlg_flag", FLD_START) modparam("uac","restore_mode","auto") modparam("uac", "restore_dlg", 1) .... # account only INVITEs if (is_method("INVITE") && !has_totag()) { setflag(FLT_ACC); # do accounting * setflag(FLD_START); # do accounting* setflag(FLT_ACCMISSED); setflag(FLT_ACCFAILED); } ..... >From source/Aleg: *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* To dest/Bleg: *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Incorrect display name in BYE and ACK: *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* interface: any filter: ( port 5060 ) and (ip or ip6) # U 2016/12/06 15:52:13.638731 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:5555555555@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: "SIPP" <sip:sipp@172.16.213.21:5060>. Max-Forwards: 70. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Remote-Party-ID: <sip:sipp@172.16.213.21 >:5060;party=calling;id-type=subscriber;privacy=off;screen=no>. Content-Length: 549. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.21. s=-. c=IN IP4 172.16.213.21. t=0 0. m=audio 6000 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.734159 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Server: kamailio (4.4.4 (x86_64/linux)). Content-Length: 0. . ................ # U 2016/12/06 15:52:13.878600 172.16.213.38:5060 -> 172.16.213.31:5060 INVITE sip:1111111111@172.16.213.31:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Max-Forwards: 69. Subject: Performance Test. Content-Type: multipart/mixed;boundary=uniqueBoundary. Content-Length: 788. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . --uniqueBoundary. Content-Type: application/sdp. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40236 RTP/AVP 8. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11,16. a=sendrecv. a=rtcp:40237. a=ice-ufrag:QNCzXhUr. a=ice-pwd:Ci90BRpdZH51Hx5BaPgGMrGCeh. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40236 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40237 typ host. . --uniqueBoundary. Content-Type: application/gtd. Content-Disposition: signal;handling=optional. . IAM,. CGN,04,y,1,y,4,1143617400. CIC,0000000573. CPC,09. CPN,02,y,1,52381660. FCI,n,n,n,n,y,n,n,u. GCI,f7140cc78a611601838a002128d7e512. NOC,0,n,1,n. PRN,q761*,AR*,oper2,1993. TMR,02. . --uniqueBoundary--. ................ # U 2016/12/06 15:52:13.879441 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. ................ # U 2016/12/06 15:52:13.936565 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 180 Ringing. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 316. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:20.882165 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.840cc83114975a34a37b7af020d37b65.1. *From: 1111111111 <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: 2222222222 <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 213. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.31. s=-. c=IN IP4 172.16.213.31. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. ................ # U 2016/12/06 15:52:21.173798 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 INVITE. Content-Type: application/sdp. Content-Length: 380. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-0. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . v=0. o=user1 53655765 2353687637 IN IP4 172.16.213.38. s=-. c=IN IP4 172.16.213.38. t=0 0. m=audio 40248 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=sendrecv. a=rtcp:40249. a=candidate:P1djl8LFNb5kexbG 1 UDP 2130706431 172.16.213.38 40248 typ host. a=candidate:P1djl8LFNb5kexbG 2 UDP 2130706430 172.16.213.38 40249 typ host. ................ # U 2016/12/06 15:52:21.174390 172.16.213.21:5060 -> 172.16.213.38:5060 ACK sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-4. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:21.248172 172.16.213.38:5060 -> 172.16.213.31:5060 ACK sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bKabaa.98116b5795d4c30f52156187a72059c8.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 1 ACK. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.178826 172.16.213.21:5060 -> 172.16.213.38:5060 BYE sip:atpsh-58470879-6c61-2@172.16.213.38 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: sip:sipp@172.16.213.21:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.468639 172.16.213.38:5060 -> 172.16.213.31:5060 BYE sip:172.16.213.31:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Max-Forwards: 69. Subject: Performance Test. Content-Length: 0. Contact: <sip:btpsh-58470879-6c61-1@172.16.213.38>. . ................ # U 2016/12/06 15:52:51.469153 172.16.213.31:5060 -> 172.16.213.38:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.16.213.38;branch=z9hG4bK7baa.3cb44935b506e759e3418501b0688b08.0. *From: sipp <sip:111111111@172.16.213.38 <sip%3A111111111(a)172.16.213.38>>;tag=1.* *To: sut <sip:2222222222@172.16.213.38 <sip%3A2222222222(a)172.16.213.38>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Contact: <sip:172.16.213.31:5060;transport=UDP>. Content-Length: 0. . ................ # U 2016/12/06 15:52:51.519423 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 200 OK. *From: sipp <sip:sipp@172.16.213.21:5060 <http://sip:sipp@172.16.213.21:5060>>;tag=1.* *To: sut <sip:5555555555@172.16.213.38:5060 <http://sip:5555555555@172.16.213.38:5060>>;tag=2.* Call-ID: 1-12775(a)172.16.213.21. CSeq: 2 BYE. Content-Length: 0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-12775-1-7. Contact: <sip:atpsh-58470879-6c61-2@172.16.213.38>. . ................ exit 13 received, 0 dropped Sorry for my english. Thanks in advance! Diego.

2 7

Session Timers
by Duarte Rocha 30 Jan '18

30 Jan '18

Hey guys, I have two issues about Session Timers that need some clarification. 1) I have my Kamailio between my clients CPE's and my GW to the PSTN. I want to configure it in a way that i always try to make the CPE do the refreshing so I can save resources on my kamailio and in the GW. If the CPE doesn't support it, the refresh should be done by the GW. Is it possible to do this test and configuration in every call ? 2) SIP has the MIN-SE mechanic and is well documented, but is it possible to implement a MAX-SE in any way ? Thanks for the help. If you need more details about the issue, please let me know. Cheers

2 1

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Victor Seva 30 Jan '18

30 Jan '18

2018-01-26 17:46 GMT+01:00 Alain Bonnefoy - XOOL <a.bonnefoy(a)xool.fr>: > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > Wiki updated. Thanks

1 0

Re: [SR-Users] Problem to install Kameilio from Ubuntu packages
by Bastian Triller 30 Jan '18

30 Jan '18

There another command on http://deb.kamailio.org for people behind firewalls which block port 11371: % wget -O- http://deb.kamailio.org/kamailiodebkey.gpg | sudo apt-key add - On Jan 26, 2018 5:46 PM, "Alain Bonnefoy - XOOL" <a.bonnefoy(a)xool.fr> wrote: > Hello, > > > > I tried to install Kameilio on Ubuntu 16.04 from what explained in the > wiki <https://www.kamailio.org/wiki/packages/debs>. > > > > But when I execute the command to add your GPG key, I just get replied > that the timeout occured. > > > > What to do ? > > > > Alain > > Provenance : Courrier <https://go.microsoft.com/fwlink/?LinkId=550986> > pour Windows 10 > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

1 0

kamailio tls passthrough
by pavel＠eremina.net 30 Jan '18

30 Jan '18

Hi, All i am trying to load balancing Kamailio with TLS in passthrough mode. is it possible to do it in stateless mode?

2 1

whats happened to iptel.org
by voipspace voipspace 29 Jan '18

29 Jan '18

Anyone know whats happened to iptel.org The webiste is now a wix page iptel.org the sip uri music(a)iptel.org was a useful resource. Now the song has changed and the service is down alot Has it it changed hands or been sold?

1 0

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